- sr-users - kamailio.org

[Kamailio-Users] using a proxy and a registrar(only) server to make a cal
by Rajesh 13 Oct '09

13 Oct '09

Dear All, I would like to establish a set up where i have two servers, one acting as a proxy and the other acting as a registrar server(only). Client A ---->Proxy--->Registrar. I am able to register using the above mentioned set up, but am unable to make call.. i get "timed out error" when i try calling. am ataching my cfg below.Any help is greatly appreciated. (My proxy server wil not do registration and my registrar server wil only do registration) ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication xlog("__The next step is Registration__\n"); rewritehost("192.168.64.77"); xlog("__Registration Successful__\n"); #nr route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain #nr if( is_method("PUBLISH|SUBSCRIBE")) #nr route(2); #nr if (is_method("REGISTER")) #nr { #nr if (!save("location")) #nr sl_reply_error(); #nr exit; #nr } ########## if(uri=~"sip:q.*@") # if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { #imc#if (!proxy_authorize("", "subscriber")) { #imc#proxy_challenge("", "0"); #imc#exit; #imc#} #imc#if (is_method("PUBLISH")) #imc#{ #imc# if ($au!=$tU) { #imc# sl_send_reply("403","Forbidden auth ID"); #imc# exit; #imc# } #imc#} else { #imc# if ($au!=$fU) { #imc# sl_send_reply("403","Forbidden auth ID"); #imc# exit; #imc# } #imc#} #imc# consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } -- Regards, Rajesh

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Re: [Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Henning Westerholt 13 Oct '09

13 Oct '09

On Montag, 12. Oktober 2009, mashudi wrote: > Hi Henning Westerholt , > did you ever handle SIP-T using Kamailio? > could you share it? Hi Mashudi, sorry, i did not handled SIP-T so far with openser/kamailio. Regards, Henning > > On Friday 13 February 2009, Ovidiu Sas wrote: > >> The multipart body content parsing is embedded in the sdp parser. > >> The sdp parser is available in the source tree under the ./parser/sdp > >> directory. > > > > There is also some documentation about the parser on our wiki: > > > > http://www.kamailio.org/dokuwiki/doku.php/development:sdp-parsing > > > > Cheers, > > > > Henning > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > > > ***************************************** > > Sekarang Gratis Nelpon SLJJ Flexi diperluas ke > > Yogya > > ***************************************** > > ================================= > Mau GRATIS TELPON LOKAL, DISCOUNT 50% SMS, > DISCOUNT 20% SLJJ, dan DISCOUNT FLEXI MILIS? > Ikuti Dahsyatnya FLEXI KOMUNITAS. > Ketik CREATE<spasi>[NAMA GRUP], sms ke 345. > Contoh: CREATE SMU2, sms ke 345. > Informasi selanjutnya : > - hubungi 147 > - http://www.telkomflexi.com > - ketik INFO, sms ke 345 > > > ================================= > This Email has been scanned : Anti Virus > =================================

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[Kamailio-Users] Sip-T
by ljjimenez＠gmail.com 12 Oct '09

12 Oct '09

Hi list, can somebody guide on how to read, if it is possible, application/isup payload sent using SIP-T to do routing decisions based of the contents. Thanks in advance Luis Jimenez Luis Jimenez

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[Serusers] Via: branch tags for CANCELs sometimes don't match the one in the INVITE
by Frank Durda IV 12 Oct '09

12 Oct '09

I have been tracking a problem for some time regarding CANCELs that the switch beyond SER was randomly ignoring. There appear to be a number of issues, but one that was identified is caused by SER unexpectedly emitting a branch tag on the Via: header of the CANCEL with a value that isn't identical to the value that appeared in the INVITE message. When the CANCEL arrives at the destination switch with a different branch tag than what was shown in the INVITE, the destination switch correctly rejects the CANCEL with a 481, saying it never heard of this call. So the call doesn't really end. They say they are following RFC 3261, and that does state that the Via: header MUST exactly match. This variance of the branch tag that SER emits doesn't happen every time and is infrequent and random, occurring, perhaps 1% of the time. 99% of the time, the same branch value appears in both INVITE and CANCEL messages. This occasional mismatch is being seen in 2.0.0-rc1 on FreeBSD, Server: SER (2.0.0-rc1 (x86_64/freebsd)) syn_branch=0 is set because branches of zero in ACKs were a really bad thing too. Based on a number of these that have been reviewed, the branch tag generated by SER has three parts, eg branch=STRING1.STRING2.0 I note that STRING1 remains the same but when the malfunction occurs, STRING2 is completely different, as in: FROM INVITE branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c11079.0 FROM CANCEL branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd34935.0 Below is a sample call showing one of these events. Any suggestions on what the problem might be or if this is a known bug (hopefully with a patch) are appreciated. - - - - - - - - - - - - - - - - - - Here is a sample call. IP addresses and phone numbers have been altered, but call ID and branch tags are untouched: CALL ARRIVES AT SER U 2009/10/05 23:00:45.616070 216.0.0.1:5060 -> 208.9.10.11:5060 INVITE sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C P-Asserted-Identity: <sip:8881112222@216.0.0.1:5060> To: sip:9615551212@208.9.10.11:5060 Contact: <sip:8881112222@216.0.0.1:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Max-Forwards: 70 Content-Length: 281 Allow: INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 v=0 o=SIPUAC_1.92.1.8 2077471748 1 IN IP4 55.66.77.88 s=call c=IN IP4 55.66.77.88 t=0 0 m=audio 21356 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 SER SENDS TRYING BACK TO CALLING SWITCH U 2009/10/05 23:00:45.626174 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 100 trying -- your call is important to us From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: sip:9615551212@208.9.10.11:5060 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: SER (2.0.0-rc1 (x86_64/freebsd)) Content-Length: 0 SER SENDS CALL ON TO PSTN SWITCH U 2009/10/05 23:00:45.626192 10.100.20.42:5060 -> 10.133.44.55:5060 INVITE sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C P-Asserted-Identity: <sip:8881112222@216.0.0.1:5060> To: sip:9615551212@208.9.10.11:5060 Contact: <sip:8881112222@216.0.0.1:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE Max-Forwards: 16 Content-Length: 301 Allow: INVITE, ACK, CANCEL, BYE Content-Type: application/sdp Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 v=0 o=SIPUAC_1.92.1.8 2077471748 1 IN IP4 55.66.77.88 s=call c=IN IP4 55.66.77.88 t=0 0 m=audio 55878 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:30 a=nortpproxy:yes CALLING SWITCH TELLS SER IT IS TRYING U 2009/10/05 23:00:45.629000 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 100 Trying Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Content-Length: 0 CALLING PARTY CHANGES MIND - GOES ON-HOOK U 2009/10/05 23:00:50.579059 216.0.0.1:5060 -> 208.9.10.11:5060 CANCEL sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 SER SENDS CANCEL TO PSTN SWITCH, BUT BRANCH TAGS DON'T MATCH: U 2009/10/05 23:00:50.590039 10.100.20.42:5060 -> 10.133.44.55:5060 CANCEL sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060> Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Max-Forwards: 16 Content-Length: 0 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd3493 5.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 NOTE BRANCH MISMATCH: FROM INVITE branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c11079.0 FROM CANCEL branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd34935.0 SER SENDS 200 CANCELING BACK TO CALLING SWITCH U 2009/10/05 23:00:50.590054 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 canceling From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=68cf7f6b22161b0d4dc719dbc13e05af-90a6 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: SER (2.0.0-rc1 (x86_64/freebsd)) Content-Length: 0 PSTN SWITCH REJECTS CANCEL, NOT RECOGNIZING THE CALL PSTN NETWORK CONTINUES TO PROCESS CALL DESPITE CALLER NOW BEING ON-HOOK U 2009/10/05 23:00:50.591489 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 481 Call/Transaction Does Not Exist Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 CANCEL From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+0+b6afb3ac Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.b4c66001f96f5dd30ddbf6536fd3493 5.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Content-Length: 0 PSTN SWITCH SENDS BACK ACM/183 Session Progress TO SER U 2009/10/05 23:00:52.619487 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 183 Session Progress Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER SENDS 183 SESSION PROGRESS BACK TO CALLING SWITCH U 2009/10/05 23:00:52.629924 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 183 Session Progress Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER RECEIVES 200 OK FROM PSTN SWITCH U 2009/10/05 23:00:58.129670 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no SER PASSES 200 OK ON TO CALLING SWITCH U 2009/10/05 23:00:58.137724 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no BECAUSE OF NO RESPONSE TO 200 OK, PSTN SWITCH SENDS IT AGAIN U 2009/10/05 23:00:58.629267 10.133.44.55:5060 -> 10.100.20.42:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.6082c3659eb65543a8e2788f54c1107 9.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@10.133.44.55:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no U 2009/10/05 23:00:58.639701 208.9.10.11:5060 -> 216.0.0.1:5060 SIP/2.0 200 OK Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 INVITE From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_1 Server: DC-SIP/2.0 Contact: <sip:9615551212@208.9.10.11:5060;user=phone> Content-Type: application/sdp Content-Length: 223 v=0 o=- 3463772452 3463772517 IN IP4 10.133.44.55 s=- c=IN IP4 10.200.20.40 t=0 0 m=audio 50924 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no CALLING SWITCH FINALLY SENDS AN ACK TO SER TO GET RETRIES TO STOP. U 2009/10/05 23:00:58.690296 216.0.0.1:5060 -> 208.9.10.11:5060 ACK sip:9615551212@208.9.10.11:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 ACK Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_2 SER FORWARDS ACK TO PSTN SWITCH. PSTN NETWORK THINKS CALL IS ALIVE, VOIP NETWORK KNOWS CALL IS DEAD. U 2009/10/05 23:00:58.690355 10.100.20.42:5060 -> 10.133.44.55:5060 ACK sip:9615551212@10.133.44.55:5060;user=phone SIP/2.0 From: <sip:8881112222@216.0.0.1:5060>;tag=2077471748_C To: <sip:9615551212@208.9.10.11:5060>;tag=000a0285+1+21a4018b+2cd8ce96 Call-ID: sbcsipuac.2_11.22.33.44_b09sb07_1_2009100519004555_2077471748_716911 CSeq: 1005 ACK Max-Forwards: 16 Content-Length: 0 Via: SIP/2.0/UDP 10.100.20.42;branch=z9hG4bK7314.1fc8626bef702f1428ad6fd4fd2aa7a f.0 Via: SIP/2.0/UDP 216.0.0.1:5060;branch=z9hG4bK_2077471748_2110_2 .... So the called party has answered the call and can't hear anything. Depending on whether a human or automation was called, this call could remain active on the PSTN for some time.

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 12 Oct '09

12 Oct '09

On Montag, 12. Oktober 2009, Vibhor Singhal wrote: > Please find attached the text file that contains the error that I encounter > while installing Kamailio.I could not get any solution for this.Please > guide me how to resolve this problem. Hi Vibhor, just paste the error message in the mail, its easier then this office document.. You need to install the correct mysql development libraries to be able to compile the db_mysql module. For debian you'll find the package names in the documents in the wiki i posted to you i think. And i'd suggest that you use the latest stable version of kamailio "1.5.2" instead of 1.5.0. This way you'll not run into bugs that were already fixed. Regards, Henning

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 12 Oct '09

12 Oct '09

On Montag, 12. Oktober 2009, Vibhor Singhal wrote: > Thanks for the information. Can you please refer any forum or document that > can guide me on how to make Kamailio as Asterisk's frontend. Hi Vibhor, please CC always to the user list. Just use the search machine of your choice. ;) Here are some example posts with more information: http://www.voip-info.org/wiki/view/Asterisk+at+large http://www.kamailio.org/pipermail/users/2008-February/016181.html http://www.nabble.com/Load-Balancing-Asterisk-using-Kamailio-td23250215.html > And in Kamailio's SIP_trace module, will I get SIP message in the raw form > as mentioned below: Hm, never used this module, but i think it will store the messages in the database. But you could modfiy the module, its open source after all.. Regards, Henning

2 1

[Kamailio-Users] Retransmitted SUBSCRIBEs
by catalina oancea 12 Oct '09

12 Oct '09

Hello, If a phone re-sends a presence SUBSCRIBE requests even after 5 or 500 milliseconds, kamailio replied with '400 Bad request' to the second SUBSCRIBE The scenario is: Phone Kamailio SUBSCRIBE ----------> 401 <---------- SUBSCRIBE ----------> 202 <---------- Pause [ 5ms] (or 500 ms, the same thing happens) SUBSCRIBE ----------> (previous SUBSCRIBE retransmitted) 400 Bad request <---------- Is this the expected behavior? Is there any way to get kamailio to recognize the retransmission? This causes problems with Snom phones because they receive 400 reply when they retransmit the SUBSCRIBE, and presence stops working. Thanks Catalina

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[Kamailio-Users] off-topic :: empty user part of Contact header
by kokoska rokoska 12 Oct '09

12 Oct '09

Hello, I have a little bit off-topic question, but - may be - someone could give me an advice :-) One of our users, when replying to INVITE, sends in 200 OK Contact header without user part: SIP/2.0 200 OK CSeq: 102 INVITE Contact: <sip:1.2.3.4:1234> ... instead of: SIP/2.0 200 OK CSeq: 102 INVITE Contact: <sip:user@1.2.3.4:1234> ... and it fools our SIP<->ISUP GW and it generates completely strange R-URI in ACK :-( My questions are: 1. Is it valid Contact (without user part)? 2. What SIP<->ISUP GW should do (mirror Contact without user part to R-URI in ACK, reject/drop 200 OK etc.)? May be I can ask them to change behaviour... 3. What should do our proxy (drop 200 OK, using htable try to repair R-URI in ACK or Contact in 200 OK etc.)? 4. ??? Any suggestion is very appreciated :-) Best regards, kokoska.rokoska

2 2

Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Henning Westerholt 09 Oct '09

09 Oct '09

On Freitag, 9. Oktober 2009, Vibhor Singhal wrote: > We are developing CTI applications where we need SIP data from PBX such as > SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for > each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY > etc. > > e.g Wrieshark displays SIP packets for activities going on SIP port in the > below mentioned format: > [..] > > For this we are trying Kamailio as Asterisk frontend.Plz tell us about the > installation process and how to make kamailio as asterisk's fromtend Hallo Vibhor, for installation of kamailio take a look to the wiki, for example at: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn In the "installation" section there are also some more informations: http://www.kamailio.org:80/dokuwiki/doku.php How to use kamailio as a frontend to asterisk has been discussed a lot on this list in the past, the easiest way is probably use the "dispatcher" module. Take a look to the docs for this module for more informations. Best regards, Henning

1 0

[Kamailio-Users] Presence issue - pua dialoginfo
by Francisco Javier Lizarán Vilches 09 Oct '09

09 Oct '09

Hi all: I have a problem using Kamailio 1.5.2 as proxy&presence server. User A is subscribed to B's event dialog, and Kamailio handles this subscription. When a call arrives to pstn number C, the proxy resolves it via ENUM and sends the INVITE to B. The issue is that the PUBLISH request uri is C, because pua_dialoginfo module looks into To uri instead of Request uri, so A is not informed about B state. Has anybody any idea to work around this? Thanks: Francisco Javier Lizarán Vilches ---------------------------------------------- BARIK (Grupo Ormazabal) Departamento Técnico Tel.: +34 917 479 900 www.barik.es <http://www.barik.es/> ----------------------------------------------

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Iñaki Baz Castillo 09 Oct '09

09 Oct '09

2009/10/9 Vibhor Singhal <vibhor.singhal(a)coraltele.com>: > We are a telcom company in Noida developing CTI applications. > We are using Asterisk-1.4.and have configured manager.conf file and > connection with the AMI is successfull using username and password. We want > to monitor/fetch SIP packets from TCP port. Please let us know how to take > SIP data from tcp port. This is a list about Kamailio, not about Asterisk.

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Re: [Kamailio-Users] How to fetch SIP Data from Asterisk?
by Raúl Alexis Betancor Santana 09 Oct '09

09 Oct '09

On Friday 09 October 2009 06:56:51 Vibhor Singhal wrote: > Hi, > > > We are a telcom company in Noida developing CTI applications. > We are using Asterisk-1.4.and have configured manager.conf file and > connection with the AMI is successfull using username and password. We want > to monitor/fetch SIP packets from TCP port. Please let us know how to take > SIP data from tcp port. First of all .. this is not the correct list to ask this kind of questions . Then Asterisk does not support SIP over TCP (not yet). > Our scenario: > > We are developing CTI applications where we need SIP data from PBX such as > SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for > each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY > etc. Asterisk will not give you that information over the AMI, you will need to develop a module for asterisk or modify chan_sip to archive that goal. > e.g Wrieshark displays SIP packets for activities going on SIP port in the > below mentioned format: > After receiving these SIP Packets from Asterisk,we can parse that data > according to our requirements. > Is there any text/data file or database where asterisk stores all the > information in above format? No > If we need to fetch this data at realtime from the port than how can we do > that? You have toons of ways for skying a cat ... you could ... : - Develop your custom sip-cature program, using libpcap for example - Work with ngrep for getting about what you need - Develop a wireshark filter. - ... > For this we are trying Kamailio as Asterisk frontend.Plz tell us about the > installation process and how to make kamailio as asterisk's fromtend > > > Request for an early reply!!! > Regards If you need such kind of work and you don't have time to learn yourself, better to look up for a consultant and pay for that work. -- Raúl Alexis Betancor Santana Dimensión Virtual

2 1

[Kamailio-Users] How to fetch SIP Data from Asterisk?
by Vibhor Singhal 09 Oct '09

09 Oct '09

Hi, We are a telcom company in Noida developing CTI applications. We are using Asterisk-1.4.and have configured manager.conf file and connection with the AMI is successfull using username and password. We want to monitor/fetch SIP packets from TCP port. Please let us know how to take SIP data from tcp port. Our scenario: We are developing CTI applications where we need SIP data from PBX such as SourceIP,Username,Destination IP,TAG,VIA,Contact,CALL-ID,C-Sequence etc for each of the SIP packets such as REGISTER,BYE,INVITE,ACK,SUBSCRIBE,NOTIFY etc. e.g Wrieshark displays SIP packets for activities going on SIP port in the below mentioned format: o SIP Packet as displayed on wireshark******************* SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.35:5060;branch=z9hG4bK10e203f8;rport=5060 Contact: <sip:1000000@192.168.3.15:45118> To: <sip:1000@192.168.3.15:45118>;tag=1f0f342c From: "Unknown"<sip:Unknown@192.168.3.35>;tag=as4a585648 Call-ID: 38670a55c46d1235YzY2YWQxZTM0ODk4NTRhZDI5MjY3YjMyYjE4OTlhZjc. CSeq: 102 NOTIFY User-Agent: X-Lite release 1003l stamp 30942 Content-Length: 0 After receiving these SIP Packets from Asterisk,we can parse that data according to our requirements. Is there any text/data file or database where asterisk stores all the information in above format? If we need to fetch this data at realtime from the port than how can we do that? For this we are trying Kamailio as Asterisk frontend.Plz tell us about the installation process and how to make kamailio as asterisk's fromtend Request for an early reply!!! Regards Regards, Vibhor Singhal Software Engineer Mob : +919891380819 " If you fail to plan,you plan to fail" Coral Telecom Limited E2, Sector 63, Noida - 201301, UP, India. F: +91-120-4035888 E: corporate(a)coraltele.com www.coraltele.com <http://www.coraltele.com/index.asp>

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[Kamailio-Users] Thousands of CANCEL messages - problem with my configuration?
by CSB 08 Oct '09

08 Oct '09

We had a very strange situation last night with our proxy (running OpenSer 1.3). We received an INVITE from a UA. This was then cancelled by the UA. This then resulted in OpenSER generating 40000+ cancel messages with our server talking to itself i.e. the toip and fromip are the same in the sip_trace table. Incidentally, this proxy processes thousands of calls a day so I'm not sure what gave rise to this problem but I guess a particular set of events has highlighted a bug in my config file. The other very curious thing is that somehow our other SIP proxy gets in on the act. Cancel message 40000-odd looks like this: CANCEL sip:031234567@domain.com SIP/2.0 Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.002;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Record-Route: <sip:147.202.001.001;lr=on;ftag=as02736614> Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.5d037b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.2d037b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.67037b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.20037b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.8ff27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.e16dcb3.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.516dcb3.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.3c5dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.05e27b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.b4e27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.365dcb3.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.52e27b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.be4dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.33c27b85.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.2d3dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.84a27b85.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.da2dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.002dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.561dcb3.1 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.111dcb3.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.4b627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.ca627b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.5a627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.1a627b85.0 Via: SIP/2.0/UDP 147.202.001.002;branch=z9hG4bK223e.ed0dcb3.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.d9627b85.1 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.c9627b85.0 Via: SIP/2.0/UDP 147.202.001.001;branch=z9hG4bK223e.b9627b85.0 Via: SIP/2.0/UDP 210.48.001.001:5060;branch=z9hG4bK519fe576;rport=5060 From: "Customer" <sip:12345678@domain.com>;tag=as02736614 To: <sip:031234567@domain.com> Contact: <sip:6496306317@210.48.001.001> Call-ID: 771d537932ab79225888a341190691c2(a)domain.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 41 Content-Length: 0 I have pasted what I think are the relevant parts from the config file. Could anyone advise where the problem might be? route { # ----------------------------------------------------------------- ### SipTrace. # ----------------------------------------------------------------- if ! is_method("OPTIONS|SUBSCRIBE|NOTIFY") { sip_trace(); setflag(28); } # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- #If it's an INVITE & client is NATed, #xlog("welcome"); if (method=="INVITE" && client_nat_test("3")) { #Record-route and specify the record-route header explicitly record_route_preset("147.202.001.001:5060;nat=yes"); # insert IP address xlog("invite and nated so record_route_preset"); #If not Nated and not REGISTER then normal record-route } else if (method!="REGISTER") { record_route(); xlog("not register and not nated so record_route"); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); #call end_media_session in case we're proxying media end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { #Ensure we are dealing with a re-INVITE. Only connected calls have tag= #entry in the To header. So if it's loose routed and doesn't have totag #and it's an invite or reply then something's wrong xlog("loose_route"); if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); xlog("403 in lr"); return; }; if (method=="INVITE") { if (!allow_trusted()) { xlog("!allow_ trusted for r-uri <$ru>"); if (!proxy_authorize("","subscriber")) { proxy_challenge("domain.com","1"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; #check client is nated or that we've already identified it's nated xlog("Re-invite. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); xlog("Recipient is nated so setbflag 6 and use mediaproxy: r-uri <$ru>"); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # Processes out-of-dialogue messages i.e. # 1. new dialogue # 2. message not destined for us (but we are relaying or proxying) xlog("Call Type Processing Section"); if (!is_uri_host_local()) { xlog("!is_uri_host_local"); if (is_from_local() || allow_trusted() ) { xlog("is_from_local() || allow_trusted()"); route(5); route(1); } else { sl_send_reply("403", "Forbidden"); xlog("403 in call type processing. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); }; return; }; sip_trace(); if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="SUBSCRIBE") { route(7); return; }; route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- xlog("hit route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); #Call reply_route(1) to intercept response messages heading to the client t_on_reply("1"); #Try to relay the message to its destination if (!t_relay()) { #If it can't and it's an INVITE or ACK end the media proxying log(1,"route[1] !t_relay"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); } else { log(1,"route[1] t_relay"); }; } onreply_route[1] #Handles message that are returned to the sender i.e. response to caller's original request { xlog("hit onreply_route(1). rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); if (isbflagset(6)) { xlog("flag(6) is set for $fu . We're currently in onreply_route(1)"); }; if (isbflagset(7)) { xlog("flag(7) is set for $fu . We're currently in onreply_route(1)"); }; if (status=~"(180)|(183)|2[0-9][0-9]") { xlog("status matches for $fu . We're currently in onreply_route(1)"); }; if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { #Check the SDP payload length. Assume if we have something we can call mediaproxy if (!search("^Content-Length:[ ]*0")) { log(1,"using media proxy in onreply_route(1)"); use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Any advice appreciated. Cameron

3 3

[Kamailio-Users] BLF and MWI problem on Grandstreams
by David 08 Oct '09

08 Oct '09

All, I upgraded to Kamailio 1.5 and had a lot of problems getting BLF and MWI working. So I will list here some of my notes in case some body else ever has the same issue. Firstly, in my tests I am using a GXP2000, GXP2020 and a SPA962. I also tested with a SNOM360. my issue was the MWI seemed to fail after an hour, and I could not find why. So in searching my config, I found that when the phones were resubscribing to MWI and BLF, it was being caught in this condition : if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(4); } else { sl_send_reply("404","Not here"); exit; } } My subscribe code handling was after this block, I noticed the issue from the 404 the grandstream was getting when trying to subscribe. I therefore added handling for the subscribe packets in this section, and my MWI and BLF seems to be working. else if ( is_method("PUBLISH") || is_method("SUBSCRIBE") ) { route(5); exit ; } Where route(5) does the appropriate handling of these two types of SIP packets. It should be noted, when I speak of MWI I am referring to presence message-summary which is sent via a publish by my media application. When I write BLF, I am referring to Presence Dialog. So when my phones were resubscribing, they are being sent a 404. I tested the Grandstream and the Linksys, and both seem to have the problem resolved. I will test this with the Snom tomorrow and post my results. David

3 4

[Kamailio-Users] Somebody going to Astricon?
by Klaus Darilion 08 Oct '09

08 Oct '09

..so we should hang out for a beer :-)

4 3

[Kamailio-Users] help with custom module
by liran tal 07 Oct '09

07 Oct '09

Hey list, I've written a very simple module, just exporting one function I need which does something simple like checking a file and returns -1 or 1 if the file exist or not. The problem I'm facing is as follows: During call flow (extension1 calling extension2), it seems that the logic of the call flow is passed via that module and hangs the entire call process. And this happens without even calling that single exported function, but rather just by including the module itself via loadmodule. Log from the call flow: ... 8(3827) SIP Reply (status): 8(3827) version: <SIP/2.0> 8(3827) status: <100> 8(3827) reason: <Trying> 8(3827) parse_headers: flags=2 8(3827) Found param type 232, <branch> = <z9hG4bKde5a.32fd75d7.0>; state=9 8(3827) parse_via: next_via 8(3827) Found param type 232, <branch> = <z9hG4bK-d8754z-303e27391e65ac7d-1---d8754z->; state=6 8(3827) Found param type 235, <rport> = <21644>; state=16 8(3827) end of header reached, state=5 8(3827) parse_headers: Via found, flags=2 8(3827) parse_headers: this is the first via 8(3827) After parse_msg... * 8(3827) forward_reply: found module exec_custom, passing reply to it * 8(3827) DEBUG:destroy_avp_list: destroying list (nil) 8(3827) receive_msg: cleaning up ... that forward_reply thing has the module involved there for whatever reason, I've no idea. This is on openser1.0.1 I'd appreciate some feedback Regards, Liran.

2 1

[Kamailio-Users] PostgreSQL TLS support
by Olle E. Johansson 07 Oct '09

07 Oct '09

The db_postgres documentation doesn't say much about anything, really... Is there support for TLS connections and how do I configure it if it exist? Thanks, /O

2 4

[Kamailio-Users] LDAP and TLS
by Olle E. Johansson 07 Oct '09

07 Oct '09

Greetings, fellow Kamailians! Looking at the LDAP module for Kamailio, I see that we can use LDAPS: and get TLS connections. But where are the SSL configurations so I can use a client cert, or verify the server cert? /O

1 0

[Kamailio-Users] use as registrar only server
by Rajesh 07 Oct '09

07 Oct '09

I would like to know what all can be done using openser/kamailio as a registrar server alone. i.e. in the real world scenario how does openser can be useful as a registrar sever alone.I would like to make myself clear by pointing to the below example. i can integrate openser and asterisk and make openser as a registrar (only)server and can make use of asterisk for call functionality purposes.. Similarly, i would like to know where openser fits in (openser should only be used as a registrar server again). Your guidance is greatly appreciated. -- Regards, Rajesh

1 0

[Kamailio-Users] How can I access dialog attributes from within timeout route?
by Nathan Angelacos 07 Oct '09

07 Oct '09

Hi, In kamailio 1.5.2-notls, I'm trying to get the CallID of a dialog that has timed out: if (is_method( "INVITE" ) && !has_totag()) { $dlg_ctx(timeout_route) = 21; } ... route[21] { xlog ("L_WARN", "Dead dialog: $dlg(callid)"); exit; } However, syslog says: daemon.warn /usr/sbin/kamailio[6578]: Dead Dialog: <null> daemon.warn /usr/sbin/kamailio[6578]: WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID \ '8b2eb5e3-20493c20-14e681d(a)192.168.1.1' and tags '87F489DA-922DF157' '2XrwB5OcEJ2zQvyiTUIh-l2Zd1wjL51M' dlg_handlers.c leads me to believe that the context is loaded before calling the route(21), and the dialog timeout handler is nice enough to report it in syslog just after my script fails. Am I missing something fundamental, or is this just not possible? Thanks!

2 8

[Serusers] SER and RTPproxy
by Jeff Russell 06 Oct '09

06 Oct '09

I Think I have thing configured correctly and yet I can not make things work correctly. This is what I am do I have SER on an internal (10.1.1.x) IP address I have RTPproxy on a different machine with a public IP address 206.75.15.x I am routing the call out over our firewall and things 'almost' work. SER sets up the call using the RTPproxy to set up the ports for the RTP stream, and the RTP stream is returned from the far end to the public IP address, however non of the RTP traffic is sent to the internal address, I am guessing it is waiting for an rtp stream from the internal address but neither SER nor the ip-phone send anything or even seem to know about the RTPproxy. Can anyone offer ANY suggestions as to how to direct the internal RTP stream to the RTPproxy address? Many thanks, Jeffr

2 2

Re: [Serusers] SER and RTPproxy
by Edson - Lists 06 Oct '09

06 Oct '09

Please, don't forget to copy the list... _this_ is a list... ;) Edson. Jeff Russell escreveu: > I think this is where the error is: > > 0(60402) ERROR: extract_body: message body has length zero > 0(60402) ERROR: force_rtp_proxy2: can't extract body from the message > > ideas? > > > > On 2009-10-06, at 11:08 AM, Edson - Lists wrote: > >> I would suggest You to use NGREP and sniff both legs of the >> communication flow... >> >> See what SER receives and responds, what's on the SDP body and look at >> syslog to understand what RTPproxy is doing. Maybe also take a look on >> what the internal client is sending and receiving. >> >> It should give You some hints. >> >> Edson. >> >> Jeff Russell escreveu: >>> I Think I have thing configured correctly and yet I can not make >>> things work correctly. >>> This is what I am do >>> I have SER on an internal (10.1.1.x) IP address >>> I have RTPproxy on a different machine with a public IP address >>> 206.75.15.x >>> I am routing the call out over our firewall and things 'almost' work. >>> SER sets up the call using the RTPproxy to set up the ports for the >>> RTP stream, and the RTP stream is returned from the far end to the >>> public IP address, however non of the RTP traffic is sent to the >>> internal address, I am guessing it is waiting for an rtp stream from >>> the internal address but neither SER nor the ip-phone send anything >>> or even seem to know about the RTPproxy. >>> Can anyone offer ANY suggestions as to how to direct the internal RTP >>> stream to the RTPproxy address? >>> Many thanks, >>> Jeffr >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Kamailio-Users] use of kamailo as a registrar (only)server
by Rajesh 06 Oct '09

06 Oct '09

I could implement openser as a registrar(only) server .Thanks to the team openser for the timely help.I would like to know what all can be done using openser/kamailio as a registrar server alone. i.e. in the real world scenario how does openser can be useful as a registrar sever alone.I would like to make myself clear by pointing to the below example. i can integrate openser and asterisk and make openser as a registrar (only)server and can make use of asterisk for call functionality purposes.. Similarly, i would like to know where openser fits in (openser should only be used as a registrar server again). Your responses can speed up my research work.. > > Saúl Ibarra wrote: > > You are doing save_location before authentication. You should first > challenge the REGISTER for authentication and then do save_location. > > 2009/10/6, Rajesh <Rajesh.sreedharan(a)siptech.com>: > > > Dear All, > I would like my kamailio server to act (only) as a > Registrar server. I have configured my kamilio for performing > authentication successfully, but this time i find that authentication is > not being done.Your views on my cfg is greatly appreciated. > > # ------------------ module loading ---------------------------------- > debug=3 > log_stderror=yes > mpath="/usr/local/lib/kamailio/modules/" > > loadmodule "db_mysql.so" > loadmodule "mi_fifo.so" > loadmodule "sl.so" > loadmodule "pv.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "uri_db.so" > > loadmodule "auth.so" > loadmodule "auth_db.so" > > port=5060 > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- auth_db params ----- > /* uncomment the following lines if you want to enable the DB based > authentication */ > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "load_credentials", "") > > # -- usrloc params -- > > modparam("usrloc", "db_url", > "mysql://openser:openserrw@192.168.64.75/openser") > > > modparam("usrloc", "db_mode", 3) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > # initial sanity checks -- too long messages > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > if (!uri==myself) { > sl_send_reply("400", "Bad destination, only register > messages destined for this server are accepted"); > exit; > }; > > # the only valid message for the registrar is the register message > # other messages are answered with an error message > if (uri==myself) { > > if (method=="REGISTER") { > > sl_send_reply("100", "Trying"); > > if(!save("location")) { > sl_reply_error(); > }; > exit; > } else { > sl_send_reply("403", "Forbidden"); > exit; > }; > }; > > # Authentication route > > > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("192.168.64.75", "subscriber")) > { > www_challenge("192.168.64.75", "0"); > exit; > } > ###the following lines are uncommented > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > else > { > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if (!proxy_authorize("", "subscriber")) > { > proxy_challenge("", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($au!=$tU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > if ($au!=$fU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > > consume_credentials(); > # caller authenticated > } > } > return; > } > > > > -- > Regards, > Rajesh >

1 0

Re: [Kamailio-Users] kamailio as registrar server
by Saúl Ibarra 06 Oct '09

06 Oct '09

This should have gone to the list... :) On Tue, Oct 6, 2009 at 12:00 PM, Rajesh <Rajesh.sreedharan(a)siptech.com> wrote: > I could implement openser as a registrar(only) server .Thanks to the team > openser for the timely help.I would like to know what all can be done using > openser/kamailio as a registrar server alone. i.e. in the real world > scenario how does openser can be useful as a registrar sever alone.I would > like to make myself clear by pointing to the below example. i can integrate > openser and asterisk and make openser as a registrar (only)server and can > make use of asterisk for call functionality purposes.. Similarly, i would > like to know where openser fits in (openser should only be used as a > registrar server again). > Your responses can speed up my research work.. > > > Saúl Ibarra wrote: > > You are doing save_location before authentication. You should first > challenge the REGISTER for authentication and then do save_location. > > 2009/10/6, Rajesh <Rajesh.sreedharan(a)siptech.com>: > > > Dear All, > I would like my kamailio server to act (only) as a > Registrar server. I have configured my kamilio for performing > authentication successfully, but this time i find that authentication is > not being done.Your views on my cfg is greatly appreciated. > > # ------------------ module loading ---------------------------------- > debug=3 > log_stderror=yes > mpath="/usr/local/lib/kamailio/modules/" > > loadmodule "db_mysql.so" > loadmodule "mi_fifo.so" > loadmodule "sl.so" > loadmodule "pv.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "uri_db.so" > > loadmodule "auth.so" > loadmodule "auth_db.so" > > port=5060 > # ----------------- setting module-specific parameters --------------- > > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > > > # ----- auth_db params ----- > /* uncomment the following lines if you want to enable the DB based > authentication */ > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "load_credentials", "") > > # -- usrloc params -- > > modparam("usrloc", "db_url", > "mysql://openser:openserrw@192.168.64.75/openser") > > > modparam("usrloc", "db_mode", 3) > > # ------------------------- request routing logic ------------------- > > # main routing logic > > route{ > > # initial sanity checks -- too long messages > if (msg:len >= 2048 ) { > sl_send_reply("513", "Message too big"); > exit; > }; > > if (!uri==myself) { > sl_send_reply("400", "Bad destination, only register > messages destined for this server are accepted"); > exit; > }; > > # the only valid message for the registrar is the register message > # other messages are answered with an error message > if (uri==myself) { > > if (method=="REGISTER") { > > sl_send_reply("100", "Trying"); > > if(!save("location")) { > sl_reply_error(); > }; > exit; > } else { > sl_send_reply("403", "Forbidden"); > exit; > }; > }; > > # Authentication route > > > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("192.168.64.75", "subscriber")) > { > www_challenge("192.168.64.75", "0"); > exit; > } > ###the following lines are uncommented > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > else > { > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if (!proxy_authorize("", "subscriber")) > { > proxy_challenge("", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($au!=$tU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > if ($au!=$fU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > > consume_credentials(); > # caller authenticated > } > } > return; > } > > > > -- > Regards, > Rajesh > > > > > > -- > Regards, > Rajesh -- /Saúl http://www.saghul.net | http://www.sipdoc.net

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Re: [Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) for-march (mtune)= switch
by Iñaki Baz Castillo 06 Oct '09

06 Oct '09

¿?¿? El Lunes, 5 de Octubre de 2009, ?? escribió: > vendor_id : AuthenticAMD > cpu family : 16 > model : 2 > model name : Quad-Core AMD Opteron(tm) Processor 2350 > stepping : 2 > cpu MHz : 2010.303 > cache size : 512 KB > physical id : 1 > siblings : 4 > core id : 2 > cpu cores : 4 > fpu : yes > fpu_exception : yes > cpuid level : 5 > wp : yes > flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca > cmov pat pse36 clflush mmx fxsr sse sse2 ht syscall nx mmxext fxsr_opt > pdpe1gb rdtscp lm 3dnowext 3dnow pni monitor cx16 popcnt lahf_lm > cmp_legacy svm extapic cr8_legacy altmovcr8 abm sse4a misalignsse > 3dnowprefetch osvw bogomips : 4020.69 > TLB size : 1104 4K pages > clflush size : 64 > cache_alignment : 64 > address sizes : 48 bits physical, 48 bits virtual > power management: ts ttp tm stc 100mhzsteps hwpstate <NULL> > > > above is the info of the CPU. > > ************************************************************* > L Xue > Building 102, Room 102 > Department of Nuclear Physics > Shanghai Institute of Applied Physics, CAS > No. 2019, JiaLuo Roud > Shanghai, 201800 > P. R. China > Email:xueliang@sinap.ac.cn > MSN: xliang1984(a)hotmail.com > ************************************************************** > > > > -----Original Message----- > From: users-bounces(a)lists.kamailio.org on behalf of Iñaki Baz Castillo > Sent: Mon 2009-10-5 23:37 > To: users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) > for-march (mtune)= switch > > El Lunes, 5 de Octubre de 2009, Daniel-Constantin Mierla escribió: > > I am doing MC simulation at STAR. > > Is it a new VoIP service? > -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] kamailio as registrar server
by Rajesh 06 Oct '09

06 Oct '09

Dear All, I would like my kamailio server to act (only) as a Registrar server. I have configured my kamilio for performing authentication successfully, but this time i find that authentication is not being done.Your views on my cfg is greatly appreciated. # ------------------ module loading ---------------------------------- debug=3 log_stderror=yes mpath="/usr/local/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "auth.so" loadmodule "auth_db.so" port=5060 # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # -- usrloc params -- modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") modparam("usrloc", "db_mode", 3) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- too long messages if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!uri==myself) { sl_send_reply("400", "Bad destination, only register messages destined for this server are accepted"); exit; }; # the only valid message for the registrar is the register message # other messages are answered with an error message if (uri==myself) { if (method=="REGISTER") { sl_send_reply("100", "Trying"); if(!save("location")) { sl_reply_error(); }; exit; } else { sl_send_reply("403", "Forbidden"); exit; }; }; # Authentication route if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } ###the following lines are uncommented if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } -- Regards, Rajesh

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[Serusers] Can SER 2.0.0-rc1 pstn.cfg file actually work with BYE from PSTN?
by Frank Durda IV 06 Oct '09

06 Oct '09

Does SER 2.0.0-rc1/examples/pstn.cfg or the much more complete one in Chapter 8 of "SER Getting Started" actually work when it comes to a BYE comming from the PSTN gateway? So far, it appears these two cfg files don't cope with the called party on the PSTN hangs up first, and it isn't clear how it could be made to work in anything more complex than a one-SIP-call-source one pstn-gateway configuration. Let's say I have four clients from which a call can originate and arrive at SER, with the caller source IP addresses of "A", "B", "C" and "D". "C" sends the INVITE of a call that SER determined is destined for the PSTN gateway, which is at IP address "P". Here, the INVITE, 100 Trying, 180/183, 200 OK and ACK messages all go to the right places and look more or less correct. If "C" sends a BYE to end the call, this also works correctly, with SER forwarding that on to "P" and the 200 OK being returned. So far, so good. However, if the called party on the PSTN hangs up first, "P" sends SER a BYE and SER simply sends the BYE right back to "P", whom responds with a "481" back. The originating party at "C" never finds out that the call has ended, because SER didn't pass the BYE on to the calling device. The SIP phone stays in the "Connected" state. In my case, the calling sites A, B, C and D don't include a Record-Route of their own, but sometimes have a Via: header. In either case, SER still doesn't send a BYE or re-INVITE coming from P to the correct destination of A, B, C or D. I don't know how I could force SER to send the BYE or INVITE on to the right place, given four possible call-initiator IP addresses, of which only "C" would know about this sample call. I don't believe I can stick in an extra Record-Route when handling the original INVITE that has the call source IP address in it (in addition to a Record-Route of my own), simply because I don't believe SER will let me dynamically select which senders IP address should go in the forged Record-Route, or if doing so would really help the situation later when that trail of crumbs might be helpful. Also, even though the pstn.cfg sample in the book has lots of steps to check for fraud before processing a re-INVITE coming from the PSTN (and the footnotes say that this is what these checks are for), I don't see how SER would know the right place to send the re-INVITE that came from the PSTN once it the authentication checks were done. It looks like SER will always send it to "P" regardless of which direction it should be sent, which doesn't help much, and puts any call using Session-Expires: to failure, when 50% of that time (plus 32.5 seconds) has passed. So, is there with a corrected pstn.cfg out there, or better still, has anybody actually made these two situations work, which are: (1) The called party on the PSTN hangs-up first which causes a BYE to be sent to SER. SER forwards the BYE to the right caller (or whatever the so-called proxy is supposed to do), AND (2) the SIP-PSTN gateway device sends a re-INVITE to SER and it goes somewhere that will generate an OK reply If someone has made it work, can they show an example, rather than just say "it should be possible" or "works for me"? Such responses are encouraging, but not that helpful. Thanks in advance!

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[Serusers] Is there a way to strip ";cic=nnnn" from INVITES?
by Frank Durda IV 06 Oct '09

06 Oct '09

I am receiving intermittent INVITES from a couple of sources where the INVITE includes a cic=nnnn. I need to strip this off before it reaches the PSTN gateway switch because that switch will try to use that value. However, it appears to be a pseudo-CIC and isn't an assigned carrier number so it can't be used for routing outside the network that stuck it in there. That network should be stripping it off before the call leaves their network, but they are not and the call has passed through a few entities by the time it reaches us, so we don't know whom to contact about the problem. Does SER have any mechanism that will strip this ONE item and leave the rest of the INVITE intact, eg so I can call more than one hard-coded number? The documentation I examined (iptel.org SIP Express Router v0.11.0 -- Admin's Guide, chapter 2.3, page 22) shows functions to delete or replace everything but attributes following a semi-colon and preceding the @. Is there something that is undocumented? Sample: (values altered) INVITE sip:9615551212;cic=9234@208.37.38.39;user=phone SIP/2.0 Need to change that to: INVITE sip:9615551212@208.37.38.39;user=phone SIP/2.0 (and leave any other parameters that might be present intact.) Thanks in advance for suggestions.

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[Kamailio-Users] new developer: Marius Zbihlei
by Henning Westerholt 05 Oct '09

05 Oct '09

Hi all, i'm happy to announce a new developer for the kamailio and sip-router project: Marius Zbihlei. Marius is a experienced C/C++ developer in our offices in Bukarest (Romania). He develops since a few years internal applications for the web hosting plattform of 1&1. Since app. a half year he is part of the team that handles the development and maintenance of our VoIP backend systems. Most of his work so far was targeted to our internal repositories, but Marius also already contributed a bunch of patches to the project. He'll support us with the maintenance of the kamailio code we contributed to the project in the past, and also work on new sip-router features in the future. Best regards, Henning -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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Re: [Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) for -march (mtune)= switch
by Daniel-Constantin Mierla 05 Oct '09

05 Oct '09

Hello, On 05.10.2009 17:29 Uhr, ?? wrote: > > hello， > > > I am doing MC simulation at STAR. I met this problem when I compile. > which mail list should I send? > had no idea what "mc simulation at star" is. Do you use kamailio sip server? http://www.kamailio.org If not, you are wrong here and I do not know what mailing list you should use. Daniel > Thanks. > > > xueliang > > ************************************************************* > L Xue > Building 102, Room 102 > Department of Nuclear Physics > Shanghai Institute of Applied Physics, CAS > No. 2019, JiaLuo Roud > Shanghai, 201800 > P. R. China > Email:xueliang@sinap.ac.cn > MSN: xliang1984(a)hotmail.com > ************************************************************** > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Mon 2009-10-5 16:21 > To: ?? > Cc: Users(a)lists.kamailio.org > Subject: Re: [Kamailio-Users] fs_index.cxx:1: error: bad value > (x86_64) for -march (mtune)= switch > > Hello, > > what application do you compile? Is not kamailio, so you probably sent > to wrong mailing list. > > Cheers, > Daniel > > On 04.10.2009 5:57 Uhr, ?? wrote: > > > > Hi , > > > > > > When I try to compile *.cxx on "Quad-Core AMD Opteron(tm) Processor > 2350" > > the follow happens: > > > > make[1]: Entering directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' > > g++ -lpthread -lrt -Wall -g -I > > /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/include -L > > /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/lib -O3 -Wall -pipe > > -fverbose-asm -march=x86_64 -D_REENTRANT > > -DRTS_DAQMAN=\""172.16.0.1"\" -DTARGET_SYSTEM=\"LINUX\" > > -DPROJDIR=\"/RTScache\" -DINSTALL_SUFFIX=\"LINUX/x86_64\" > > -DRTS_PROJECT_STAR -DRTS_BIG_ENDIAN -DRTS_LOG_COLORED -g -I. -I../ > > -I../../include -I../../trg/include -c -o fs_index.o fs_index.cxx > > fs_index.cxx:1: error: bad value (x86_64) for -march= switch > > fs_index.cxx:1: error: bad value (x86_64) for -mtune= switch > > make[1]: *** [fs_index.o] Error 1 > > make[1]: Leaving directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' > > > > How can I solve that > > Thanks and regards! > > > > > > xueliang > > > > > > ************************************************************* > > L Xue > > Building 102, Room 102 > > Department of Nuclear Physics > > Shanghai Institute of Applied Physics, CAS > > No. 2019, JiaLuo Roud > > Shanghai, 201800 > > P. R. China > > Email:xueliang@sinap.ac.cn > > MSN: xliang1984(a)hotmail.com > > ************************************************************** > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Kamailio (OpenSER) - Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > -- > Daniel-Constantin Mierla > * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin > * http://www.asipto.com/index.php/sip-router-masterclass/ > > > -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

2 1

asking odd mediaproxy,,
by Rakhmadhany Primananda 05 Oct '09

05 Oct '09

I've just setting SIP Express Router and Mediaproxy in 1 PC. Then, I'm setting in NAT Router that only activated for MASQUERADE like this: iptables -A POSTROUTING -s 172.17.2.0/24 -o eth1 -j MASQUERADE NB:I assume in network 10.100.0.0/24 (eth1) as public IP and 172.17.2.0/24as private IP. When I calling with x-lite and activated mediaproxy service, the connection running well with 2 direction (full duplex from 10.x.x.x to 172.x.x.x conversely).But,when I stopping mediaproxy service (for proving that voice packet if pass NAT will connect half duplex or maybe broken),apparently this connection still running well with 2 direction my question,, why this happened? are having trouble with mediaproxy or SIP Express Router? i'm confusing,, thanks very much,, :)

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[Kamailio-Users] Siremis v0.9.4 released
by Elena-Ramona Modroiu 05 Oct '09

05 Oct '09

Hello, Siremis - the web management interface for Kamailio SIP server - has been released with version 0.9.4. This is a patch release fixing the issues reported since v0.9.3 related to db connection samples in default configuration file. More details at: http://siremis.asipto.com Best regards, Ramona Modroiu

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[Kamailio-Users] Functions that don't accept pvars.
by Alex Balashov 05 Oct '09

05 Oct '09

It's very frustrating that some functions don't accept pseudovariables and there's no workaround. For example, in this case I need rtpproxy_offer() and rtpproxy_answer() to accept an IP address argument that is retrieved from a database. However, this won't work. rtpproxy_offer("", $var(src_ip)); # Config compilation error rtpproxy_offeR("", "$var(src_ip)"); # Results in $var(src_ip) # literally being placed in the # SDP body. I understand it is difficult to go back and update all legacy functions to accept PVs everywhere. But isn't it possible to provide a wrapper/compatibility function in the core that will parse a PV and generate as a result something that other functions can see as a string literal? -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] fs_index.cxx:1: error: bad value (x86_64) for -march (mtune)= switch
by ?? 05 Oct '09

05 Oct '09

Hi , When I try to compile *.cxx on "Quad-Core AMD Opteron(tm) Processor 2350" the follow happens: make[1]: Entering directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' g++ -lpthread -lrt -Wall -g -I /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/include -L /afs/rhic//star/ROOT/5.22.00/.sl44_gcc346/rootdeb/lib -O3 -Wall -pipe -fverbose-asm -march=x86_64 -D_REENTRANT -DRTS_DAQMAN=\""172.16.0.1"\" -DTARGET_SYSTEM=\"LINUX\" -DPROJDIR=\"/RTScache\" -DINSTALL_SUFFIX=\"LINUX/x86_64\" -DRTS_PROJECT_STAR -DRTS_BIG_ENDIAN -DRTS_LOG_COLORED -g -I. -I../ -I../../include -I../../trg/include -c -o fs_index.o fs_index.cxx fs_index.cxx:1: error: bad value (x86_64) for -march= switch fs_index.cxx:1: error: bad value (x86_64) for -mtune= switch make[1]: *** [fs_index.o] Error 1 make[1]: Leaving directory `/eliza8/rnc/xueliang/MC/StRoot/RTS/src/SFS' How can I solve that Thanks and regards! xueliang ************************************************************* L Xue Building 102, Room 102 Department of Nuclear Physics Shanghai Institute of Applied Physics, CAS No. 2019, JiaLuo Roud Shanghai, 201800 P. R. China Email:xueliang@sinap.ac.cn MSN: xliang1984(a)hotmail.com **************************************************************

2 1

[Kamailio-Users] days to kamailio 3.0
by Daniel-Constantin Mierla 05 Oct '09

05 Oct '09

Hello, here is the roadmap planned to get to Kamailio 3.0: - this week testing using common git repository - end of this week have Kamailio 3.0 branched - 2 weeks testing of kamailio 3.0 branch and fix of discovered missing features from 1.5, packaging selected modules, name updates, installed tools, documentation updates, etc - 26 or 27 Oct day of release Comments/adjustments are welcome! I also think that we should look at releasing 1.5.3 since there were few important updates since 1.5.2. Cheers, Daniel -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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SER/Kamailio 3.0 by what is kamctl replaced by?
by heinz huber 05 Oct '09

05 Oct '09

Hi, I've checked out the latest head from the git repository, compiled sip-router and installed it. How can i check how many users are registred at the proxy. In openser/kamailio days I used a simple 'kamctl ul show' and options to get some infos about the proxy 'internals'. I've read some lines about a ser_ctl command but can't find it my git working copy. I've followed the instructions on http://sip-router.org/wiki/migration/kamailio-3.0-config. OS: 2.6.18-5-686, debian 5.0.3 thx && br Andy -- Neu: GMX Doppel-FLAT mit Internet-Flatrate + Telefon-Flatrate für nur 19,99 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02

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Re: [Kamailio-Users] content about rtpproxy
by zhangchao00001 04 Oct '09

04 Oct '09

Hello everybody: I start rtpproxy as the root, on the same computer, rtpproxy 1.1 works well while rtpproxy 1.2 dosen't. >On 27.09.2009 2:56 Uhr, zhangchao00001 wrote: >> Hello everyone, i meet a problem when intergrating rtpproxy with >> kamailio. The two software start well, can communicate with each >> other, however, the rtpproxy always report "can't create listener" error. >> Dose anyone meet the same problem? >is the user that starts the rtpproxy has rights to create sockets? Is >there a firewall blocking higher ports? > >Cheers, >Daniel

1 0

[Serusers] simple blind call forwarding needed
by Martin Lane 02 Oct '09

02 Oct '09

SER Users: Thanks in advance for any help! I am running ser 0.9.6 and have a relatively simple cfg file. I have only four (4) digit extensions, no PSTN, no NAT and I do not care about authorization so my config file is basically the Hello World ser.cfg with my system parameters and a few statements in route that allowed me to create standing conference rooms and an echo application with sems. Now I need to implement call forwarding and I have been able to setup the MySQL ser usr_preferences table per chapter 10 of the 'SER - Getting Started' and the 'mySQL newbie? Problems with mySQL and SER?' documents. After adding the call forwarding functionality discussed in chapter 10 inside my config file any number dialed to a SIP device rings busy. All SIP devices register and can call the conference rooms or the echo application but they can not call each other. I will spare you my WireShark logs but can provide if that is needed. I'm looking for trouble shooting suggestions. Is there a way to print to the std i/o from inside the cfg file? This is the first time I needed to handle the INVITE message so I've included my INVITE Message Handler and the Call Forwarding Handler that I added for this effort: ... if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } if (method=="REGISTER") { route(2); break; }; ... route[3] { # ---------------------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------------------- # Note: We are using this fuction only as a hook into the # blind call forwarding feature. Simply want to change the # R-URI and relay the message. if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); # Would love to do a printf here to see if this code is being hit!!! # Wireshark shows INVITE messages are being sent. # debug/printf("\n\n****** Inside route(3) ********\n\n"); # At this point the a blind call forwarding record was found and the # new destination was written in the R-URI. # DEBUG: try just calling route(1). # route(1); # Send to Call Forwarding Handler route(6); break; }; route(1); } route[6] { # ---------------------------------------------------------------------- # Blind Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------------------- lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{4}@") { route(1); break; }; sl_send_reply("404", "User Not Found"); break; }; # DEBUG: There are no alias' and we are not sending calls to other # networks. We may just need to call route(1) and not this # function. route(1); }

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[Serusers] Recent posting status
by ML Gmail 02 Oct '09

02 Oct '09

Hi: I posted a question to the list yesterday. I am a subscriber and the attributes for my profile have me included on all MSG traffic but I have not seen the MSG go out. Can you check on that? Should I resend the question? Thank you. Martin Lane

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Re: [Kamailio-Users] Siremis JSON error
by Elena-Ramona Modroiu 01 Oct '09

01 Oct '09

Luis Guaman wrote: > > Thanks Santiago, now it works great! > > siremis/metadata/Config.xml with changes: > > <?xml version="1.0" standalone="no"?> > > <Config> > > <DataSource> > > <Database Name="Default" Driver="Pdo_Mysql" Server="x.x.x.x" > DBName="openser" User="openser" Password="xxxx" /> > > <Database Name="Siremis" Driver="Pdo_Mysql" Server="x.x.x.x" > DBName="openser" User="openser" Password="xxxx" /> > > </DataSource> > > </Config> > so that was missing in the sample metadata/Config.xml file -- I did a quick 0.9.4 release to fix it. http://siremis.asipto.com/pub/downloads/siremis/siremis-0.9.4.tgz I will do a separate email to announce it. Best regards, Ramona > *De:* Santiago Gimeno [mailto:santiago.gimeno@gmail.com] > *Enviado el:* Jueves, 01 de Octubre de 2009 4:27 > *Para:* Luis Guaman > *CC:* users(a)lists.kamailio.org > *Asunto:* Re: [Kamailio-Users] Siremis JSON error > > Hi Luis, > > I ran into this problem too. In order to solve it I had to add a new > Database element to the siremis/metadata/Config.xml file with the Name > attribute set to Default and pointing to your openser database. I > don't know the reason of having two Databases elements pointing to the > same databases though. > > Best regards, > > Santi > > 2009/9/30 Luis Guaman <luis.guaman(a)interlancompu.com > <mailto:luis.guaman@interlancompu.com>> > > Hi, > > When we tried to run a Siremis Chart got the following error message: > > “Open Flash Chart > > JSON Parse Error [Syntax Error] > > Error at character 0, line 1: > > 0: <body style='font-family:Arial; font-size:12px; > background-color:#FCFCFC;'>Error m” > > Some help?? > > Atentamente, > > Luis Guamán > > Desarrollo de Negocios > > República 396 y Diego de Almagro > > Ph:+593.2.2526585 > > Cel:+593.9.4264354 > > Quito - Ecuador > > logoReducido > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > ------------------------------------------------------------------------ > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users

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[Serusers] blind call forwarding addition
by Martin Lane 01 Oct '09

01 Oct '09

SER Users: Thanks in advance for any help! I am running ser 0.9.6 and have a relatively simple cfg file. I have only four (4) digit extensions, no PSTN, no NAT and I do not care about authorization so my config file is basically the Hello World ser.cfg with my system parameters and a few statements in route that allowed me to create standing conference rooms and an echo application with sems. Now I need to implement call forwarding and I have been able to setup the MySQL ser usr_preferences table per chapter 10 of the 'SER - Getting Started' and the 'mySQL newbie? Problems with mySQL and SER?' documents. After adding the call forwarding functionality discussed in chapter 10 inside my config file any number dialed to a SIP device rings busy. All SIP devices register and can call the conference rooms or the echo application but they can not call each other. I will spare you my WireShark logs but can provide if that is needed. I'm looking for trouble shooting suggestions. Is there a way to print to the std i/o from inside the cfg file? This is the first time I needed to handle the INVITE message so I've included my INVITE Message Handler and the Call Forwarding Handler that I added for this effort: if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } if (method=="REGISTER") { route(2); break; }; ... route[3] { # ---------------------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------------------- # Note: We are using this fuction only as a hook into the # blind call forwarding feature. Simply want to change the # R-URI and relay the message. if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); # Would love to do a printf here to see if this code is being hit!!! # Wireshark show INVITE messages are being sent. # debug/printf("\n\n****** Inside route(3) ********\n\n"); # At this point the a blind call forwarding record was found and the # new destination was written in the R-URI. # DEBUG: try just calling route(1). # route(1); # Send to Call Forwarding Handler route(6); break; }; route(1); } route[6] { # ---------------------------------------------------------------------- # Blind Call Forwarding Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ---------------------------------------------------------------------- lookup("aliases"); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{4}@") { route(1); break; }; sl_send_reply("404", "User Not Found"); break; }; # DEBUG: There are no alias' and we are not sending calls to other # networks. We may just need to call route(1) and not this # function. route(1); }

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Re: [Kamailio-Users] Siremis JSON error
by Santiago Gimeno 01 Oct '09

01 Oct '09

Hi Luis, I ran into this problem too. In order to solve it I had to add a new Database element to the siremis/metadata/Config.xml file with the Name attribute set to Default and pointing to your openser database. I don't know the reason of having two Databases elements pointing to the same databases though. Best regards, Santi 2009/9/30 Luis Guaman <luis.guaman(a)interlancompu.com> > Hi, > > When we tried to run a Siremis Chart got the following error message: > > > > “Open Flash Chart > > > > JSON Parse Error [Syntax Error] > > Error at character 0, line 1: > > > > 0: <body style='font-family:Arial; font-size:12px; > background-color:#FCFCFC;'>Error m” > > > > Some help?? > > > > Atentamente, > > > > Luis Guamán > > Desarrollo de Negocios > > República 396 y Diego de Almagro > > Ph:+593.2.2526585 > > Cel:+593.9.4264354 > > Quito - Ecuador > > [image: logoReducido] > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] Book: SIP Security
by Daniel-Constantin Mierla 01 Oct '09

01 Oct '09

This book might be an interesting reading: http://by-miconda.blogspot.com/2009/09/book-sip-security.html Cheers, Daniel -- Daniel-Constantin Mierla * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin * http://www.asipto.com/index.php/sip-router-masterclass/

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[Serusers] Quick question on pattern matching in the uri variable
by Frank Durda IV 30 Sep '09

30 Sep '09

The following piece of code is supposed to change a uri of 18885551212 to 8885551212 and change a uri of +18885551212 to 8885551212. The 18885551212 situation works fine, but in the +1 situation, the "if" test fails for some reason the code below falls to the "nothing to strip" log message. What am I missing? route[STRIP_LEADING_ONE] { if (method == "INVITE") { if (uri =~ "^sip:1.*" ) { #Leading 1? xlog("L_ERR", "Strip leading '1' - From URI:<%fu> From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); strip(1); } else if (uri =~ "^sip:+1.*" ) { #Leading +1? xlog("L_ERR", "Strip leading '+1' - From URI:<%fu> From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); strip(2); } else { xlog("L_ERR", "Nothing to Strip - From URI:<%fu> >From Tag:<%ft> Request's R-URI:<%ru> Before treatment To URI:<%tu>\n"); } } } Thanks in advance!

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Re: [Kamailio-Users] content about rtpproxy
by zhangchao00001 30 Sep '09

30 Sep '09

Hello everyone, i meet a problem when intergrating rtpproxy with kamailio. The two software start well, can communicate with each other, however, the rtpproxy always report "can't create listener" error. Dose anyone meet the same problem?

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Re: [Kamailio-Users] kamailio 1.4.3 slow
by Juan Perez 30 Sep '09

30 Sep '09

we finally found what was causing the prob, at least the prob is gone after doing this: In the syslog config file of the server where kamailio was working fine we found this entry: local0.none /var/log/messages we added it to the server with probs and problem solved. Since we were using the facility 0 to log the messages from kamailio apparently we need to add that entry listed above as well to make the logging work better. this is how we setup kamailio's logging, using the example from the web: # log messages with LOG_LOCAL0 in /var/log/ser.log local0.* -/var/log/proxy.log I do not know the reason why but it fixed our issue. I wanted to share with you. cheers and thank you a lot for your assistance jp ________________________________ From: Juan Perez <jperezsip2008(a)yahoo.com> To: Henning Westerholt <henning.westerholt(a)1und1.de>; users(a)lists.kamailio.org Cc: myoung(a)redmonsters.net Sent: Thursday, September 17, 2009 8:49:00 AM Subject: Re: [Kamailio-Users] kamailio 1.4.3 slow than you a lot Henning for your suggestions, I will try to do all of that but the lab stuff because we already have a lab setup but it works superb too, only on the new machine is when we have found the prob. in all our other scenarios kamailio runs super fast, processing a lot of calls per seconds and the cpu is 99-100 % idle. fortunately we can do this in the same machine because it has an amount of traffic controllable by us so at any given time we can take it out or put more or use sipp, I will post the results as soon as I have something new, thank you guys a lot again cheers jp ________________________________ From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)lists.kamailio.org Cc: Juan Perez <jperezsip2008(a)yahoo.com>; myoung(a)redmonsters.net Sent: Thursday, September 17, 2009 6:05:07 AM Subject: Re: [Kamailio-Users] kamailio 1.4.3 slow On Mittwoch, 16. September 2009, Juan Perez wrote: > In another scenario in our secondary site I am using another Dell Poweredge > 1950, with dual quad core almost at the same speed and 64 bits , same > version of CentOS and it is working like a charm. A lot of calls and CPU > idle almost 100% all the time. That is we decided to buy a more powerful > server to have a more robust platform in our main site but we haven't been > able to move forward with this because of the bad performance. Any other > ideas guys ? Hello Juan, as this looks more like a hardware issue to me i'd try to find issue that causes this particular server to be that slow. Somewhere it must spend this time. Some ideas: - Start kamailio with only one worker children, and then use strace to see what its doing when its hangs - enable debugging (only for a few seconds if you've production load on the system) and then look to the timestamps where its spend the time - use the benchmark module as Daniel already suggested - try to replicate the setup in a lab environment and then use a load tester (e.g. sipp) to give some traffic to the system. then change certain parts of the configuration and setup to find the culprit Cheers, Henning

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[Kamailio-Users] Siremis JSON error
by Luis Guaman 30 Sep '09

30 Sep '09

Hi, When we tried to run a Siremis Chart got the following error message: Open Flash Chart JSON Parse Error [Syntax Error] Error at character 0, line 1: 0: <body style='font-family:Arial; font-size:12px; background-color:#FCFCFC;'>Error m Some help?? Atentamente, Luis Guamán Desarrollo de Negocios República 396 y Diego de Almagro Ph:+593.2.2526585 Cel:+593.9.4264354 Quito - Ecuador logoReducido

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[Kamailio-Users] Kamailio not starting - no clue in syslog
by Anders 29 Sep '09

29 Sep '09

Hi, I get an error when trying to start Kamailio: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I've searched around and found others with the same problem - but they usually found a lead to the problem in the syslog. All I get in the syslog when trying to start Kamailio is this line: Sep 28 05:34:01 server /USR/SBIN/CRON[10786]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u xxx -pyyy kamailio >/dev/null) Any ideas where and how to look? Thanks!

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[Kamailio-Users] DNS lookup for requests within dialog
by catalina oancea 28 Sep '09

28 Sep '09

Hi all, I tested the following scenario: astreisk kamailio provider(hostname) --------------------> dns lookup --------------------> REGISTER 1 REGISTER 1 <-------------------- <-------------------- 401 401 --------------------> dns lookup --------------------> REGISTER 2 REGISTER 2 (same dialog, increased cseq) It seems that kamailio does dns lookup again when receiving REGISTER 2, although both registers are part of the same dialog. Is this the expected behavior? Thanks, Catalina

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[Kamailio-Users] add_path_received and OBP problems
by Florian Meister 28 Sep '09

28 Sep '09

Hi, I'm trying to set up a HA-OBP for both natted and non-natted users. Atm my setup looks like that: [UAC] <---> (NAT) <---> [LB] <---> [REG + Proxy] <---> [PSTN GW] At the loadbalancer I use add_path and add_path_received to add a path-header to the request sent to the Registrar. (Of course only for register requests). I use a few parameters for telling the registrar module to use the path header. When looking in the mysql-db I can see the correct value in the path-field in the location-table. When getting an invite from the PSTN gateway for one of the UAC's the Proxy is adding a route-header according to the path-value in the mysql. That seems also correct. Now to my problem: The loadbalancer is sending out the INVITE to the private-ip, not to the public-nat-ip of the UAC. It seems that the received-parameter in the Route-header is somehow ignored. I am using the use_received parameter in the loadbalancer, but it does not work. In the following you can find the SIP trace (made on the loadbalancer) and the loadbalancer config, since I think there is the problem. Thanks, Florian Here is the SIP-trace of the INVITE: 09:49:07.785294 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 1101 E..i..@.?.Yb..,F..,+.....U.BINVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 69 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [PSTN Gateway] t=0 0 m=audio 19178 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 09:49:07.786459 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 445 E.....@.@.Z...,+..,F........SIP/2.0 100 Giving a try Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:07.786608 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:08.258116 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:09.258109 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:11.258107 IP [Loadbalancer].5060 > [PRIVATE IP of UAC].5060: SIP, length: 1307 E..7..@.@..@..,+.........#>.INVITE sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Record-Route: <sip:[Loadbalancer];lr;ftag=as33339e2e;nat=yes> Record-Route: <sip:[Proxy+Registrar];lr;ftag=as33339e2e> Via: SIP/2.0/UDP [Loadbalancer];branch=z9hG4bK3d51.207ba0a5.0 Via: SIP/2.0/UDP [Proxy+Registrar];rport=5060;received=[Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 Route: <sip:[Loadbalancer];lr;received="sip:[PUBLIC NAT IP of UAC]:64224"> Max-Forwards: 68 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]> Contact: <sip:4366480588134@[PSTN Gateway]> Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Fri, 25 Sep 2009 07:40:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 330 Path: <sip:[Loadbalancer];lr> v=0 o=root 1940822246 1940822246 IN IP4 [PSTN Gateway] s=Asterisk PBX 1.6.1.5 c=IN IP4 [Loadbalancer] t=0 0 m=audio 35480 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes 09:49:15.023388 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 397 E.....@.?.\"..,F..,+......a.CANCEL sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]> CSeq: 102 CANCEL Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 09:49:15.024138 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 413 E.....@.@.[...,+..,F........SIP/2.0 200 canceling Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 CSeq: 102 CANCEL Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:15.024576 IP [Loadbalancer].5060 > [Proxy+Registrar].5060: SIP, length: 493 E.. ..@.@.Z...,+..,F........SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0;rport=5060;received=[Proxy+Registrar] Via: SIP/2.0/UDP [PSTN Gateway]:5060;branch=z9hG4bK521a62f5;rport=5060 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] CSeq: 102 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 09:49:15.025180 IP [Proxy+Registrar].5060 > [Loadbalancer].5060: SIP, length: 433 E.....@.?.[...,F..,+.......MACK sip:435572949012@[PRIVATE IP of UAC];user=phone SIP/2.0 Via: SIP/2.0/UDP [Proxy+Registrar];branch=z9hG4bK3d51.bf3830b5.0 From: "4366480588134" <sip:4366480588134@[PSTN Gateway]>;tag=as33339e2e Call-ID: 067ebe30188025880d0c9f6e78c1b86d@[PSTN Gateway] To: <sip:+435572949012@[Proxy+Registrar]>;tag=77b8b5897dbef36847302656cec9825e-db87 CSeq: 102 ACK Max-Forwards: 70 User-Agent: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 ----------------This is the config of my loadbalancer--------------------: ####### Global Parameters ######### debug=100 log_stderror=yes log_facility=LOG_LOCAL0 fork=no children=4 disable_tcp=yes port=5060 listen=[Loadbalancer] ####### Modules Section ######## mpath="/usr/lib/kamailio/modules/" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "nathelper.so" loadmodule "dispatcher.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs #modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #modparam("rr", "append_fromtag", 0) # -- nathelper modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy/rtpproxy.sock") modparam("nathelper", "received_avp", "$avp(i:80)") # -- dispatcher params -- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") # -- path params -- modparam("path", "use_received", 1) ####### Routing Logic ######## # main request routing logic route{ xlog("L_ALERT", "[$rm] from [$fu] to [$tu]\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { xlog("L_ALERT", "_____HASTOTAG [$rm] from [$fu] to [$tu]\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_ALERT", "_____LOOSE_ROUTE [$rm] from [$fu] to [$tu]\n"); t_relay(); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("L_ALERT", "_____NON-LOOSE, STATEFUL-ACK [$rm] from [$fu] to [$tu]\n"); # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { xlog("L_ALERT", "_____ACK_WITHOUT_TRANSACTION [$rm] from [$fu] to [$tu]\n"); # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing if (!is_method("REGISTER|MESSAGE")) xlog("L_ALERT", "_____RECORD_ROUTING [$rm] from [$fu] to [$tu]\n"); record_route(); #if ($rU==NULL) { # # request with no Username in RURI # sl_send_reply("484","Address Incomplete"); # exit; #} if (check_route_param("nat=yes")) { setbflag(6); } #if (isflagset(5) || isbflagset(6)) { route(5); #} /* example how to enable some additional event routes */ if (is_method("INVITE")) { t_on_reply("1"); t_on_failure("1"); } if(!ds_is_from_list()) { ds_select_domain("0", "0"); } if (!t_relay()) { sl_reply_error(); } } # Caller NAT detection route route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { add_path_received(); fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } else { add_path(); } return; } # RTPProxy control route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } onreply_route[1] { xdbg("incoming reply\n"); #if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { #if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { if (is_method("INVITE")) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } }

3 13

Re: [Kamailio-Users] won't start - no error found in syslog
by Florian Meister 28 Sep '09

28 Sep '09

> There's rwx for root on /var/run - and I'm on with root trying to start > it... What about the user/group in the config (kamailio.conf) file ? Regards, Florian

1 0

[Kamailio-Users] won't start - no error found in syslog
by Anders 28 Sep '09

28 Sep '09

Hi, I get an error when trying to start Kamailio: ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I've searched around and found others with the same problem - but they usually found a lead to the problem in the syslog. All I get in the syslog when trying to start Kamailio is this line: Sep 28 05:34:01 server /USR/SBIN/CRON[10786]: (root) CMD (echo "call kamailio_cdrs();"|/usr/bin/mysql -u xxx -pyyy kamailio >/dev/null) Any ideas where and how to look? Thanks!

2 1

Re: [Kamailio-Users] content about kamailio start error
by zhangchao00001 27 Sep '09

27 Sep '09

Dear sir: I get the error during during processing of SIP message. The config file i used is below (the config works well on other kamailio server). By the way , the email system i used dosen't support:all the time please keep CC-ing to mailing list address ormaybe iadd the mailing list address in a wrong place, mails with a cc-ing added will be posted back. is it CC:users@lists.kamailio.org? debug=3 fork=yes log_stderror=yes port=5060 ####### Modules Section ######## mpath="/usr/local/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "domain.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) modparam("registrar", "method_filtering", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "max_contacts", 10) modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 0) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("domain", "db_url", "mysql://root:123456@localhost/SIP") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://root:123456@localhost/SIP") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://root:123456@localhost/SIP") modparam("auth_db", "load_credentials", "") modparam("domain", "db_url", "mysql://root:123456@localhost/SIP") modparam("domain", "db_mode", 1) modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) modparam("presence|presence_xml", "db_url", "mysql://root:123456@localhost/SIP") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:26.28.29.253:5060") modparam("nathelper", "rtpproxy_sock", "udp:26.28.249.253:7722") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "rtpproxy_tout", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("path", "use_received", 1) ####### Routing Logic ######## route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(msg:len>=2048) { sl_send_reply("513","Message too big"); exit; } route(4); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); if(!uri==myself) { append_hf("P-hint: outbound\r\n"); if(is_method("REGISTER|MESSAGE")) { if(is_method("REGISTER")) { if(!add_path_received()) { sl_send_reply("503","Internal Path Error!"); exit(); } } } else { record_route(); } route(1); } # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) /*if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; }*/ } else { # authenticate if from local subscriber (uncomment to enable auth) /*if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated }*/ } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } if (t_check_status("486|408")) { sethostport("26.28.249.125:5060"); append_branch(); # do not set the missed call flag again t_relay(); } } 在2009-09-25，"Daniel-Constantin Mierla" <miconda(a)gmail.com> 写道： >Hello, > >do keep mailing list CC-ed for every mail, otherwise I will stop replying. > >On 25.09.2009 3:33 Uhr, zhangchao00001 wrote: >> Dear sir: >> thank you for your reply. >> The Kamailio -V result : >> version: kamailio 1.5.2-notls (i386/linux) >> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> svnrevision: unknown >> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >> main.c compiled on 10:23:04 Sep 23 2009 with gcc 4.3.0 >> when there's no user, it reports no error, when a UAC register, it >> reports the error below: >> Sep 25 09:24:21 [3323] ERROR:core:build_req_buf_from_sip_req: out of >> pkg memory >> Sep 25 09:24:21 [3323] ERROR:tm:print_uac_request: no more shm_mem >> Sep 25 09:24:21 [3323] ERROR:tm:t_forward_nonack: failure to add branches >> Sep 25 09:24:21 [3323] ERROR:tm:_reply_light: failed to allocate shmem >> buffer >> Sep 25 09:24:21 [3323] ERROR:tm:relay_reply: no more share memory >> what should i do to fix it? >does it happen after startup, during processing of SIP message? Is a >stress test? What is the list of loaded modules? If you can share the >config, would be good. > >Daniel > >-- >Daniel-Constantin Mierla >* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin >* http://www.asipto.com/index.php/sip-router-masterclass/ > 网易邮箱用户购物独享现金返还网易邮箱用户购物独享现金返还网易邮箱用户购物独享现金返还 "中国制造",讲述中国60年往事 "中国制造",讲述中国60年往事

1 0

[Kamailio-Users] Need Assistance Tracking Down A SIP Signaling Problem
by Geoffrey Mina 25 Sep '09

25 Sep '09

Hello, I am wondering if anyone can help me determine what the problem with my SIP signaling is here. I have attached two Excel files (can provide other formats if someone doesn't have Excel). The environments in both scenarios are almost identical... although I am dealing with two different carriers upstream. In my environment, I have Kamailio (1.5) sitting in front of a multitude of Asterisk machines. I am using the dispatcher module to distribute INVITE requests across the network. I am doing some interop with a new carrier and we are struggling a bit with some looping scenarios. They are sending invites to my Kamailio server, I am forwarding to asterisk. The two XLS files (Success.xls) and (Fail.xls) have sip_trace information for the transactions. As you can see in the Fail.xls, the ACK to the 200 OK gets looped back to the Kamailio server and never ends up reaching the Asterisk machine. In the Success.xls, this doesn't happen. The only thing I can see that's different is the Success carrier appears to be replacing the RURI in the ACK with the value of the Contact header... in the failed one, the RURI of the ACK is the IP of my Kamailio gateway. I have read through the RFC 3261 to try and figure out what is breaking this, but haven't been able to. I basically need to determine if the signalling is correct... what is wrong with my config... and if the signalling is wrong, what the carrier is doing incorrectly. Thanks in advance! Geoff

4 9

[Kamailio-Users] Diameter
by Ghaith ALKAYYEM 24 Sep '09

24 Sep '09

Hello list, Does anybody have any idea or recommendation about integrating diameter with SER? As far as you know the available module is obsolete, i'm trying to implement the module "CDP" available with OpenIMS: http://www.openimscore.org/docs/ser_ims/CDP.html Thank you

2 1

[Kamailio-Users] Unable to lookup host in c= line,
by Tincho ylm 24 Sep '09

24 Sep '09

Hi all! I'm trying to make a realtime environment openser-asterisk. At this moment, everything is working fine but I have problems with VM feature. What I'm doing is: A call B If B is busy and returns 486 Openser send this to Asterisk. failure_route[1] { if(!t_was_cancelled()) { if (t_check_status("487|486")) { rewritehostport("200.xx.xx.xx:5060"); append_branch(); xlog("L_INFO","---> To Voicemail...\n"); setflag(10); route(1); exit; } if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } } Asterisk cant handle it because this error: chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' chan_sip.c:5121 process_sdp: Unable to lookup host in c= line, 'IN IP4 200.xx.xx.xx200.xx.xx.xx' I'm using: Kamailio 1.4.4 Rtpproxy Asterisk 1.4.22 Is it a rtpproxy issue? it can fix it? If I make a ethereal capture, I can see in the MESSAGE BODY this: connection inrfomation (c): IN IP4 200.xx.xx.xx200.xx.xx.xx What can I do? thanks in advance.

2 2

[Serusers] Search trough status line for specific reason phrase
by Josip Djuricic 23 Sep '09

23 Sep '09

Dear, is there any way to search inside Status-Line of response message? I have strange b2bua that always returns SIP 500 just with different reason phrase. I need to match on them so that I can rewrite failure messages. I thought about something like this: failure_route[1] { if (t_check_status("500")) { #here I need to search trough status line for specific reasons, but have no idea how to do that if (search("^Specific message")) { t_reply("480", "Temporarily Unavailable"); break; } } } Thank you J.

1 0

[Kamailio-Users] H323 - SIP Signalling gateway for Kamailio
by General Lee 23 Sep '09

23 Sep '09

Hi, I'm looking for a H323 - SIP signalling gateway which I can use to bridge video calls between my H323 Gatekeeper and Kamailio system. Has anyone done anything like this before? Are you able to suggest any suitable h323 - SIP video gateway solutions I could use? Thanks -- View this message in context: http://www.nabble.com/H323---SIP-Signalling-gateway-for-Kamailio-tp25530990… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 3

[Serusers] SER as a stateless proxy server
by Muhammad Ali 23 Sep '09

23 Sep '09

Hi, How SER can be configured as a stateless proxy server? Best regards M Ali

1 0

[Kamailio-Users] content about kamailio start error
by zhangchao00001 23 Sep '09

23 Sep '09

Hello everyone, i have some questions about kamailio. After i install rtpproxy 0.3version, the kamailio server can't work any more. The following is the error message. Dose any one know the reason? dose the rtpproxy 0.3 change something that kamailio need? Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module registrar Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module textops Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module uri_db Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module siputils Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module xlog Sep 23 11:07:53 [4911] INFO:core:init_mod: initializing module acc Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:53 [4911] INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255 kb Sep 23 11:07:57 [4925] ERROR:core:build_req_buf_from_sip_req: out of pkg memory Sep 23 11:07:57 [4925] ERROR:tm:print_uac_request: no more shm_mem Sep 23 11:07:57 [4925] ERROR:tm:t_forward_nonack: failure to add branches Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer Sep 23 11:07:57 [4925] ERROR:tm:relay_reply: no more share memory Sep 23 11:07:57 [4925] ERROR:tm:_reply_light: failed to allocate shmem buffer

2 1

[Kamailio-Users] Comparing AVPs to user variables
by Alex Balashov 23 Sep '09

23 Sep '09

Is there a known bug in Kamailio 1.4.1 that causes comparisons like this to be inaccurate? get_profile_size("people", "$avp(s:person_id)", "$var(u_call_count)"); ... if($var(u_call_count) >= $avp(s:call_limit)) In this case, $var(u_call_count) was 3 and $avp(s:call_limit) was 26 - I printed both values beforehand. Yet, the comparison was evaluated as true. I worked around the problem by storing the profile size in an AVP as well and comparing them instead, and it seems to have solved the problem: if($avp(s:u_call_count) >= $avp(s:call_limit)) However, I was wondering if this is a known problem in 1.4.1 and possibly 1.4.x. Thanks! -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671

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[Kamailio-Users] Stupid softswitch sending a CANCEL with "To tag" in parallel forking
by Iñaki Baz Castillo 22 Sep '09

22 Sep '09

Hi, I've experimenting an absurd issue when a softswitch calls to Kamailio and receives two 180 (with different To tags) due to parallel forking made by Kamailio. When the softswitch receives the second 180 (with the second To tag) it sends a ridiculous CANCEL with To tag matching the second To tag received (yes, a CANCEL with To tag !!!). Of course Kamailio replies "403 Why do you send a CANCEL with To tag? please read the RFC 3261". Unfortunatelly the softswitch didn't go to the school and ignores the 403 by repeating the CANCEL. And the worst, when *any* of the ringing phones answerw, the softswitch ignores the 200 so the call is never established. IMHO there is nothing I can do to "fix" this stupid issue. I've tried to reply 200 for the CANCEL but the softswitch is not happy (it seems that it expects the 487 from the canceled branch). And of course, if I do "t_relay()" for the CANCEL then Kamailio cancels both branches (the correct behavior). The only hyper exotic "solution" I can imagine is the following: - Kamailio receives the wrong CANCEL with To tag "2". - Reply 200 for the CANCEL to make the softswitch happy. - Reply 487 with To tag "2" to make the softswitch happy (IMHO this is not possible). - Generate a spoofed BYE for the second branch (that with To tag "2"). Of course, this is not possible with dialog module as it doesn't handle parallel forking. Any suggestion? (yes, I would like to kill the softswitch). Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

[Kamailio-Users] new feature: update sip msg content in request route
by Daniel-Constantin Mierla 21 Sep '09

21 Sep '09

Hello, I just committed a new function in modules_k/textops - msg_apply_changes(). Hopefully the name is suggestive enough - the function applies the changes done to the content of sip message - e.g., if you added a new header, once you use the function, that header will become visible in config file. Therefore the function tries to overcome the overhead of looping various messages back to sip server just because modifications were not visible once done. Another benefit is ability to access latest value for a field (being header or body content) in case of need to hashes or signatures for security purposes. The function can be used only in request route for now. Be carefully when using it, since it changes the expectation you had so far of always working on initially received sip message. Once the function is sued, initial content is lost forever. Since it builds a new buffer and re-parses, it is not very recommended to use it extensively. Example of usage: append_hf("My-Header: yes\r\n"); if(msg_apply_changes()) { # msg buffer has a new content if(is_present_hf("My-Header")) { # will get always here } } Testing and feedback is welcome! Cheers, Daniel

2 1

[Kamailio-Users] questions about TLS
by zhangchao00001 21 Sep '09

21 Sep '09

Hello everybody: I've have some questions about tls connection. How to build the tls connection between two proxy? there's just one function is_peer_verified() in tlsops module. Is it that we set the paramer, the proxy will connect by itself or we should build the connection in the config file explicitly? could someone give me a example? Proxy1<----TLS--->Proxy2

2 1

[Kamailio-Users] room does not exists in IMC
by Rajesh 21 Sep '09

21 Sep '09

Dear Experts, I have been trying to configure IMC for quite some time now..when ti try to create the chatroom (using my xlite phone), using the command #create q, i get the following error. *ERROR:imc:imc_handle_message: room [q] does not exist! Sep 17 10:25:36 [2562] ERROR:imc:imc_manager: failed to handle 'message' * Iam posting my cfg below.. Your sujjestion would be of immense help to me.. # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } #if(uri=~"sip:q.*@") if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } } - Regards, Rajesh

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[Kamailio-Users] allow_address and "Error while converting IP address string to in_addr"
by Marcello Lupo 21 Sep '09

21 Sep '09

Hi to all, i'm using openser-1.2.2 and i'm getting the error: "Error while converting IP address string to in_addr" while using the allow_address function. I tried both with: allow_address("$(ru{uri.host})", "$rp") allow_address("$rd", "$rp") I made debugging printng content of both $rd and $(ru{uri.host}) and both return the correct value. Don't know anymore where to search about it. Any help? Thank you, regards, Marcello

2 4

registration error:500 - Server Internal Error
by Rakhmadhany Primananda 19 Sep '09

19 Sep '09

when i create account in x-lite,then i found error code like this: Registration error:500 - Server Internal Error i'm using SIP server with SER 0.9.6. what's the problem with x-lite? thanks,,

1 0

[Kamailio-Users] PCRE / Regex Module
by Brandon Armstead 18 Sep '09

18 Sep '09

Hello, I have the following pcre regex method call: pcre_match("$hdr(From)", "(?i)^(restricted|anonymous|private|unknown|unavailable)") to match calls that are sent without a proper ANI in a case-insensitive manner, however it does not seem to return true for the following example From Header: >From Header: "Unavailable" <sip:Restricted@192.168.47.68<sip%3ARestricted(a)192.168.47.68> >;tag=gK07327e23 Any thoughts / input is appreciated, and as always thank you for your help! Sincerely, Brandon Armstead

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Re: [Kamailio-Users] Kamailio speed dial
by Elena-Ramona Modroiu 17 Sep '09

17 Sep '09

Hi, please do not forget to CC the list. Your xlog statements print $tu - to header uri, not $ru, request-uri, which is the one updated by speed dial. Make the update and if you still get same result either is an error printed by sd_lookup() in log or the table is not populated properly. Send over the result of the mysql query: select * from speed_dial where sd_username='51' Regards, Ramona Mythili D wrote: > Hi, Good morning....I have seen your mail..Thank you very much for > your reply.. > I have followed your suggestions. > > if(uri=~"sip:[0-9]{2}@.*"){ > > xlog("L_INFO","mylog: test1.\n"); > > xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n"); > > #sd_lookup("speed_dial", "sip:$au@$fd"); > > sd_lookup("speed_dial"); > > xlog("L_INFO","mylog: test2.\n"); > > xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n"); > > } > > I called fromm 1002 to 51. > Then i got the output as : > mylog test1. > SIP Request: method [INVITE] from [sip:1002@192.168.64.94 > <mailto:sip%3A1002@192.168.64.94>] to [sip:51@192.168.64.94 > <mailto:sip%3A51@192.168.64.94>] > mylog: test2. > SIP Request: method [INVITE] from [sip:1002@192.168.64.94 > <mailto:sip%3A1002@192.168.64.94>] to [sip:51@192.168.64.94 > <mailto:sip%3A51@192.168.64.94> > > > Here, see the control goes to the function sd_lookup("speed_dial") and > it must take the value 5001 instead of 51 and sholud make call to 5001. > and after mylog:test2 message, it sholud return the message with OK. > But see it is not like that. So, I think, error might be with the > sd_lookup(" "). Please clarify my doubt. > > Thanks.... > Regards, > Mythili. > On Mon, Sep 14, 2009 at 2:27 PM, Elena-Ramona Modroiu > <ramona(a)asipto.com <mailto:ramona@asipto.com>> wrote: > > Hi, > > I see in one config you use $au@$fd for lookup, that is wrong > unless you have the two digits in username of auth header. Better > use sd_lookup() without second parameter. > > Then print $ru in xlog() before and after sd_lookup() -- that will > reveal what is looked up and if the function changed the r-uri. > > Regards, > Ramona > > > Mythili D wrote: > > Hiii...I installed kamailio 1.5.1 and mysql 5.0.37 properly. > Authnetication is working properly. Now i'm going to configure > the feature "speed dial". > I tried with two config files. I'm facing the same problem > with two config files.. > 1st config file > ----------------------------------------------------------- > # > # $Id: speeddial.cfg 4585 2008-08-06 08:20:30Z klaus_darilion $ > # > # sample config script to use speeddial module > # > debug=3 > log_stderror=yes > log_facility=LOG_LOCAL0 > fork=yes > children=4 > # ----------- global configuration parameters > ------------------------ > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > # ------------------ module loading > ---------------------------------- > loadmodule "/usr/local/lib/kamailio/modules/sl.so" > loadmodule "/usr/local/lib/kamailio/modules/tm.so" > loadmodule "/usr/local/lib/kamailio/modules/rr.so" > loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so" > loadmodule "/usr/local/lib/kamailio/modules/usrloc.so" > loadmodule "/usr/local/lib/kamailio/modules/registrar.so" > loadmodule "/usr/local/lib/kamailio/modules/textops.so" > loadmodule "/usr/local/lib/kamailio/modules/db_mysql.so" > loadmodule "/usr/local/lib/kamailio/modules/speeddial.so" > loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so" > loadmodule "/usr/local/lib/kamailio/modules/xlog.so" > # ----------------- setting module-specific parameters > --------------- > modparam("speeddial", "db_url", > "mysql://openser:openserrw@localhost/openser") > modparam("speeddial", "use_domain", 1) > # -- mi_fifo params -- > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > # -- usrloc params -- > modparam("usrloc", "db_mode", 0) > # -- rr params -- > # add value to ;lr param to make some broken UAs happy > modparam("rr", "enable_full_lr", 1) > # ------------------------- request routing logic > ------------------- > # main routing logic > route{ > # initial sanity checks > if (!mf_process_maxfwd_header("10")) > { > sl_send_reply("483","Too Many Hops"); > exit; > }; > if (msg:len >= max_len ) > { > sl_send_reply("513", "Message too big"); > exit; > }; > if (!method=="REGISTER") record_route(); > if (loose_route()) > { > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > }; > if (!uri==myself) > { > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > }; > if (uri==myself) > { > if (method=="REGISTER") > { > save("location"); > exit; > }; > if(uri=~"sip:[0-9]{2}@.*"){ > xlog("L_INFO","mylog: test1.\n"); > sd_lookup("speed_dial"); > #sd_lookup("speed_dial", "sip:$au@$fd"); > xlog("L_INFO","mylog: test2.\n"); > } > lookup("aliases"); > if (!uri==myself) > { > if (!t_relay()) > { > sl_reply_error(); > }; > exit; > }; > if (!lookup("location")) > { > sl_send_reply("404", "Not Found"); > exit; > }; > }; > if (!t_relay()) > { > sl_reply_error(); > }; > } > -------------------------------------------------------------------- > I'm using the table speed_dial. My data in table is > username=5001 > domain=192.168.64.94 > sd_username=51 > sd_domain=192.168.64.94 > I dialled 51 from useragent 1001. the output is not found. > but it is displaying test1 and test2 messages from my cfg file > which i gave, i.e., it is entring into loop and function > sd_lookup("speed_dial"). But the problem with speed_dial i > think so... > Please help me...... > 2nd config file > --------------------------------------------------------------------- > # > # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ > # > # Kamailio (OpenSER) SIP Server - basic configuration script > # - web: http://www.kamailio.org <http://www.kamailio.org/> > # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ > # > # Direct your questions about this file to: > <users(a)lists.kamailio.org <mailto:users@lists.kamailio.org> > <mailto:users@lists.kamailio.org > <mailto:users@lists.kamailio.org>>> > > # > # Refer to the Core CookBook at > http://www.kamailio.org/dokuwiki/doku.php > # for an explanation of possible statements, functions and > parameters. > # > # There are comments showing how to enable different features > in th econfig > # file. Such commented code starts with #X# where X is a > letter to identify > # a feature. Delete entire #X# if you want to enable that > feature. Next are > # sed commands that help you enable such features. > # > # *** To enamble mysql execute: > # sed -i 's///g' kamailio.cfg > # > # *** To enamble authentication execute: > # - enable mysql > # sed -i 's///g' kamailio.cfg > # - add users using 'kamctl' > # > # *** To enamble persistent user location execute: > # - enable mysql > # sed -i 's/#u#//g' kamailio.cfg > # > # *** To enamble presence server execute: > # - enable mysql > # sed -i 's/#p#//g' kamailio.cfg > # > # *** To enamble nat traversal execute: > # sed -i 's/#n#//g' kamailio.cfg > # - install RTPProxy: http://www.rtpproxy.org > <http://www.rtpproxy.org/> > # - start RTPProxy: > # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 > # > # *** To enhance accounting execute: > # - enable mysql > # sed -i 's/#c#//g' kamailio.cfg > # - add following columns to database > # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL > DEFAULT ''; > # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL > DEFAULT ''; > # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL > DEFAULT ''; > # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL > DEFAULT ''; > # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL > DEFAULT ''; > # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT > NULL DEFAULT ''; > # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) > NOT NULL DEFAULT ''; > # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) > NOT NULL DEFAULT ''; > # ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT > NULL DEFAULT ''; > # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) > NOT NULL DEFAULT ''; > # > ####### Global Parameters ######### > debug=3 > log_stderror=yes > log_facility=LOG_LOCAL0 > fork=yes > children=4 > /* uncomment the following lines to enable debugging */ > #debug=6 > #fork=no > #log_stderror=yes > /* uncomment the next line to disable TCP (default on) */ > #disable_tcp=yes > /* uncomment the next line to enable the auto temporary > blacklisting of > not available destinations (default disabled) */ > #disable_dns_blacklist=no > /* uncomment the next line to enable IPv6 lookup after IPv4 dns > lookup failures (default disabled) */ > #dns_try_ipv6=yes > /* uncomment the next line to disable the auto discovery of > local aliases > based on revers DNS on IPs (default on) */ > #auto_aliases=no > /* uncomment the following lines to enable TLS support > (default off) */ > #disable_tls = no > #listen = tls:your_IP:5061 > #tls_verify_server = 1 > #tls_verify_client = 1 > #tls_require_client_certificate = 0 > #tls_method = TLSv1 > #tls_certificate = > "/usr/local/etc/kamailio/tls/user/user-cert.pem" > #tls_private_key = > "/usr/local/etc/kamailio/tls/user/user-privkey.pem" > #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" > port=5070 > /* uncomment and configure the following line if you want > Kamailio to > bind on a specific interface/port/proto (default bind on all > available) */ > #listen=udp:192.168.1.2:5060 <http://192.168.1.2:5060/> > <http://192.168.1.2:5060 <http://192.168.1.2:5060/>> > > ####### Modules Section ######## > #set module path > mpath="/usr/local/lib/kamailio/modules/" > /* uncomment next line for MySQL DB support */ > loadmodule "db_mysql.so" > loadmodule "mi_fifo.so" > loadmodule "sl.so" > loadmodule "tm.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "usrloc.so" > loadmodule "registrar.so" > loadmodule "textops.so" > loadmodule "uri_db.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "acc.so" > loadmodule "speeddial.so" > /* uncomment next lines for MySQL based authentication support > NOTE: a DB (like db_mysql) module must be also loaded */ > loadmodule "auth.so" > loadmodule "auth_db.so" > /* uncomment next line for aliases support > NOTE: a DB (like db_mysql) module must be also loaded */ > #loadmodule "alias_db.so" > /* uncomment next line for multi-domain support > NOTE: a DB (like db_mysql) module must be also loaded > NOTE: be sure and enable multi-domain support in all used modules > (see "multi-module params" section ) */ > #loadmodule "domain.so" > /* uncomment the next two lines for presence server support > NOTE: a DB (like db_mysql) module must be also loaded */ > #p#loadmodule "presence.so" > #p#loadmodule "presence_xml.so" > #n#loadmodule "nathelper.so" > # ----------------- setting module-specific parameters > --------------- > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > # ----- rr params ----- > # add value to ;lr param to cope with most of the UAs > modparam("rr", "enable_full_lr", 1) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > # ----- rr params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via > location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts > per AOR */ > #modparam("registrar", "max_contacts", 10) > # ----- uri_db params ----- > /* by default we disable the DB support in the module as we do > not need it > in this configuration */ > modparam("uri_db", "use_uri_table", 0) > modparam("uri_db", "db_url", "") > # ----- acc params ----- > /* what sepcial events should be accounted ? */ > modparam("acc", "early_media", 1) > modparam("acc", "report_ack", 1) > modparam("acc", "report_cancels", 1) > /* by default ww do not adjust the direct of the sequential > requests. > if you enable this parameter, be sure the enable "append_fromtag" > in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "failed_transaction_flag", 3) > modparam("acc", "log_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > /* uncomment the following lines to enable DB accounting also */ > #c#modparam("acc", "db_flag", 1) > #c#modparam("acc", "db_missed_flag", 2) > #c#modparam("acc", "db_url", > #c# "mysql://openser:openserrw@localhost/openser") > #c#modparam("acc", "db_extra", > #c# > "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > # ----- usrloc params ----- > /* uncomment the following lines if you want to enable DB > persistency > for location entries */ > #u#modparam("usrloc", "db_mode", 0) > #u#modparam("usrloc", "db_url", > #u# "mysql://openser:openserrw@localhost/openser") > # ----- auth_db params ----- > /* uncomment the following lines if you want to enable the DB > based > authentication */ > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "db_url", > "mysql://openser:openserrw@localhost/openser") > modparam("auth_db", "load_credentials", "") > # ----- alias_db params ----- > /* uncomment the following lines if you want to enable the DB > based > aliases */ > #modparam("alias_db", "db_url", > # "mysql://openser:openserrw@localhost/openser") > # ----- domain params ----- > /* uncomment the following lines to enable multi-domain detection > support */ > #modparam("domain", "db_url", > # "mysql://openser:openserrw@localhost/openser") > #modparam("domain", "db_mode", 1) # Use caching > # ----- multi-module params ----- > /* uncomment the following line if you want to enable > multi-domain support > in the modules (dafault off) */ > #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) > # ----- presence params ----- > /* uncomment the following lines if you want to enable presence */ > #p#modparam("presence|presence_xml", "db_url", > #p# "mysql://openser:openserrw@localhost/openser") > #p#modparam("presence_xml", "force_active", 1) > #p#modparam("presence", "server_address", > "sip:192.168.1.2:5060 <http://192.168.1.2:5060/> > <http://192.168.1.2:5060 <http://192.168.1.2:5060/>>") > # -- nathelper > #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722 > <http://127.0.0.1:7722/> <http://127.0.0.1:7722 > <http://127.0.0.1:7722/>>") > > #n#modparam("nathelper", "natping_interval", 30) > #n#modparam("nathelper", "ping_nated_only", 1) > #n#modparam("nathelper", "sipping_bflag", 7) > #n#modparam("nathelper", "sipping_from", > "sip:pinger@kamailio.org <mailto:sip%3Apinger@kamailio.org> > <mailto:sip%3Apinger@kamailio.org > <mailto:sip%253Apinger@kamailio.org>>") > > #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") > #n#modparam("usrloc", "nat_bflag", 6) > ####### Routing Logic ######## > # main request routing logic > route{ > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > # > # NAT detection > route(4); > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if (is_method("BYE")) { > setflag(1); # do accounting ... > setflag(3); # ... even if the transaction fails > } > route(1); > } else { > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(2); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # non loose-route, but stateful ACK; must be an ACK after a > 487 or e.g. 404 from upstream server > t_relay(); > exit; > } else { > # ACK without matching transaction ... ignore and discard.\n"); > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > #initial requests > # CANCEL processing > if (is_method("CANCEL")) > { > if (t_check_trans()) > t_relay(); > exit; > } > t_check_trans(); > # authentication > route(3); > # record routing > if (!is_method("REGISTER|MESSAGE")) > record_route(); > # account only INVITEs > if (is_method("INVITE")) { > setflag(1); # do accounting > } > if (!uri==myself) > /* replace with following line if multi-domain support is used */ > ##if (!is_uri_host_local()) > { > append_hf("P-hint: outbound\r\n"); > # if you have some interdomain connections via TLS > ##if($rd=="tls_domain1.net <http://tls_domain1.net/> > <http://tls_domain1.net <http://tls_domain1.net/>>") { > ## t_relay("tls:domain1.net <http://domain1.net/> > <http://domain1.net <http://domain1.net/>>"); > ## exit; > ##} else if($rd=="tls_domain2.net <http://tls_domain2.net/> > <http://tls_domain2.net <http://tls_domain2.net/>>") { > ## t_relay("tls:domain2.net <http://domain2.net/> > <http://domain2.net <http://domain2.net/>>"); > > ## exit; > ##} > route(1); > } > # requests for my domain > if( is_method("PUBLISH|SUBSCRIBE")) > route(2); > if (is_method("REGISTER")) > { > if (!save("location")) > sl_reply_error(); > exit; > } > if ($rU==NULL) { > # request with no Username in RURI > sl_send_reply("484","Address Incomplete"); > exit; > } > # apply DB based aliases (uncomment to enable) > ##alias_db_lookup("dbaliases"); > if(uri=~"sip:[0-9]{2}@.*"){ > xlog("L_INFO","mylog: test1.\n"); > sd_lookup("speed_dial", "sip:$au@$fd"); > #sd_lookup("speed_dial"); > xlog("L_INFO","mylog: test2.\n"); > } > if (!lookup("location")) { > switch ($retcode) { > case -1: > case -3: > t_newtran(); > t_reply("404", "Not Found"); > exit; > case -2: > sl_send_reply("405", "Method Not Allowed"); > exit; > } > } > # when routing via usrloc, log the missed calls also > setflag(2); > route(1); > } > route[1] { > #n# if (check_route_param("nat=yes")) { > #n# setbflag(6); > #n# } > #n# if (isflagset(5) || isbflagset(6)) { > #n# route(5); > #n# } > /* example how to enable some additional event routes */ > if (is_method("INVITE")) { > #t_on_branch("1"); > t_on_reply("1"); > t_on_failure("1"); > } > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > # Presence route > /* uncomment the whole following route for enabling presence > server */ > route[2] > { > #p# if (!t_newtran()) > #p# { > #p# sl_reply_error(); > #p# exit; > #p# }; > #p# > #p# if(is_method("PUBLISH")) > #p# { > #p# handle_publish(); > #p# t_release(); > #p# } > #p# else > #p# if( is_method("SUBSCRIBE")) > #p# { > #p# handle_subscribe(); > #p# t_release(); > #p# } > #p# exit; > > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==null) > { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > # Authentication route > /* uncomment the whole following route for enabling > authentication */ > route[3] { > if (is_method("REGISTER")) > { > # authenticate the REGISTER requests (uncomment to enable auth) > if (!www_authorize("", "subscriber")) > { > www_challenge("", "0"); > exit; > } > if ($au!=$tU) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > # authenticate if from local subscriber (uncomment to enable auth) > if (from_uri==myself) > { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > exit; > } > if (is_method("PUBLISH")) > { > if ($au!=$tU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } else { > if ($au!=$fU) { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > } > consume_credentials(); > # caller authenticated > } > } > return; > } > # Caller NAT detection route > /* uncomment the whole following route for enabling Caller NAT > Detection */ > route[4]{ > #n# force_rport(); > #n# if (nat_uac_test("19")) { > #n# if (method=="REGISTER") { > #n# fix_nated_register(); > #n# } else { > #n# fix_nated_contact(); > #n# } > #n# setflag(5); > #n# } > return; > } > # RTPProxy control > /* uncomment the whole following route for enabling RTPProxy > Control */ > route[5] { > #n# if (is_method("BYE")) { > #n# unforce_rtp_proxy(); > #n# } else if (is_method("INVITE")){ > #n# force_rtp_proxy(); > #n# } > #n# if (!has_totag()) add_rr_param(";nat=yes"); > return; > } > branch_route[1] { > xdbg("new branch at $ru\n"); > } > onreply_route[1] { > xdbg("incoming reply\n"); > #n# if ((isflagset(5) || isbflagset(6)) && > status=~"(183)|(2[0-9][0-9])") { > #n# force_rtp_proxy(); > #n# } > #n# if (isbflagset(6)) { > #n# fix_nated_contact(); > #n# } > } > failure_route[1] { > #n# if (is_method("INVITE") > #n# && (isbflagset(6) || isflagset(5))) { > #n# unforce_rtp_proxy(); > #n# } > if (t_was_cancelled()) { > exit; > } > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > # uncomment the following lines if you want to redirect the failed > # calls to a different new destination > ##if (t_check_status("486|408")) { > ## sethostport("192.168.2.100:5060 > <http://192.168.2.100:5060/> <http://192.168.2.100:5060 > <http://192.168.2.100:5060/>>"); > > ## append_branch(); > ## # do not set the missed call flag again > ## t_relay(); > ##} > } > ------------------------------------------------------------------- > Here i'm using the table speed_dial. My data in table is > username=5001 > domain=192.168.64.94 > sd_username=51 > sd_domain=192.168.64.94 > I dialled 51 from useragent 1001. the output is not found. > but it is displaying test1 and test2 messages from my cfg file > which i gave, i.e., it is entring into loop and function > sd_lookup("speed_dial"). But the problem with speed_dial i > think so > Problem is same in two files... > Please help me... > Thanks in advance.... > regards, > Mythili. > > ------------------------------------------------------------------------ > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > > >

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Re: [Kamailio-Users] RTPproxy pre-filling callee // media address
by Nick Lemberger 17 Sep '09

17 Sep '09

I thought those flags/ip address settings was the change the SDP part of the SIP message. I tried your suggestion anyway, but it didn't work. What I'm really looking for is how to change what Kamailio is telling RTPProxy via the socket. -Nick >>> On 9/17/2009 at 8:31 AM, in message <a950b2030909170631s24d5dab4pd989c7a2e5846315(a)mail.gmail.com>, Laszlo <laszlo(a)voipfreak.net> wrote: > 2009/9/17 Nick Lemberger <Nick.Lemberger(a)lkfd.net> > >> Sone background: >> I'm trying to use Kamailio as a multi-homed proxy in front of a softswitch, >> one NIC has a public IP, the other a private. The softswitch is in the >> private network and all SIP messaging goes through a 'virtual IP' on the >> softswitch which allows for statefull failover between the softswitch CPUs. >> The media on the other had goes to an address stated in the SDP message. >> >> I don't have any problem with the SIP messaging, it flows as expected. >> Also, the RTP stream from my UAC to Kamailio/RTPproxy is working as >> expected as well (I can record audio on this leg). >> >> There is a problem when RTP proxy sets up the leg on the internal side of >> the network, I see the following in the logs: >> Sep 17 07:56:36 voip.lakefield.net rtpproxy[20388]: [ID 408714 mail.info] >> INFO:handle_command: pre-filling callee's address with >> 192.168.249.40:33020 >> >> I'm wondering what variable Kamailio gets the 192.168.249.40 address from - >> this is the virtual IP for SIP messaging. In the SDP body of those messages >> from the softswitch, I see that it's supposed to use 192.168.249.8 as the >> media IP (and I see that it's setting up the correct port information, >> source port of 33020 in this instance, but with the wrong IP). Where can I >> tell Kamailio what IP should be telling RTPProxy? >> >> Thanks, >> -Nick >> <http://lists.openser-project.org/cgi-bin/mailman/listinfo/users> > > > you can try using force_rtp_proxy with IP and some flags( o and/or c) > > http://www.kamailio.org/docs/modules/1.5.x/nathelper.html#id2468157 > > -Laszlo

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[Kamailio-Users] RTPproxy pre-filling callee // media address
by Nick Lemberger 17 Sep '09

17 Sep '09

Sone background: I'm trying to use Kamailio as a multi-homed proxy in front of a softswitch, one NIC has a public IP, the other a private. The softswitch is in the private network and all SIP messaging goes through a 'virtual IP' on the softswitch which allows for statefull failover between the softswitch CPUs. The media on the other had goes to an address stated in the SDP message. I don't have any problem with the SIP messaging, it flows as expected. Also, the RTP stream from my UAC to Kamailio/RTPproxy is working as expected as well (I can record audio on this leg). There is a problem when RTP proxy sets up the leg on the internal side of the network, I see the following in the logs: Sep 17 07:56:36 voip.lakefield.net rtpproxy[20388]: [ID 408714 mail.info] INFO:handle_command: pre-filling callee's address with 192.168.249.40:33020 I'm wondering what variable Kamailio gets the 192.168.249.40 address from - this is the virtual IP for SIP messaging. In the SDP body of those messages from the softswitch, I see that it's supposed to use 192.168.249.8 as the media IP (and I see that it's setting up the correct port information, source port of 33020 in this instance, but with the wrong IP). Where can I tell Kamailio what IP should be telling RTPProxy? Thanks, -Nick

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[Kamailio-Users] kamailio 1.4.3 slow
by Juan Perez 17 Sep '09

17 Sep '09

hello I installed kamailio in a pc with dual quad core @ 3.3GHz, 4 GB RAM and see that kamailio is very slow answering to INVITEs from the inbound carriers. The DEBUG is set to 3 and the idle CPU lowers only to 87% or 90 %, which is still decent but anyways the performance is very bad. This is the first time that I see this, I have it installed in other machines with less resources, same OS [CentOS 5.2 64 bits] and it is performing very very good. I know that a high debug level is a enemy but in my case is set to 3 like in the other system, and it is using the same configuration file. What other thing may affect the performance? I have made sure that it is not a network issue because the captures are taken in the same machine and I see multiples INVITEs hitting it and no response at all until more than 2 seconds I would appreciate any help or suggestions cheers jp

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[Serusers] ser 0.9.8 bugfix release
by Andrei Pelinescu-Onciul 17 Sep '09

17 Sep '09

There is a new bugfix release for the old 0.9.x ser (if you are using a 2.x version or sip-router, please disregard this message). The release is made from the top of the rel_0_9_0 branch (so if you already have a ser version compiled from the top of rel_0_9_0 in the last 10 month you don't need to upgrade). It contains 3 more fixes when compared to 0.9.7: - tm: fix delete_cell() when the transaction is referenced - core: fix memory leak in subst_user - acc: misspeled function name and wrong check You can either get it from cvs (v_0_9_8 tag or the top of rel_0_9_0 branch) or you can download a tarball with the sources: http://ftp.iptel.org/pub/ser/0.9.8/src/ser-0.9.8_src.tar.gz. There are no binary packages provided, only sources. Upgrading from older 0.9.x versions should only involve re-compiling and re-installing ser. There are no changes in the config file format or in the DB schemes. Andrei

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[Serusers] HowTo add manually ContactURI to registered user
by Evgeny Miloslavsky 17 Sep '09

17 Sep '09

Hi All. I have sip:ABCD@SomeDomain.com user registered successfully at SER. Now I want manually to add another ContactURI to the same user. Is there any possibility to do so? Thanks in advance Regards, Evgeny Miloslavsky Systest Engineer Juniper Networks Solutions Israel LTD. Office: 972-9-9712355 Office: 972-74-7170072

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[Kamailio-Users] sip_resolvehost: NATPR query timed out
by Joao Gomes Pereira 16 Sep '09

16 Sep '09

Hello Im getting this error in my Kamailio-1.5.2: Sep 16 15:43:08 sip kamailio[25298]: ERROR:core:sip_resolvehost: NATPR query timed out, no server could be reached, stop resolving (Connection timed out) Sep 16 15:43:08 sip kamailio[25298]: CRITICAL:core:mk_proxy: could not resolve hostname: "sip.server.pt" Sep 16 15:43:08 sip kamailio[25298]: ERROR:tm:uri2proxy: bad host name in URI <sip:sip.server.pt> Sep 16 15:43:08 sip kamailio[25298]: ERROR:tm:t_forward_nonack: failure to add branches Why cant Kamailio resolve the name? Could this be causing my clients SIP registrations to fail? Thanks regards Joao Pereira

2 4

[Kamailio-Users] avp_db_query to SELECT a DOUBLE colomn
by BERGANZ François 16 Sep '09

16 Sep '09

Hello, I need help. I try to select a DOUBLE(10,4). Kamailio report <null>. Example: avp_db_query("select CREDIT from subscriber where client='$avp(s:679)'", "$avp(i:513)"); $avp(i:513) is null! But in my database, it is 15.0000 Know you how to do ? Cordialement, BERGANZ François cid:image001.gif@01C8F7CD.6BC1D2C0 <http://www.acropolistelecom.net/> http://www.acropolistelecom.net P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] Job Offer: VoIP System Administrator
by Andreas Granig 15 Sep '09

15 Sep '09

Sipwise is currently hiring a VoIP System Administrator for an interesting position based in Vienna/Austria. You have strong skills in Linux system administration (Monitoring and Alerting using SNMP/Nagios/Cacti/MRTG, Scripting in Perl/M4/sh), a deep understanding of highly available system deployments and good knowledge regarding SIP (preferably Kamailio, Sems, Asterisk)? We offer you a challenging position to help our team further improving our Kamailio-based Class5 Softswitches, deploying them at customer sites and supporting our customers (large DSL and Cable Providers throughout Europe) and sales teams with technical details. If you are interested, please send me an email for further information. Best regards, Andreas

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[Kamailio-Users] SIP Router Development Meeting - Oct 2, 2009, Berlin, Germany
by Daniel-Constantin Mierla 15 Sep '09

15 Sep '09

Hello, I want to introduce SIP Router Development Meeting 2009 planned for Friday, October 2 in Berlin, Germany. More details about the event at: http://sip-router.org/2009/09/14/development-meeting-2009/ Shortly, the goal of the meeting is to analyze the evolution so far, discuss the plans for the future and socialize within the community. The integration between SER and Kamailio is 99% done (only seas module needs some work), first major release based on SIP router core is due in about one month: Kamailio 3.0, today being the date of going into testing phase. Participation is free for anybody upon registration via email at: registration(a)lists.sip-router.org Feel free to submit suggestions for discussion topics, presentations and event extensions. A hacking session is already proposed, now just show your interest. For general questions about the meeting (and/or hacking session), please address them to: sr-dev(a)lists.sip-router.org See you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Kamailio speed dial
by Mythili D 14 Sep '09

14 Sep '09

Hiii...I installed kamailio 1.5.1 and mysql 5.0.37 properly. Authnetication is working properly. Now i'm going to configure the feature "speed dial". I tried with two config files. I'm facing the same problem with two config files.. 1st config file ----------------------------------------------------------- # # $Id: speeddial.cfg 4585 2008-08-06 08:20:30Z klaus_darilion $ # # sample config script to use speeddial module # debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 # ----------- global configuration parameters ------------------------ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/kamailio/modules/sl.so" loadmodule "/usr/local/lib/kamailio/modules/tm.so" loadmodule "/usr/local/lib/kamailio/modules/rr.so" loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so" loadmodule "/usr/local/lib/kamailio/modules/usrloc.so" loadmodule "/usr/local/lib/kamailio/modules/registrar.so" loadmodule "/usr/local/lib/kamailio/modules/textops.so" loadmodule "/usr/local/lib/kamailio/modules/db_mysql.so" loadmodule "/usr/local/lib/kamailio/modules/speeddial.so" loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so" loadmodule "/usr/local/lib/kamailio/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("speeddial", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("speeddial", "use_domain", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial"); #sd_lookup("speed_dial", "sip:$au@$fd"); xlog("L_INFO","mylog: test2.\n"); } lookup("aliases"); if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; }; if (!t_relay()) { sl_reply_error(); }; } -------------------------------------------------------------------- I'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so... Please help me...... 2nd config file --------------------------------------------------------------------- # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's///g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's///g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5070 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "speeddial.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 0) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial", "sip:$au@$fd"); #sd_lookup("speed_dial"); xlog("L_INFO","mylog: test2.\n"); } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ------------------------------------------------------------------- Here i'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so Problem is same in two files... Please help me... Thanks in advance.... regards, Mythili.

2 1

[Kamailio-Users] Does forward() DNS SRV failover?
by Arturo Díaz Almagro 14 Sep '09

14 Sep '09

Hello, I am using Kamailio 1.4.1 and I have to relay REGISTER messages to several other proxies (2). In one case I need to forward all request/responses and in the other case I don not need to propagate the responses. In the first case I am using forward() and in the other case t_replicate(). What I have realized from my test is that forward() does not perform DNS SRV failover as t_relay does, but if I use t_relay, the t_replicate does not work. Is that a bug in forward()? Is there any other way to relay messages, in the case explained above? Thanks -- Arturo Díaz

2 1

[Kamailio-Users] passing parameters in cfg for IMC
by Rajesh 14 Sep '09

14 Sep '09

Dear All, Iam trying to configure IMC. I find that different parameters are passed in the sample cfg and in the README. I would like to know if the following two parameters are essential inorder to configure IMC or can i run IMC without passing the two parameters. modparam("imc", "rooms_table", "rooms") modparam("imc", "rooms_table", "members") I would also like to know the use of these parameters. Thanks Rajesh

2 1

[Kamailio-Users] monioring and alert
by ALAEDDINE abbech 11 Sep '09

11 Sep '09

Hello all, I want have an alert when number of simultaneous call in kamailio/openser reach 200call/second. Who have an idea to do this. regards, alasupcom

3 3

[Kamailio-Users] new module: number portability functionality
by Henning Westerholt 11 Sep '09

11 Sep '09

Hi all, i just commited several parts related to number portability functionality to the sr repository. This can be used as a source for routing informations related to the ownership of the number. This functionality consists of several parts. 1. pdb server This server loads serialized routing data from the disk and stores it in memory. It then listens on an UDP Port for requests containing a number and returns ID of the carrier which owns the number. This server uses the same datastructure as the carrierroute module and provides a really good performance. According our experience it usually consumes only a few percent CPU, even you use only one server for your complete callrouting cluster. 2. pdb connector module This module connects the sip-router server to the pdb server. It supports load-balancing and aggressive timeouts. Normally it not need more than a few ms to query the remote server and return the reply to the configuration script. 3. pdb tool (data compiler) This tool provides the functionality to compile the carrier and number informations into the binary data format the pdb server expect. It supports optimizing the generated trie structure, so that for example the complete number to carrier mapping for germany (app. 150 million numbers) don't need more than a few hundred megabytes. You can also combine not interesting carriers in order to save even more space and get better performance. You find the module in the modules/pdb directory, the server and tool is in utils/pdbt. This directory also contains documentation (README for module, utils/pdbt/docs/* for data format and network protocol). We decided to develop a custom solution as none of the alternatives that were evaluated (like enum over DNS, databases e.g. berkeley DB, some commercial products) provides the wanted performance, failure tolerance and storage efficieny. This module has been in production use since some time, so if you find some bugs they are most probably introduced from me because of the sr adaption. The original implementation was done from Hardy Kahl. Best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

1 0

[Kamailio-Users] Kamailio 1.5.1 + WeSip
by Chandrakant Solanki 11 Sep '09

11 Sep '09

Hello I have install kamailio-1.5.1-tls_src now i would like to install WeSip... How can I ... and can I installed WeSip into same PC in which I installed kamailio... -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] error message ...
by Rajesh 11 Sep '09

11 Sep '09

IAm trying to configure IMC in kamailio.I created two tables in the database named, imc_rooms and imc_members.IN my xlite phone i give the command q:sip@192.168.64.75 ('q' is used as the default chatroom name in .cfg). when i give the command , # create room_name, i get the following error, ERROR:imc:imc_handle_message: room [q] does not exist! Sep 9 12:28:25 [2632] ERROR:imc:imc_manager: failed to handle 'message' Kindly guide me.. Thanks

3 6

[Kamailio-Users] kamailio speed dial
by mythili ram 11 Sep '09

11 Sep '09

Hiii...I installed kamailio 1.5.1 and mysql 5.0.37 properly. Authnetication is working properly. Now i'm going to configure the feature "speed dial". I tried with two config files. I'm facing the same problem with two config files.. 1st config file ----------------------------------------------------------- # # $Id: speeddial.cfg 4585 2008-08-06 08:20:30Z klaus_darilion $ # # sample config script to use speeddial module # debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 # ----------- global configuration parameters ------------------------ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/kamailio/modules/sl.so" loadmodule "/usr/local/lib/kamailio/modules/tm.so" loadmodule "/usr/local/lib/kamailio/modules/rr.so" loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so" loadmodule "/usr/local/lib/kamailio/modules/usrloc.so" loadmodule "/usr/local/lib/kamailio/modules/registrar.so" loadmodule "/usr/local/lib/kamailio/modules/textops.so" loadmodule "/usr/local/lib/kamailio/modules/db_mysql.so" loadmodule "/usr/local/lib/kamailio/modules/speeddial.so" loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so" loadmodule "/usr/local/lib/kamailio/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("speeddial", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("speeddial", "use_domain", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial"); #sd_lookup("speed_dial", "sip:$au@$fd"); xlog("L_INFO","mylog: test2.\n"); } lookup("aliases"); if (!uri==myself) { if (!t_relay()) { sl_reply_error(); }; exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; }; if (!t_relay()) { sl_reply_error(); }; } -------------------------------------------------------------------- I'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so... Please help me...... 2nd config file --------------------------------------------------------------------- # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's///g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's///g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5070 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "speeddial.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 0) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if(uri=~"sip:[0-9]{2}@.*"){ xlog("L_INFO","mylog: test1.\n"); sd_lookup("speed_dial", "sip:$au@$fd"); #sd_lookup("speed_dial"); xlog("L_INFO","mylog: test2.\n"); } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ------------------------------------------------------------------- Here i'm using the table speed_dial. My data in table is username=5001 domain=192.168.64.94 sd_username=51 sd_domain=192.168.64.94 I dialled 51 from useragent 1001. the output is not found. but it is displaying test1 and test2 messages from my cfg file which i gave, i.e., it is entring into loop and function sd_lookup("speed_dial"). But the problem with speed_dial i think so Problem is same in two files... Please help me... Thanks in advance.... regards, Mythili. _________________________________________________________________ News, sports, entertainment and fine living…learn the ropes on MSN India http://in.msn.com

1 0

[Kamailio-Users] room does not exits
by Rajesh 11 Sep '09

11 Sep '09

Dear All, I am configurig IMC and i am curently facing an issue. When i send a mesage to the charroom named q (the default nmae is q), i get an error message which says, " ERROR:imc:imc_handle_message: room [q] does not exist! ERROR:imc:imc_manager: failed to handle 'message' " i created two tables in my database.Iam sure that iam doing something silly which is invoking this error message . Kindly guide me with my cfg and let me know if I need to make any changes in my cfg. Iam posting my cfg below. -- ****************************************************************************************************************************************** # # $Id: kamailio.cfg 5881 2009-06-22 09:13:04Z henningw $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's///g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's///g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "imc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ #p#loadmodule "presence.so" #p#loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("acc", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 2) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@localhost/openser") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #p#modparam("presence|presence_xml", "db_url", #p# "mysql://openser:openserrw@localhost/openser") #p#modparam("presence_xml", "force_active", 1) #p#modparam("presence", "server_address", "sip:192.168.1.2:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #-------tm params----- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) #-------imc params----- modparam("imc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") #the below lines are additionally added for imc #modparam("imc", "rooms_table", "imc_rooms") #modparam("imc", "rooms_table", "imc_members") modparam("imc", "hash_size", 8) modparam("imc", "imc_cmd_start_char", "#") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if(uri=~"sip:q.*@") #if(uri=~"sip:q@192.168.64.75") { # IMC - message xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE") #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } Thanks

1 0

[Kamailio-Users] monitoring tool
by Chandrakant Solanki 11 Sep '09

11 Sep '09

Hello I want to know that .... Is there any monitoring tool for kamailio .. which shows user details numbers of user connected @ that time... which automatically updated every 2/5 minutes... I knew about siremis... is there any other tools..??? -- Regards, Chandrakant Solanki

2 1

[Kamailio-Users] Textops / Search / Regex
by Brandon Armstead 10 Sep '09

10 Sep '09

Hello, I currently have the following search method use: search("(restricted|anonymous|Anonymous|Private|private|Unknown|unknown)") unfortunately this is catching cases where there is a Diversion header i.e. Diversion: < sip:1234@someHost > ; reason = unknown So I attempted to do an assertion: search("(restricted|anonymous|Anonymous|Private|private|Unknown|(?<!reason=)unknown)") However it does not seem that Kamailio is completely PCRE compatible? thus server fail.... / parse error Does anyone have a solution so I can keep the same matching, except exclude matching "unknown|Unknown" when reason=unknown, however match it in all other conditions? Thank you for any and all help ahead of time. Sincerely, Brandon

2 2

[Kamailio-Users] tls and notls kamailio
by Mark Ryden 10 Sep '09

10 Sep '09

Hello, I am trying to build kamailio from source. I am a newbie to kamailio. In the download area of 1.5.2, http://www.kamailio.org/pub/kamailio/1.5.2/src/, there are two files: kamailio-1.5.2-notls_src.tar.gz 3.3M kamailio-1.5.2-tls_src.tar.gz 3.4M I have a question: what is the difference, in short, between these two tar balls ? Regards, Mark

2 1

[Serusers] Trouble with Record-Route
by Frank Durda IV 10 Sep '09

10 Sep '09

I have a situation where SER sits between two independent networks, one being an isolated network and the other being the rest of the Internet. The isolated network is also a PSTN gateway with only one switch on that network. SER has two physical interfaces, and uses force_send_socket and rewritehost to make sure things go out the right interfaces. The problem is when the called network hangs up first and generates a BYE* first, if SER didn't add a record-route, the called network switch bypasses SER and attempts to send directly to the calling IP address, which doesn't work because the two networks cannot be reached directly. If I add just one record-route in SER, the called network now knows how to get back to SER to tell it about the BYE, and SER passes that BYE on to the calling network. So far so good. Unfortunately, this also causes the called network to add Record-Route to responses it sends out, like 183/200 with its own IP address. Those reach the SER without problem, but SER doesn't add its own Record-Route to the reply, so when this reaches the calling network, they only have a Record-route that points to an IP address they cannot possibly reach. I tried a number of things to get around this, and all failed. 1. Tried to add my own Record-Route to the reply, but SER doesn't allow this. Won't even start. 2. Tried to remove the Record-Route from the reply, but SER also doesn't allow this. Won't even start. 3. Tried to add both IP addresses for the two interfaces to by using two Record-Routes at the INVITE. This doesn't fix the response messages, and actually causes looping behavior or "I'm terribly sorry" responses. (I tried this with and without the "enable_double_rr" parameter set and it didn't seem to make any difference.) By altering the order of the two Record-Route you are pretty-much guaranteed to confuse one party or the other as each needs the Record-Routes listed in the opposite order. 4. Tried some combinations of the above with loose_route() but the usual outcome is SER sending messages back to itself rather than to the right place and eventually the dreaded "I'm terribly sorry" or "481 Transaction" gets sent out, usually to nobody in particular. So, are there any working examples out there for a SER that has two network interfaces, each in its own isolated network that correctly route messages between the two sides? I need SER to not only add Record-Routes when appropriate but to strip them off or replace them as the replies pass back through and back to the initiator. The network isolation was a security thing, not a NAT thing (although private IP space is in use), but it is not possible for a message from one network to reach the other without passing through SER, and some commercial equipment I am working with ignores the source of a message and only sends the response to wherever the record-route(s) say it should go and if no record-route, it honors the Via, while other gear ignored both and only replies to the source of the message. *The problems affecting called-party BYEs also causes major grief when re-INVITEs occur, which fail to make it to the calling network (or more likely, the 200 OK replies fail to make it back), causing the call to die after 32 seconds. Any examples would be greatly appreciated.

3 2

[Kamailio-Users] One Way Audio problem
by Chandrakant Solanki 10 Sep '09

10 Sep '09

hello I am using kamailio with asterisk as gateway... I found one way audio problem... Please find attached log on pastbin... http://pastebin.com/f25e3f34a Here is my configuration file listen = sipstaging.testing.com port = 5060 mpath = "/lib/kamailio/modules" #children = 8 debug = 7 fork = yes #group = "openser" #user = "openser" #disable_tcp = no #log_facility = LOG_DAEMON log_facility = LOG_LOCAL0 log_stderror = no #tcp_children = 4 #mhomed = no #server_signature = yes #sock_group = "openser" #sock_mode = 0600 #sock_user = "openser" #unix_sock = "/tmp/openser.sock" #unix_sock_children = 1 #reply_to_via = no #sip_warning = no #check_via = no dns = no rev_dns = no #disable_core_dump = no #dns_try_ipv6 = yes #dns_use_search_list = yes #Chandrakant #loadmodule "exec.so" #loadmodule "presence.so" #modparam("presence", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") #modparam("presence", "force_active", 1 ); #modparam("presence", "max_expires", 3600 ) #modparam("presence", "server_address", "sip: voicemailserver.testingpbx.com:5060" ) #loadmodule "presence_mwi.so" loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "pv.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 3) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 35) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) #modparam("tm", "noisy_ctimer", 1) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) modparam("tm", "fr_inv_timer_avp", "$avp(s:callee_fr_inv_timer)") loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") #modparam("mi_fifo", "fifo_mode", 0660) #modparam("mi_fifo", "fifo_group", "openser") #modparam("mi_fifo", "fifo_user", "openser") #modparam("mi_fifo", "reply_dir", "/tmp/") #modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "textops.so" #loadmodule "uri.so" loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "db_mysql.so" modparam("db_mysql", "ping_interval", 300) modparam("db_mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 1) modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") #modparam("uri_db", "db_table", "uri") #modparam("uri_db", "user_column", "username") #modparam("uri_db", "domain_column", "domain") #modparam("uri_db", "uriuser_column", "uri_user") #modparam("uri_db", "subscriber_table", "subscriber") #modparam("uri_db", "subscriber_user_column", "username") #modparam("uri_db", "subscriber_domain_column", "domain") #modparam("uri_db", "use_uri_table", 0) #modparam("uri_db", "use_domain", 1) loadmodule "avpops.so" modparam("avpops", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") loadmodule "lcr.so" modparam("lcr", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr", "gw_uri_avp", "$avp(i:1400)") modparam("lcr", "ruri_user_avp", "$avp(i:1402)") #modparam("lcr", "contact_avp", "$avp(1401)") modparam("lcr", "fr_inv_timer_avp", "$avp(s:fr_inv_timer_avp)") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "$avp(s:rpid)") loadmodule "alias_db.so" modparam("alias_db", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) loadmodule "uac.so" modparam("uac", "rr_store_param", "vsf") modparam("uac", "from_restore_mode", "auto") modparam("uac", "from_passwd", "s1p:Wiz4rd!") loadmodule "acc.so" modparam("acc", "db_url", "mysql:// openser:openserrw@voicemailserver.testingpbx.com/openser") # flag to record to db modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # flag to log to syslog modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) # use extra accounting to record caller and callee username/domain # - take them from From URI and R-URI modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") #loadmodule "nat_traversal.so" ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO","***** IN route[0]\n"); xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); xlog("L_INFO","***** OUT route[0.1]\n"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); xlog("L_INFO","***** OUT route[0.2]\n"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } # allow NOTIFY messages only from the Asterisk server # if (is_method("NOTIFY")) { # if (!t_relay()) { # sl_reply_error(); # exit; # } # } if (is_method("SUBSCRIBE")) { # challenge/response if (!www_authorize("", "subscriber")) { www_challenge("", "0"); xlog("L_INFO","***** OUT route[0.3]\n"); exit; } # notify the Asterisk server to update the UAs MWI. I # believe most UAs that support SUBSCRIBE will # periodically send a SUBSCRIBE message. These cause # the Asterisk server to update their MWI. # xlog("L_INFO","******************************** $ru\n" ); # exec_msg("/usr/bin/mwi/subscribe $ru"); exit; xlog("L_INFO","***** OUT route[0.4]\n"); } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { # if(!has_totag()) # { # # xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sl_send_reply("403", "Initial Loose-Routing Rejected"); # exit; # } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(8); } if(is_method("REGISTER")) { route(7); } if(is_method("INVITE")) { route(9); } if(is_method("CANCEL") || is_method("ACK")) { route(13); } route(14); xlog("L_INFO","***** OUT route[0]\n"); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO","***** IN route[1]\n"); xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_cli)/g"); avp_delete("$avp(s:clir)/g"); xlog("L_INFO","***** OUT route[1]\n"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO","***** IN route[2]\n"); xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:callee_fr_inv_timer)/g"); avp_delete("$avp(s:cfu)/g"); avp_delete("$avp(s:cfc)/g"); xlog("L_INFO","***** OUT route[2]\n"); } ######################################################################## # Request route 'usr-preferences-caller' ######################################################################## route[3] { xlog("L_INFO","***** IN route[3]\n"); route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", "eq/i:1")) { # mark for anonymization setflag(28); } xlog("L_INFO","***** OUT route[3]\n"); } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO","***** IN route[4]\n"); xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); if(is_avp_set("$avp(s:cfu)/s")) { xlog("L_INFO", "Call-forward-unconditional to '$avp(s:cfu)' found - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_uuid)/g"); avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); avp_pushto("$ru", "$avp(s:cfu)"); route(3); route(10); xlog("L_INFO","***** OUT route[4.1]\n"); exit; } if(is_avp_set("$avp(s:ringtimeout)/n")) { xlog("L_INFO", "Setting ring timeout to $avp(s:ringtimeout) secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_copy("$avp(s:ringtimeout)", "$avp(s:callee_fr_inv_timer)/d"); } xlog("L_INFO","***** OUT route[4]\n"); } ######################################################################## # Request route 'clir' ######################################################################## route[5] { xlog("L_INFO","***** IN route[5]\n"); if(isflagset(28) && !isflagset(27)) { setflag(27); xlog("L_INFO", "Anonymize caller - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); if(is_present_hf("Privacy")) { remove_hf("Privacy"); } append_hf("Privacy: id\r\n"); } xlog("L_INFO","***** OUT route[5]\n"); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[6] { xlog("L_INFO","***** IN route[6]\n"); if(isflagset(22)) { unforce_rtp_proxy(); } xlog("L_INFO","***** OUT route[6]\n"); } ######################################################################## # Request route 'base-route-register' ######################################################################## route[7] { xlog("L_INFO","***** IN route[7]\n"); sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); xlog("L_INFO","***** OUT route[7.1]\n"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); xlog("L_INFO","***** OUT route[7.2]\n"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); xlog("L_INFO","***** OUT route[7.3]\n"); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO","***** OUT route[7]\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[8] { xlog("L_INFO","***** IN route[8]\n"); if(is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } if(is_present_hf("Remote-Party-ID")) { remove_hf("Remote-Party-ID"); } if(is_avp_set("$avp(s:caller_cli)/s")) { if(!isflagset(28)) { xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n"); } } route(5); if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } xlog("L_INFO","***** OUT route[8]\n"); exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[9] { xlog("L_INFO","***** IN route[9]\n"); sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid)="0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); $avp(s:ringtimeout)=1000; route(3); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(51); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); xlog("L_INFO","***** OUT route[9.1]\n"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); xlog("L_INFO","***** OUT route[9.2]\n"); exit; } route(3); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(10); } # route(3); # if(nat_uac_test("19")) # { # fix_nated_contact(); # setbflag(6); # } # route(10); xlog("L_INFO","***** OUT route[9]\n"); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[10] { xlog("L_INFO","***** IN route[10]\n"); if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(12); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO", ">>>>>>>>>>>>>>>>>>>>LENGTH - STORED = $var(X) RURI=$rU len=$(rU{s.len}) \n"); if($rU=="*86") { route(12); } else { if($(rU{s.len})<=7) { route(11); } else { route(12); } } } xlog("L_INFO","***** OUT route[10]\n"); exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[11] { xlog("L_INFO","***** IN route[11]\n"); route(4); revert_uri(); seturi("sip:*89@openser.org"); rewritehostport("voicemailserver.testingpbx.com:5060"); route(8); #if(!lookup("location")) #{ # xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sl_send_reply("404", "User Offline"); # route(50); #} #else #{ # xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # #route(8); # route(50); #} xlog("L_INFO","***** OUT route[11]\n"); exit; } ############################## # TEST ROUTE 51 ########################### route[51] { if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(12); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(52); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(12); } exit; } ########################## #TEST 52 ############################## route[52] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(8); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[12] { xlog("L_INFO","***** IN route[12]\n"); if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(8); xlog("L_INFO","***** OUT route[12.1]\n"); exit; } if(!isflagset(23)) { if(uri =~ "^sip:\*86@.*") { xlog("L_INFO", "User accessing Voicemail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); rewritehostport("voicemailserver.testingpbx.com:5060"); route(8); } # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded else if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); xlog("L_INFO","***** OUT route[12.2]\n"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); xlog("L_INFO","***** OUT route[12.3]\n"); exit; } setflag(21); t_on_failure("1"); route(8); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); xlog("L_INFO","***** OUT route[12]\n"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[13] { xlog("L_INFO","***** IN route[13]\n"); t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO","***** OUT route[13.1]\n"); exit; } if(!t_relay()) { sl_reply_error(); } xlog("L_INFO","***** OUT route[13]\n"); exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[14] { xlog("L_INFO","***** IN route[14]\n"); xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); xlog("L_INFO","***** OUT route[14]\n"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[15] { xlog("L_INFO","***** IN route[15]\n"); if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); xlog("L_INFO","***** OUT route[15.1]\n"); exit; } xlog("L_INFO","***** OUT route[15]\n"); } ######################################################################## #Send the call out ######################################################################## route[16] { xlog("L_INFO","***** IN route[16]\n"); #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); }; if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; xlog("L_INFO","***** OUT route[16]\n"); } ######################################################################## # Request route 'Voicemail and Asterisk for further processing' ######################################################################## failure_route[3] { xlog("L_INFO","***** IN failure route[3]\n"); if(!t_was_cancelled()) { revert_uri(); rewritehostport("voicemailserver.testingpbx.com:5060"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "No Answer Sending the call to Voicemail - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(10); route(8); } xlog("L_INFO","***** OUT failure route[3]\n"); } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO","***** IN onreply route[1]\n"); xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_INFO","***** OUT onreply route[1]\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO","***** IN onreply route[2]\n"); xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } xlog("L_INFO","***** OUT onreply route[2]\n"); exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO","***** IN failure route[1]\n"); xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(15); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); xlog("L_INFO","***** OUT failure route[1.1]\n"); exit; } t_on_failure("3"); route(8); xlog("L_INFO","***** OUT failure route[1]\n"); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { xlog("L_INFO","***** IN failure route[2]\n"); if(!t_was_cancelled()) { revert_uri(); seturi("sip:*87@openser.org"); rewritehostport("voicemailserver.testingpbx.com:5060"); append_branch(); xlog("L_INFO","Call to Voicemail"); setflag(10); route(10); } if(t_check_status("422|481|487")) { xlog("L_INFO", "Final reply - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); xlog("L_INFO","***** OUT failure route[2.1]\n"); exit; } t_on_branch("1"); xlog("L_INFO", "Redirect from UAC intercepted - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if($avp(s:cfc) != NULL) { avp_pushto("$ru", "$avp(s:cfc)"); setflag(29); append_branch(); t_on_branch("1"); xlog("L_INFO", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); } route(15); route(6); xlog("L_INFO","***** OUT failure route[2]\n"); } ######################################################################## # Branch route 'cfc-drop-local' ######################################################################## branch_route[1] { xlog("L_INFO","***** IN branch route[1]\n"); if(is_domain_local("$rd")) { xlog("L_INFO", "Dropping local branch - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); drop(); } xlog("L_INFO","***** OUT branch route[1]\n"); } -- Regards, Chandrakant Solanki

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[Kamailio-Users] How Kamailio Generate INVITE
by ALAEDDINE abbech 10 Sep '09

10 Sep '09

Hello all How can i generate an INVITE from Kamailio. I want generate many INVITE using parametre from data base. Who have an idea for this pb. Thanks Regards Aladin

3 2

[Kamailio-Users] kam 1.5 generating 407 on new branch
by Robert McGilvray 10 Sep '09

10 Sep '09

I'm pulling two destinations out of a sql database that I'd like to parallel fork on. I rewrite the main branches RURI to the first result that's returned then append the second result as a branch using assignment to $br. The first call always rings correctly but the second hangs and it looks like it's sending a 407 to my UAC which already authenticated on the initial INVITE. if ($avp(s:alias_counter) == 0) { # Replace the main branch URI with the first result of the DB $ru = "sip:" + $dbr(result=>[0,0]) + "@" + $dbr(result=>[0,1]); } else { # For all other rows append them as branches $br = "sip:" + $dbr(result=>[$avp(s:alias_counter),0]) + "@" + $dbr(result=>[$avp(s:alias_counter),1]); } Sep 9 14:20:36 [12535] DBG:core:parse_msg: SIP Reply (status): Sep 9 14:20:36 [12535] DBG:core:parse_msg: version: <SIP/2.0> Sep 9 14:20:36 [12535] DBG:core:parse_msg: status: <407> Sep 9 14:20:36 [12535] DBG:core:parse_msg: reason: <Proxy Authentication Required> Do I need to bypass my authentication route for new branches or am going about this wrong? In openser 1.3 using enum for forking I don't have to do any special logic for bypassing proxy_auth so I assumed it'd be similar. Thanks Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto.

2 1

Re: [Kamailio-Users] statistics not working
by Henning Westerholt 10 Sep '09

10 Sep '09

On Mittwoch, 9. September 2009, Anders wrote: > Thanks for the VERY quick answer! Hi Anders, np. :) Please always CC to the user list. > server:/# kamctl fifo get_statistics inuse_transactions > database engine 'MYSQL' loaded > Control engine 'FIFO' loaded > entering fifo_cmd get_statistics inuse_transactions > tm:inuse_transactions = 23 > > FIFO command was: > :get_statistics:openser_receiver_1853 > > inuse_transactions > > That looks alright, doesn't it? It looks ok, but perhaps the additional debugging output you got confuse the subsequent commands you want to run? Can you check your kamctlrc file? There is something like this: ## verbose - debug purposes - default '0' # VERBOSE=1 Perhaps it works when you deactivate this setting? Henning

2 2

[Kamailio-Users] statistics not working
by Anders 09 Sep '09

09 Sep '09

Hi list, I'm pretty green, so please bare with me if the questions are common-sense to some of you - I ask here only because I was unable to find or understand the answer elsewhere, like it the very technical documentation or somewhere else on the web. I suspect that something is wrong with the configuration somewhere, because I can't get much out of the statistics. Just look at the responses from these commands: > server:/etc/munin/plugins# kamctl fifo get_statistics active_dialogs | > cut -d" " -f3 'MYSQL' > 'FIFO' > get_statistics > 0 > was:\n:get_statistics:openser_receiver_6342\nactive_dialogs\n\n > server:/etc/munin/plugins# kamctl fifo get_statistics early_dialogs | > cut -d" " -f3 'MYSQL' > 'FIFO' > get_statistics > 0 > was:\n:get_statistics:openser_receiver_6580\nearly_dialogs\n\n > server:/etc/munin/plugins# kamctl fifo get_statistics > inuse_transactions | cut -d" " -f3 'MYSQL' > 'FIFO' > get_statistics > 0 > was:\n:get_statistics:openser_receiver_6593\ninuse_transactions\n\n > server:/etc/munin/plugins# kamctl fifo get_statistics database engine > 'MYSQL' loaded Control engine 'FIFO' loaded entering fifo_cmd > get_statistics 400 Too few or too many arguments FIFO command was: > :get_statistics:openser_receiver_6610 Doesn't look right, does it? Where do you suggest that I start looking? Thanks!

2 1

[Kamailio-Users] Freezing for Kamailio 3.0 - Monday, Sep 14, 2009
by Daniel-Constantin Mierla 09 Sep '09

09 Sep '09

Hello, who ever has brand new features to push into upcoming Kamailio 3.0 has to do it this week. The plan is o freeze the code and start the testing phase at the beginning of next week, with bug hunting as main activity. As usual, work on additional tools and documentation can go on during testing. If discovered, important missing features from kamailio 1.5 will be considered bugs. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] error message with imc
by Rajesh 09 Sep '09

09 Sep '09

Hi , I have beeen configuring imc for quite some time and finally i got some error message that is related to IMC. thanks to you for helping me reach this point. With the error message, i think i should be able to rectify it with your help.My error message reads the following , ERROR:imc:imc_handle_message: room [q] does not exist! ERROR:imc:imc_manager: failed to handle 'message' . kindly guide me how to rectify the error.. -- Regards, Rajesh

3 4

/var/log/syslog in mediaproxy
by Rakhmadhany Primananda 09 Sep '09

09 Sep '09

hi everybody,, I'm using mediaproxy 1.7.2 and ser-0.9.6,, this is my output from syslog after I installed mediaproxy and running the process : root@serbuntu:/# cat /var/log/syslog | grep proxy > Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for remote commands on `10.100.0.45:25060' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 13:38:12 serbuntu mediaproxy[2214]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Traceback (most recent call last): Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File"/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: dispatcher.start() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: mp = MediaproxyCommandHandler() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 13:38:13 serbuntu proxydispatcher[2217]: RuntimeError Sep 3 13:38:13 serbuntu proxydispatcher[2217]: : Sep 3 13:38:13 serbuntu proxydispatcher[2217]: couldn't create server (Address already in use) Sep 3 13:38:13 serbuntu proxydispatcher[2217]: exiting. Sep 3 14:08:01 serbuntu mediaproxy[2214]: received signal 15 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for remote commands on `10.100.0.45:25060' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Traceback (most recent call last): Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: dispatcher.start() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: mp = MediaproxyCommandHandler() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 21:01:19 serbuntu proxydispatcher[7498]: RuntimeError Sep 3 21:01:19 serbuntu proxydispatcher[7498]: : Sep 3 21:01:19 serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) Sep 3 21:01:19 serbuntu proxydispatcher[7498]: exiting. my question: what's the meaning of serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] and serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) ? is it a bug or error? and this is my mediaproxy.ini configuration (mediaproxy and SER running in one machine) : root@serbuntu:/# cat /usr/local/mediaproxy/mediaproxy.ini > ; ; Configuration file for MediaProxy ; [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = 10.100.0.45 allow = 10.100.0.45 proxyIP = 10.100.0.45 portRange = 60000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 [Accounting] dbaccounting = off [Database] user = dbuser password = dbpass host = dbhost database = radius table = radacct thankq,, regards,, dhany

2 1

[Kamailio-Users] what does the following line mean
by Rajesh 08 Sep '09

08 Sep '09

Dear Experts, Iam configuring IMC and i would like to know what does the following line mean. xdbg("script: message from [$fu] chatroom000 [$ru] msg [$rb]\n"); .I get error in this line when i start kamailio. what does $fu ,$ru , $rb mean. Thanks

3 2

[Kamailio-Users] issues with IMC module
by Rajesh 08 Sep '09

08 Sep '09

Dear Experts, I have been trying to configure IMC for quite some time.I would like to know the reason why am i not gettin the module working.I understand that the command to create chatroom is #create roomname, but iam not very sure as to where do i add the create chat room command .Kindly take time to go through my cfg file and tel me where do i need to add the command to create chatroom. debug=4 # debug level (cmd line: -dddddddddd) fork=no # don't fork log_stderror=yes # log to stderr (cmd line: -E) children=0 # number of children check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5066 listen=192.168.64.75 #alias="stefan.lo" # ------------------ module loading ---------------------------------- mpath="/usr/local//lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "textops.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "imc.so" loadmodule "tm.so" loadmodule "xlog.so" loadmodule "registrar.so" loadmodule "usrloc.so" loadmodule "mi_fifo.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- registrar params -- modparam("registrar", "default_expires", 120) # -- registrar params -- modparam("usrloc", "db_mode", 0) # -- imc params -- modparam("imc", "imc_cmd_start_char", "#") modparam("imc|usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; # registrations - if(is_method("REGISTER")) { save("location"); exit; } #imc##if(uri=~"sip:q.*@") if(uri=~"sip:chatroom000@192.168.64.75") { # IMC - message ##imc## xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { ##edited as below for imc xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message not for room, from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; } -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

1 0

[Serusers] STUN server software?
by SIP 07 Sep '09

07 Sep '09

Anyone know any STUN server software that supports XOR_MAPPED_ADDRESS functionality? Open source or commercial is acceptable. N.

2 1

[Kamailio-Users] kamailio 1.4 INVITE authentication bypass
by Asim Riaz 07 Sep '09

07 Sep '09

Hi List, I am using kamailio 1.4 and authenticating INVITE if the source ip address is not in trusted table but one of the IP which is not in the trusted table was able to bypass INVITE authentication, . I don’t have SIP traces saved from the called but when that was happening I could see that the INVITE didn’t have auth credentials but caller was able to bypass authentication and was sending calls to my upstream gateway. Caller’s IP is definitely not in the trusted table, I am just wondering is it something wrong in my script or similar issue has reported before ; Thanks in Advance Asim route[2] { xlog("L_INFO", "[ROUTE-2] Received initial INVITE from $si\n"); setflag(2); setflag(3); if(is_from_local()) { if(!allow_trusted()) { xlog("L_INFO", "[ROUTE-2 !] Issuing proxy challenge\n"); if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "1"); exit; } else if(!check_from()) { xlog("L_INFO", "[ROUTE-2 !] From URI denied\n"); sl_send_reply("403", "Forbidden"); exit; } } else { xlog("L_INFO", "[ROUTE-2 !] From URI domain not local - denied\n"); sl_send_reply("403", "Forbidden"); exit; } } consume_credentials(); xlog("L_INFO", "[ROUTE-2 ->] Authentication credentials valid\n"); if(nat_uac_test("1")) { xlog("L_INFO", "[ROUTE-2 ->] RFC1918 contact found - fixing up\n"); fix_nated_contact(); force_rport(); setbflag(7); } if(nat_uac_test("8") && search("Content-Type: application/sdp")) { xlog("L_INFO", "[ROUTE-2 ->] RFC1918 SDP endpoint found - fixing up\n"); fix_nated_sdp("10"); } # Apply outbound translations and figure out where to route the call. route(4); # this route the calls to upstream gateway. } These messages i was getting in syslog [ROUTE-2] Received initial INVITE from xxx.xxx.xxx.xxx(Caller_IP) ERROR:auth:consume_credentials: no authorized credentials found (error in scripts) [ROUTE-2 ->] Authentication credentials valid [ROUTE-4] Applying outbound translations to: 0022334455 [ROUTE-4 ->] Translated RURI user part to: 22334455 [ROUTE-4 ->] Gateway election: my_upstream_gateway [ROUTE-5] Accounting translation: sip:0022334455@my_upstream_gateway [ROUTE-2 ->] Relaying

3 9

[Kamailio-Users] imc related issues
by Rajesh 07 Sep '09

07 Sep '09

Hi, Iam trying to configure IMC. as i was told that this works fine, on a users system,I would like to know if i need to integrate it to my kamailio.cfg file or "can i use it all alone".. My cfg file is as folllows. debug=5 # debug level (cmd line: -dddddddddd) fork=no # don't fork log_stderror=yes # log to stderr (cmd line: -E) children=0 # number of children check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5063 listen=192.168.64.75 #alias="stefan.lo" # ------------------ module loading ---------------------------------- mpath="/usr/local//lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "textops.so" loadmodule "sl.so" loadmodule "maxfwd.so" loadmodule "imc.so" loadmodule "tm.so" loadmodule "xlog.so" loadmodule "registrar.so" loadmodule "usrloc.so" loadmodule "mi_fifo.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- registrar params -- modparam("registrar", "default_expires", 120) # -- registrar params -- modparam("usrloc", "db_mode", 0) # -- imc params -- modparam("imc", "imc_cmd_start_char", "#") modparam("imc|usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; # registrations - if(is_method("REGISTER")) { save("location"); exit; } #imc##if(uri=~"sip:q.*@") if(uri=~"sip:chatroom000@192.168.64.75") { # IMC - message ##imc## xdbg("script: message from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { ##edited as below for imc xdbg("script: message not for room, from [$fu] r-uri [$ru] msg [$rb]\n"); xdbg("script: message not for room, from [$fu] chatroom000 [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; } -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

1 0

Re: [Kamailio-Users] unknown column 'name' in field list
by Henning Westerholt 07 Sep '09

07 Sep '09

On Montag, 7. September 2009, Rajesh wrote: > > can you do issue a mysql "desc imc_rooms;" on your openser/ kamailio > > DB to check if the table is correctly setup? And do you configured a > > database url for the imc module? > > Thanks Henning for your prompt reply. > > I did give the command desc imc_rooms and found that the table do exist, > indeed i created them manually before setting up imc module.What else > could be the cause of the problem. Hi Rajesh, please always CC to the list. As hinted in some other mail, perhaps the module use not the correct DB URL? I just checked it on my system with 1.5 branch, it loads ok: Sep 7 12:56:14 ca ../kamailio[9312]: INFO:core:init_mod: initializing module imc Sep 7 12:56:14 ca ../kamailio[9312]: DBG:imc:mod_init: db_url=mysql://XXXX:XXXX@localhost/openser/43/0xb7dd4cac Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:find_mod_export: found <db_bind_api> in module db_mysql [../modules/] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_bind_mod: using db bind api for db_mysql Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_do_init: connection 0x816ca10 not found in pool Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: opening connection: mysql://xxxx:xxxx@localhost/openser Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: connection type is Localhost via UNIX socket Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: protocol version is 10 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_new_connection: server version is 5.0.75-0ubuntu10.2 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_new_result: allocate 28 bytes for result set at 0x816c8f8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: 3 columns returned from the query Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_allocate_columns: allocate 12 bytes for result names at 0x816caa8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_allocate_columns: allocate 12 bytes for result types at 0x816c988 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[0] at 0x816cb60 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x816cb60)[0]=[name] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[1] at 0x816cb40 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x816cb40)[1]=[domain] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: use DB_STRING result type Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: allocate 8 bytes for RES_NAMES[2] at 0x816c858 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: RES_NAMES(0x816c858)[2]=[flag] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_get_columns: use DB_INT result type Sep 7 12:56:14 ca ../kamailio[9312]: DBG:db_mysql:db_mysql_convert_rows: no rows returned from the query Sep 7 12:56:14 ca ../kamailio[9312]: INFO:imc:add_from_db: the query returned no result Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing 3 columns Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing RES_NAMES[0] at 0x816cb60 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing RES_NAMES[1] at 0x816cb40 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing RES_NAMES[2] at 0x816c858 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing result names at 0x816caa8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_columns: freeing result types at 0x816c988 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_rows: freeing 0 rows Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:db_free_result: freeing result set at 0x816c8f8 Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:find_cmd_export_t: found <load_tm>(0) in module tm [../modules/] Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:pool_remove: removing connection from the pool Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:init_mod: registering stats for imc Sep 7 12:56:14 ca ../kamailio[9312]: DBG:core:init_mod: register MI for imc Sep 7 12:56:14 ca ../kamailio[9312]: INFO:core:init_mod: initializing module maxfwd [..] I'd suggest that you increase the debug level to '4' in your cfg, and take a look to the output if the DB credentials are correct. Regards, Henning

1 0

[Kamailio-Users] Could some body pls provide me with a sample cfg file for IMC
by Rajesh 07 Sep '09

07 Sep '09

Dear All, I havae been trying to configure imc for quite some time..If any of you have configured IMC successfully, kindly send me the cfg file .Iam pretty sure that it would help me greatly and reduce my time taken to implement the module,IMC.I would appreciate your efforts for sending me the cfg file Thanks in advance!!!

3 2

[Kamailio-Users] unknown column 'name' in field list
by Rajesh 07 Sep '09

07 Sep '09

Dear All, Am configuring imc for kamailio and i would like to know , why am i getting the error on starting kamailio.My error is as follows " ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'name' in 'field list' Sep 7 14:23:25 [4034] ERROR:core:db_do_query: error while submitting query Sep 7 14:23:25 [4034] ERROR:imc:add_from_db: failed to querry table Sep 7 14:23:25 [4034] ERROR:imc:mod_init: failed to get information from db Sep 7 14:23:25 [4034] ERROR:core:init_mod: failed to initialize module imc Sep 7 14:23:25 [4034] ERROR:core:main: error while initializing modules " -- Thanks in advance

2 1

[Kamailio-Users] Siremis Login
by Davis Doe 06 Sep '09

06 Sep '09

Dear alberto, I am newbie to kamailio pabx and followed the steps by step and i have a problem by accessing the web interface. error: "You don't have permission to access /siremis/bin/ on this server." please help me out. davis

2 1

[Serusers] SER doesn't want to restart
by inge 04 Sep '09

04 Sep '09

Hello all, I have done some modification in my ser.cfg. And when I decided to restart, it failed. Here the output from "messages" : Sep 3 06:44:57 solo /usr/local/sbin/ser[23168]: Maxfwd module- initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: permissions - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: AVPops - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_row(): No memory left Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_rows(): Error while converting row #4207 Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_result(): Error while converting rows Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: store_result(): Error while converting result Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: preload_udomain(): Error while doing db_query Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: register_udomain(): Error while preloading domain 'location' Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: domain_fixup(): Error while registering domain Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: ERROR: fix_expr : fix_actions error Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - destroing module I had thinking that my modification were bad, so I leaved them. But SER failed again with the same output. I put some comments before the lines which used "check_ua" module, and then it worked ! I had never experience a problem with "check_ua" before that. Can someone help me to understand why it failed ?

2 2

[Kamailio-Users] NAT kamailio
by san la 04 Sep '09

04 Sep '09

Hi! I want out of Internet and NAT kamailio registration extension from the internet but do not know how. You can help me letters.! can send your configuration file kamailio.cfg to see also. Thank!

2 1

[Kamailio-Users] t_realy in failure_route?
by Ricardo Martinez 03 Sep '09

03 Sep '09

Hello I have a question. I hope that someone could clarify this to me. I have the next situation. I'm using a t_on_failure route along with the LCR module. So I have something like this : failure_route[1] { if (t_was_cancelled()) { xlog("L_INFO", "[$ci] $rm:$ru t_was_cancelled in failure_route\n"); exit; } if ( t_check_status("486") ) { xlog("L_INFO","[$ci] 486 Received\n"); } if (t_check_status("603")) { xlog("L_INFO","[$ci] 603 Received\n"); if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); xlog("L_INFO", "[$ci] No hay mas gateways para $rm:$ou\n"); exit; } else { xlog("L_INFO", "[$ci] $rm cambiando $ou por $ru en failure_route"); } t_on_failure("1"); } t_relay(); } Maybe I'm misunderstanding how the failure_route works but with this configuration I was expecting that If I have a 603 from the callee I will do the "next_gateway" configuration. And If I have a 486 I will put a line in the log and then continue with the t_relay. So.. the 486 will be passed to the caller, but with the final answer from the caller ( ACK ) to the 486.. and I'm getting this error... Sep 2 16:43:41 ERROR:tm:t_forward_nonack: no branch for forwarding Sep 2 16:43:41 ERROR:tm:w_t_relay: t_forward_nonack failed So it seems that tries to forward the ACK with t_relay? Caller Proxy Callee ------> INVITE -----> INVITE <------ 486 -------> ACK <------ 486 -----> (error?) ACK But... When I modified the next lines in the failure_route if ( t_check_status("486") ) { xlog("L_INFO","[$ci] 486 Received\n"); exit; } And add the "exit" parameter, i don't have the above error and the "486" is passed back to the caller as "expected". Can someone explain me this? Thanks in advance.. Regards Ricardo Martinez.-

2 4

[Serusers] Need a way to delete a URI parameter if present from the INVITE
by Frank Durda IV 03 Sep '09

03 Sep '09

A given system is sending me INVITEs of the form: INVITE sip:6663338888@111.222.33.44;cic=9876;rn=6663331212;npdi SIP/2.0 that includes some URI parameters, eg the things after the semi-colon. In this case, the cic=nnnn value is really some internal routing number and not a real PSTN CIC value (like 0288 for AT&T). So when this eventually reaches a switch on the PSTN, that switch sees that someone has already determined the CIC and it attempts to route using what really isn't a CIC at all. I would like to strip this "cic=nnnn" out of the call within SER, and while there are a number of documented rewriteuri functions for altering the URI in SER, there doesn't seem to be anything that can be used to get at URI parameters. For the above, I would like to trim it to: INVITE sip:6663338888@111.222.33.44;rn=6663331212;npdi SIP/2.0 as the rn= and npdi values are sane and can be used by the destination switch. Is there such a beast, or is the only options to add a function/module or use the exec to invoke an external program to do the job? Thanks in advance.

2 3

[Serusers] /var/log/syslog mediaproxy
by Akhmad Dhany 03 Sep '09

03 Sep '09

hi everybody,, this is my output from syslog after I installed mediaproxy and running the process : root@serbuntu:/# cat /var/log/syslog | grep proxy Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Listening for remote commands on `10.100.0.45:25060' Sep 3 13:38:12 serbuntu mediaproxy[2214]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 13:38:12 serbuntu mediaproxy[2214]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 13:38:12 serbuntu proxydispatcher[2217]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Traceback (most recent call last): Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: dispatcher.start() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: mp = MediaproxyCommandHandler() Sep 3 13:38:13 serbuntu proxydispatcher[2217]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 13:38:13 serbuntu proxydispatcher[2217]: Sep 3 13:38:13 serbuntu proxydispatcher[2217]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 13:38:13 serbuntu proxydispatcher[2217]: RuntimeError Sep 3 13:38:13 serbuntu proxydispatcher[2217]: : Sep 3 13:38:13 serbuntu proxydispatcher[2217]: couldn't create server (Address already in use) Sep 3 13:38:13 serbuntu proxydispatcher[2217]: exiting. Sep 3 14:08:01 serbuntu mediaproxy[2214]: received signal 15 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for commands on local socket `/var/run/mediaproxy.sock' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Listening for remote commands on `10.100.0.45:25060' Sep 3 14:09:09 serbuntu mediaproxy[4987]: Remote commands are allowed from: 10.100.0.45, 127.0.0.1 Sep 3 14:09:09 serbuntu mediaproxy[4987]: Using IP address `10.100.0.45' for the RTP/RTCP proxy Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 14:09:10 serbuntu proxydispatcher[4990]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for commands on local socket `/var/run/proxydispatcher.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Listening for mediaproxy media timeout requests on 0.0.0.0:25061 Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Default mediaproxy server is on `/var/run/mediaproxy.sock' Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Traceback (most recent call last): Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/proxydispatcher.py", line 87, in <module> Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: dispatcher.start() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 114, in start Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: mp = MediaproxyCommandHandler() Sep 3 21:01:19 serbuntu proxydispatcher[7498]: File "/usr/local/mediaproxy/modules/dispatcher.py", line 232, in __init__ Sep 3 21:01:19 serbuntu proxydispatcher[7498]: Sep 3 21:01:19 serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] Sep 3 21:01:19 serbuntu proxydispatcher[7498]: RuntimeError Sep 3 21:01:19 serbuntu proxydispatcher[7498]: : Sep 3 21:01:19 serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) Sep 3 21:01:19 serbuntu proxydispatcher[7498]: exiting. my question: what's the meaning of serbuntu proxydispatcher[7498]: raise RuntimeError, "couldn't create server (%s)" % why[1] and serbuntu proxydispatcher[7498]: couldn't create server (Address already in use) ? is it a bug or error? thankq,, regards,, dhany

1 0

[Kamailio-Users] Siremis with postgres
by Karl Heinz Wolf 03 Sep '09

03 Sep '09

I just tried to get Siremis working with postgres. First I had problems because I created the table and columns with uppercase letters and the behaviour of mysql and postgres is different in this aspect so the Siremis queries weren't successful. Now I created the user access table the following way: CREATE TABLE ob_users ( sysid serial NOT NULL, userid varchar(15) NOT NULL, password varchar(15) default NULL, PRIMARY KEY (sysid) ); I also modified Config.xml: Driver="Pdo_Pgsql" instead of Driver="Pdo_Mysql" and Siremis seems to work fine with postgres. cheers, karl heinz

1 0

SER doesn't want to restart
by inge 03 Sep '09

03 Sep '09

Hello all, I have done some modification in my ser.cfg. And when I decided to restart, it failed. Here the output from "messages" : Sep 3 06:44:57 solo /usr/local/sbin/ser[23168]: Maxfwd module- initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: permissions - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default allow file (/usr/local/etc/ser/permissions.allow) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: Default deny file (/usr/local/etc/ser/permissions.deny) not found => empty rule set Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: AVPops - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - initializing Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_row(): No memory left Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_rows(): Error while converting row #4207 Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: convert_result(): Error while converting rows Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: store_result(): Error while converting result Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: preload_udomain(): Error while doing db_query Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: register_udomain(): Error while preloading domain 'location' Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: domain_fixup(): Error while registering domain Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: ERROR: fix_expr : fix_actions error Sep 3 06:44:58 solo /usr/local/sbin/ser[23168]: CHECK_UA - destroing module I had thinking that my modification were bad, so I leaved them. But SER failed again with the same output. I put some comments before the lines which used "check_ua" module, and then it worked ! I had never experience a problem with "check_ua" before that. Can someone help me to understand why it failed ?

1 0

[Kamailio-Users] support for 'include' in cfg file
by Daniel-Constantin Mierla 03 Sep '09

03 Sep '09

Hello, just committed to GIT support for 'include' in config file. Hopefully this helps in nicer structuring and maintenance of your configs. The syntax is simple: include path_to_file The path_to_file can be relative or absolute. If it is not absolute path, first attempt is to locate it relative to current directory, and if fails, relative to directory of the file that includes it. There is no restriction where include can be used or what can contain - any part of config file is ok. There is a limit of maximum 10 includes in depth, otherwise you can use as many includes as you want. Reporting of the cfg file syntax errors prints now the file name for easier troubleshooting. Example: route { ... include /sr/checks.cfg ... } --- /sr/checks.cfg --- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } --- Testing and feedback is much appreciated. Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

5 13

[Kamailio-Users] imc issues
by Rajesh 03 Sep '09

03 Sep '09

On 03.09.2009 12:00 Uhr, Rajesh wrote: Yes i did cretae.. i completed authentication and presence module successfully.. iam caught up with this.. am posting my cfg file below. kindly correct me if iam wrong. ************************************************************************************************************************************************** ####### Global Parameters ######### debug=3 log_stderror=yes log_facility=LOG_LOCAL0 ##fork changed to NO manually from yes fork=yes #fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local//etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local//etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local//etc/kamailio/tls/user/user-calist.pem" port = 5067 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local//lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "usrloc.so" loadmodule "maxfwd.so" loadmodule "imc.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "registrar.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "domain.so" loadmodule "presence.so" loadmodule "presence_xml.so" #n#loadmodule "nathelper.so" ##insert function tested here # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc","db_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") modparam("auth_db", "load_credentials", "") ###registrar params added for imc # -- registrar params -- modparam("registrar", "default_expires", 120) # -- registrar params -- modparam("usrloc", "db_mode", 0) # -- imc params -- modparam("imc", "imc_cmd_start_char", "#") modparam("imc|usrloc", "db_url", "mysql://openser:openserrw@192.168.64.75/openser") ##the following three lines were added seperately for imc modparam("imc", "rooms_table","imc_rooms") modparam("imc", "rooms_table","imc_members") modparam("imc", "hash_size", 8) # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url","mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:192.168.64.75:5060") # -- nathelper #n#modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #n#modparam("nathelper", "natping_interval", 30) #n#modparam("nathelper", "ping_nated_only", 1) #n#modparam("nathelper", "sipping_bflag", 7) #n#modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") #n#modparam("registrar|nathelper", "received_avp", "$avp(i:80)") #n#modparam("usrloc", "nat_bflag", 6) ####### Routing Logic ######## # main request routing logic route{ insert_hf("To-username: test"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } ########The following set of lines were added for implementing imc # registrations - ##the following line has been added for creating chat rooms ##insert_hf (#create chat-000) if(is_method("REGISTER")) { save("location"); exit; } ##if(uri=~"sip:q*@") ## if(uri=~"sip:chat0000@192.168.64.75") ##the below line has been edited by Rajesh. if(uri=~"sip:chat0000@192.168.64.75") { # IMC - message #imc xdbg("script: message from [$fu ] r-uri [$ru ] msg [$rb]\n"); ##the above two lines are edited for imc by Rajesh. xdbg("script: message from [$fu ] chatroom000 [$ru ] msg [$rb]\n"); if(is_method("MESSAGE")) { log("MESSAGE received -> processing with imc\n"); sl_send_reply("200", "ok"); imc_manager(); exit; } else { sl_send_reply("404", "not found"); exit; }; } else { xdbg("script: message not for room, from [$fu ] r-uri [$ru] msg [$rb]\n"); if(!lookup("location")) { sl_send_reply("404", "not found"); exit; } t_relay(); #sl_send_reply("200", "ok"); exit(); }; #######The above lines were added for implementing imc #insert function tested below insert_hf("Test1"); # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { ## insert functioin tested below insert_hf("Test_in_ACK"); if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { #insert_hf("To-username: test"); setflag(1); # do accounting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain insert_hf ("Test2"); if( is_method("PUBLISH|SUBSCRIBE")) route(2); if (is_method("REGISTER")) { if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } ##insert function added below ##insert_hf ("Test4"); route[1] { ##insert function added below insert_hf ("Test5"); #n# if (check_route_param("nat=yes")) { #n# setbflag(6); #n# } #n# if (isflagset(5) || isbflagset(6)) { #n# route(5); #n# } /* example how to enable some additional event routes */ if (is_method("INVITE")) { insert_hf ("Test5"); ##tried insert function here. #t_on_branch("1"); ##tried here insert_hf("To-username: test"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { ##tried here for insert function. #insert_hf("Test_Presence"); #p# if (!t_newtran()) #p# { #p# sl_reply_error(); #p# exit; #p# }; #p# #p# if(is_method("PUBLISH")) #p# { #p# handle_publish(); #p# t_release(); #p# } #p# else #p# if( is_method("SUBSCRIBE")) #p# { #p# handle_subscribe(); #p# t_release(); #p# } #p# exit; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("192.168.64.75", "subscriber")) { www_challenge("192.168.64.75", "0"); exit; } ###the following lines are uncommented if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ #n# force_rport(); #n# if (nat_uac_test("19")) { #n# if (method=="REGISTER") { #n# fix_nated_register(); #n# } else { #n# fix_nated_contact(); #n# } #n# setflag(5); #n# } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { #n# if (is_method("BYE")) { #n# unforce_rtp_proxy(); #n# } else if (is_method("INVITE")){ #n# force_rtp_proxy(); #n# } #n# if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); #n# if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { #n# force_rtp_proxy(); #n# } #n# if (isbflagset(6)) { #n# fix_nated_contact(); #n# } } failure_route[1] { #n# if (is_method("INVITE) #n# && (isbflagset(6) || isflagset(5))) { #n# unforce_rtp_proxy(); #n# } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } ************************************************************************************************************************************************** Daniel-Constantin Mierla wrote: Hello, have you created the db tables: kamdbctl create? Cheers, Daniel On 03.09.2009 11:44 Uhr, Rajesh wrote: Dear Team, Iam configuring imc for kamailio-1.5 and while iam trying to load the module, i get the following error .Kindly guide me with a solution ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'name' in 'field list' Sep 3 12:24:20 [18583] ERROR:core:db_do_query: error while submitting query Sep 3 12:24:20 [18583] ERROR:imc:add_from_db: failed to querry table Sep 3 12:24:20 [18583] ERROR:imc:mod_init: failed to get information from db Sep 3 12:24:20 [18583] ERROR:core:init_mod: failed to initialize module imc Sep 3 12:24:20 [18583] ERROR:core:main: error while initializing modules --

1 0

[Kamailio-Users] DIVERSION or UAC_REDIRECT or BOTH
by Omar Mendoza 03 Sep '09

03 Sep '09

I got confused: Using Kamailio 1.4.3 uac_redirect is been used to handle diversion, also the draft expired there is anew one http://tools.ietf.org/html/draft-levy-sip- diversion-10, is this used together with the diversion module, or no need at all. or is diversion module when you have customers registered in your system? I am do not have phones registered in my system at all, i use mainly the CR module, and i need to allow diversion headers. which one to use, or both?! Also i am scare to use the diversion due the CDRs will create a new record for that, who i will charge the call for?! :) and if i use UAC_REDIRECT should i be able to use the avp $dip and the following, from ROUTE to add to the accounting? Diversion header URI $di - reference to Diversion header URI Diversion "privacy" parameter $dip - reference to Diversion header “privacy” parameter value Diversion "reason" parameter $dir - reference to Diversion header “reason” parameter value __________________________ O the Great

3 4

[Kamailio-Users] problems with imc
by Rajesh 03 Sep '09

03 Sep '09

Dear Team, Iam configuring imc for kamailio-1.5 and while iam trying to load the module, i get the following error .Kindly guide me with a solution ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'name' in 'field list' Sep 3 12:24:20 [18583] ERROR:core:db_do_query: error while submitting query Sep 3 12:24:20 [18583] ERROR:imc:add_from_db: failed to querry table Sep 3 12:24:20 [18583] ERROR:imc:mod_init: failed to get information from db Sep 3 12:24:20 [18583] ERROR:core:init_mod: failed to initialize module imc Sep 3 12:24:20 [18583] ERROR:core:main: error while initializing modules -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

3 2

[Kamailio-Users] Problem configuring imc
by Rajesh 02 Sep '09

02 Sep '09

Deat Team, I would like to know how to embedd the message #create chatroom into the message body of the SIP message. I cameacross a post, wher i was told that instead of SIP,XMPP is better used for the purpose of instant messaging conference.It would be greatly helpful to me, if u can explain it to me -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

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Re: [Kamailio-Users] question about htable size
by jh＠tutpro.com 02 Sep '09

02 Sep '09

Edson Gellert Schubert writes: > http://lists.kamailio.org/pipermail/users/2009-August/024463.html ok, looks like size has the same meaning as in lcr module, i.e., it just affects the size of hash table, not number of entries stored in it. -- juha

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[Kamailio-Users] question about htable size
by jh＠tutpro.com 02 Sep '09

02 Sep '09

htable module readme says abotu htable param: size - number specifying the size of hash table. The number of entries in the table is 2^size does it mean the size of the hash table or how many entries can be stored in hash table? in lcr module, for example, lcr_hash_size param specifies the former and there is no limit on how many entries a hash table can contain: Defines the size of hash table used to store <prefix, from_pattern, priority> tuples. Hashing is done based on prefix. Larger value means less collisions with other prefixes. Hash size value should be a power of 2. it is the same with htable size param? -- juha

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[Kamailio-Users] job offer: VoIP developer, VoIP admin
by Henning Westerholt 02 Sep '09

02 Sep '09

Job offer: Interested in working with one of the biggest Voice over IP networks in europe? We're hiring and have two open positions in Karlsruhe, Germany. For more informations (requirements, how to apply..) please refer to the openings on our job page (in german). Some facts about our VoIP system can be found at e.g. [1], please feel free to contact me per mail as well. VoIP Operation, system administration position: http://www.unitedinternet.de/xml/ui/JobsJobDetail;?__frame=_top&__lf=JobsAp… VoIP Software development, R&D position: http://www.unitedinternet.de/xml/ui/JobsJobDetail;?__frame=_top&__lf=JobsAp… Best regards, Henning Westerholt [1] http://www.kamailio.org/events/2009-linuxtag/presentation.pdf -- Henning Westerholt - Development Consumer Products / Consumer Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

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Re: [Kamailio-Users] how to create chatrooms in imc
by Iñaki Baz Castillo 02 Sep '09

02 Sep '09

2009/9/2 Rajesh <Rajesh.sreedharan(a)siptech.com>: > Thanks again for your time. Please, don't send private mails. No free consultance service here. > Could you tell me how to add send messages via xmpp protocol.. Hum... use a XMPP client for that. I don't understand your question. > i was also asked to send cc to mailing list..is > users(a)lists.openser-project.org the mail address to which i need to mark cc > to The Kamailio users maillist is users(a)lists.kamailio.org. Use it instead of OpenSER's maillist as Kamailio is the new name of the project. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] how to create chatrooms in imc
by Daniel-Constantin Mierla 02 Sep '09

02 Sep '09

always cc to mailing list, private messages are ignored unless they are used for advanced troubleshooting where privacy is required. On 02.09.2009 12:32 Uhr, Rajesh wrote: > Dear Daniel, > Thankyou so much for your post. I read the > "readme" file and i am still unclear. I would like to know where to > add the command to create the chat room.Do i need to add it in the > .cfg file.Kindly excuse me if my questions are vague as am little new > to this Product. yes, in the config file you have to call imc_manager(): http://kamailio.org/docs/modules/1.5.x/imc.html#id2452973 Cheers, Daniel > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 02.09.2009 12:05 Uhr, Rajesh wrote: >>> Dear Openser Team, >>> I am curretnly configuring imc >>> module for kamailio and i would like to know(i) how to create chat >>> rooms , (ii)how to add users to the current chatrooms . One mail >>> from you can save a lot of time for me. >> >> see here: >> http://kamailio.org/docs/modules/1.5.x/imc.html#id2467570 >> >> With default config options for imc module, the command to create a >> chat room is: >> #create chatroom1 >> >> See more in the readme. >> >> Cheers, >> Daniel >> >>> >>> -- >>> Thanks & kind Regards, >>> Rajesh Sreedharan >>> Software Trainee-Wireless & Mobile Technology >>> >>> SIPTECH Solutions Limited >>> G4, Elnet Software City, CPT Road, >>> Taramani, Chennai - 600 113, India. >>> Tel : +91 (44) 2254 1473/ 74 >>> Extn.:206 >>> Fax : +91 (44) 2254 1475 >>> E-Mail :Rajesh.Sreedharan@Siptech.com >>> <mailto:Rajesh.Sreedharan@Siptech.com?> >>> Website : www.siptech.com <http://www.siptech.com/> >>> <http://www.siptech.com> >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> > > > -- > Regards, > Rajesh Sreedharan > Software Trainee-Wireless & Mobile Technology > > SIPTECH Solutions Limited > G4, Elnet Software City, CPT Road, > Taramani, Chennai - 600 113, India. > Tel : +91 (44) 2254 1473/ 74 > Extn.:206 > Fax : +91 (44) 2254 1475 > E-Mail :Rajesh.Sreedharan@Siptech.com > <mailto:Rajesh.Sreedharan@Siptech.com?> > Website : www.siptech.com <http://www.siptech.com/> > <http://www.siptech.com> -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] how to create chatrooms in imc
by Rajesh 02 Sep '09

02 Sep '09

Dear Openser Team, I am curretnly configuring imc module for kamailio and i would like to know(i) how to create chat rooms , (ii)how to add users to the current chatrooms . One mail from you can save a lot of time for me. -- Thanks & kind Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

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[Kamailio-Users] kindly approve my request
by Rajesh 02 Sep '09

02 Sep '09

Deat Team, Kindly approve my request for posting my questions on openser/kamailio on ur webpage. -- Regards, Rajesh Sreedharan Software Trainee-Wireless & Mobile Technology SIPTECH Solutions Limited G4, Elnet Software City, CPT Road, Taramani, Chennai - 600 113, India. Tel : +91 (44) 2254 1473/ 74 Extn.:206 Fax : +91 (44) 2254 1475 E-Mail :Rajesh.Sreedharan@Siptech.com <mailto:Rajesh.Sreedharan@Siptech.com?> Website : www.siptech.com <http://www.siptech.com/> <http://www.siptech.com>

1 0

[Kamailio-Users] registrar lookups without reason
by jh＠tutpro.com 02 Sep '09

02 Sep '09

when analyzing the weird problem, where request uri gets rewritten with aor's contact even when there is no command in config file that would do it, i noticed these lines in syslog: Aug 31 16:41:50 /usr/sbin/kamailio[23139]: DBG:core:rewrite_uri: rewriting Request-URI with 'sip:foo.bar@ip-address:5060;transport=UDP' Aug 31 16:41:50 opensipg /usr/sbin/kamailio[23139]: DBG:registrar:lookup: 'foo.bar@ip-address' Not found in usrloc Aug 31 16:41:50 opensipg /usr/sbin/kamailio[23139]: DBG:registrar:lookup: 'foo.bar@ip-address' Not found in usrloc any one has any idea why lookup was called even when it is not in the script? this is getting very weird. looks like kamailio is executing statements on its own. -- juha

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[Serusers] CANCEL passed through SER is occasionally ignored by destination switch
by Frank Durda IV 02 Sep '09

02 Sep '09

I have a situation where CANCEL messages sent by the caller through SER are sometimes ignored by the destination switch. These events are infrequent, and seem to be ignored only if they are received in a narrow window between the 100 Trying and the 183 Session Progress or 180 Ringing message. The destination switch eventually emits a 183/180 and even a 200 OK as the called party answers the call of a long-departed caller. I don't believe that this is a race condition, as the called numbers are sometimes wireless parties, who have 4-10 seconds of time between 100 Trying/IAM and 180/183/ACM responses, and I have seen ignored cancels at the beginning and middle of that lengthy period. So a race condition for that large a period with such an infrequent occurrence seems unlikely. The destination (SIP to ISUP gateway) equipment vendor points out that in the dozen samples of this occurring that the branch tag of the CANCEL didn't match the one shown in the INVITE, and per RFC 3261, their decision to ignore the CANCEL seems to be the right one. In particular, section 17.2.3 criteria 1. This immediately made me think of SER and its habit of saving time and memory by not remembering some or all call state unless you tell it to, via various knobs that are turned off by default. However, I can't explain why most cancels that arrive in the same point in the call setup apparently do meet criteria and only a small number don't. It would seem that SER would have mismatched branch tags all the time or never, depending on the knob settings. Any ideas? Thanks in advance.

1 0

[Kamailio-Users] SIP Redirect server
by Meftah Tayeb 01 Sep '09

01 Sep '09

hello all please could anyone help me to use Kamailio a a redirect server from server1.com that is the kamailio redirect server to server2.com that is the registrare server? thanks for any help __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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[Kamailio-Users] Symmetric NAT
by David 01 Sep '09

01 Sep '09

Hey, So I have setup a Kamailio server, it works well with most devices, but works terribly with Symmetric NAT. If you haven't already guessed, I am using STUN for NAT traversal. Could some people share there experiences with Symmetric NAT ? I see a lot of places saying that STUN is not good with it, but I haven't found a lot of suggestions for how to handle it. Thanks, David

3 3

[Kamailio-Users] LCR questions ( ping and flags columns )
by Ricardo Martinez 01 Sep '09

01 Sep '09

Hello list. I have a couple of questions about the LCR module. The first one is about the "ping" column in the gw table. I guess this column stores the "results" of execute a ping to the gateways in the LCR.gw table isn't?. So how this "ping" is executed? It is just a simple ICMP message? Or maybe is an OPTIONS sip message to the gateway?. I saw in the lcr_mod.c file some ping_method, ping_interval variables, but in the README file there is nothing about the "ping". Could someone clarify this to me this? Second. In the mysql "gw" table there is a column called "flags". According to the README file it said that this column can store gateways "capabilities". For example, and I hope that someone could guide me here, can i store the maximum channels available per gateway?, and use it to select or not select a gateway according the gateway channels used?. I hope that someone could help me here. Thank in advance, Regards, Ricardo Martinez.-

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[Kamailio-Users] Kamailio wins Best of Open Source award.
by Alex Balashov 31 Aug '09

31 Aug '09

http://www.infoworld.com/d/open-source/best-open-source-software-awards-200… -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

2 1

[Kamailio-Users] watchers and active_watchers tables
by catalina oancea 31 Aug '09

31 Aug '09

Hi, Could someone please explain the usage of the watchers table? I understand the usage active_watchers, but why the watchers table too? Thanks, Catalina

2 1

[Kamailio-Users] Issue in starting openser-0.9.5 for embedded application
by Ratheendran R 31 Aug '09

31 Aug '09

Hi All, I am using openser-0.9.5 version for embedded application,I am able to sucessfully cross compile(gcc 2.95) it.however the application fails to start. The above SER version is selected based on its compatibility with the cross compiler. below are my observations 1.if I run './openserctl start' from the command line I get the below message printf: No such file or directory PID file /var/run/openser.pid does not exist -- OpenSER start failed 2. If I run './openser' from command line I get the below message Listening on udp: 192.168.1.22 [192.168.1.22]:5060 tcp: 192.168.1.22 [192.168.1.22]:5060 Aliases: Too much shared memory demanded: 33554432 Please guide me in fixing this issue. Thanks in advance. Ratheendran

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[Kamailio-Users] rewriteuser( ) with avp
by Alex R.S.M 31 Aug '09

31 Aug '09

Hi, I'm trying to change "username" field of URI in the INT message. When using rewriteuser() function with avp variable as below, the username is changed to $avp(i:8) itself instead of the value of the avp. rewriteuser("$avp(i:8)"); I've tried all sorts of diff formats and none of them replace username with the value of avp variable.: rewriteuser($avp(i:8)); rewriteuser("avp(i:8)"); is this a bug? Thanks, A-RS _________________________________________________________________ Windows Live: Keep your friends up to date with what you do online. http://windowslive.com/Campaign/SocialNetworking?ocid=PID23285::T:WLMTAGL:O…

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[Kamailio-Users] scope of AVPs (AVP defined when execution of script starts)
by jh＠tutpro.com 29 Aug '09

29 Aug '09

i have always thought that AVPs exits per transaction. that does not seem to be the case, however, because when i put in the beginning of my script xlog("L_INFO", "Caller lcr_id $avp(CALLER_LCR_ID_AVP)\n"); this gets printed to syslog: Aug 29 11:34:34 localhost /usr/sbin/sip-router[10210]: INFO: Caller lcr_id 2 is this the indented behavior? do i really need to delete all AVPs in the beginning of the script in order to make sure that they don't have any values left from processing of previous message? -- juha

1 1

[Kamailio-Users] Phones that support TLS with certificate
by catalina oancea 27 Aug '09

27 Aug '09

Hello, I am trying to find the phones that can use tls with certificate, I mean that the phone doesn't accept incoming calls unless the server certificate is signed by a public authority or manually added to a list. For now my conclusions are: -Snom 360 - works with TLS. But although snom wiki states that a certificate is needed, incoming calls are accepted even if the certificate is not uploaded. -SJPhone - same as snom -Bria - this is the only phone I could find that works only if the certificate is loaded. Has anybody used other phones with tls and can give me a piece of advice? also, maybe I am not configuring Snom right and that's why it doesn't need a certificate, any ideas? Thanks Catalina

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[Kamailio-Users] Perl module and memory leak in append_branch
by James Puckett 26 Aug '09

26 Aug '09

I have followed a bit of the discussions regarding the problems with memory management between the perl modules and the core kamailio software. I need to get append_branch working from within the perl module so that it doesn't cause a memory leak. It's been a while since I did any serious straight C programming, but if someone can point me to the correct way to handle memory allocated with the module so that it gets freed in the core I'd be glad to take this problem on. Specifically, I'm using something that looks like: $m->append_branch("<sip:".$cdpn."\@".$peers{$_}."?From=%3Csip:". $sourcenumber."@".$destIp."%3E;tag=".$fromtag.">", $qval); This leaves about 190 bytes per call unfreed. Thanks. James Puckett - Engineers' Consulting Group Phone 229-316-0012 - Fax 229-316-1490

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[Kamailio-Users] What happens with filled htables?
by catalina oancea 26 Aug '09

26 Aug '09

Hello I use the htable module a lot but the only problem, when I add a new entry in a htable, is when I will delete it. My question is: if the hash table is completely filled and I try to add a new value to it, do I get an error or is an old value automatically deleted to be able to write my new value? If an old value was automatically deleted whenever a new value is added, I wouldn't have to bother deleting the values I no longer need. Thanks Catalina

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[Serusers] translate message from TLS to UDP
by qiulei212 26 Aug '09

26 Aug '09

hi everyone: sorry to trouble you again! I use the SER as a proxy between a SIP-softphone and the asterisk. The message between the sip-phone and SER is TLS and i want to translant the message by UDP to my asterisk.HOW would i do? I have tried the forward_udp , t_relay_to_tls ,t_relay_to_udp and no useless. which function should i prefer? thanks!! qiulei212 2009-08-26

2 1

[Serusers] problems with stun using uas
by lorenzo 25 Aug '09

25 Aug '09

hi! i'm the developer of a sip app that happens to be connecting to a sip server that's using SER. i'm having troublewith the way SER handles incoming calls to my stun-enabled ua (which is placed behind a router) I have two clients running on the same machine. one is my app, the other is Gizmo5. when i make an outgoing call from my app to the user i'm logged in with in Gizmo5, everything works. but when i call my app from the Gizmo5 app, sniffing the network traffic with wireshark, the only thing i see happening is the sip server sending a 100 Giving a try to the calling ua. The reason i'm posting this here, is that i know it's not a bug with my app, since everything works fine when i don't use stun. (note that i don't have direct access to ser's configuration) this is my app's REGISTER request (note that it has a *public ip*): REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 93.148.135.54:5060;rport;branch=z9hG4bK70207 Max-Forwards: 70 To: <sip:asymmetric@proxy01.sipphone.com> From: <sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK57274073 Call-ID: 557402852938(a)93.148.135.54 CSeq: 2 REGISTER Contact: <sip:asymmetric@93.148.135.54:5060;transport=udp> Expires: 3600 User-Agent: mjsip stack 1.6 Authorization: Digest username="asymmetric", realm="proxy01.sipphone.com", nonce="xxx", uri="sip:proxy01.sipphone.com", algorithm=md5, response="yyy" Content-Length: 0 and the response: SIP/2.0 200 OK Via: SIP/2.0/UDP 93.148.135.54:5060;rport=12810;branch=z9hG4bK70207 To: <sip:asymmetric@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.50aa From: <sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK57274073 Call-ID: 557402852938(a)93.148.135.54 CSeq: 2 REGISTER Contact: <sip:asymmetric@93.148.135.54:5060;transport=udp>;expires=3600 Content-Length: 0 Gizmo5's app REGISTER request: REGISTER sip:proxy01.sipphone.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-bbc1e2712d417352-1--d87543-;rport Max-Forwards: 70 Contact: <sip:17474492586@93.148.135.54:10251> To: <sip:17474492586@proxy01.sipphone.com> From: <sip:17474492586@proxy01.sipphone.com>;tag=f43be43c Call-ID: b1b4a060512d4b0c1250955005@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 2 REGISTER Expires: 1800 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER User-Agent: MacGizmo/2.0.02 (Gizmo-s2n1) Authorization: Digest username="17474492586",realm="proxy01.sipphone.com",nonce="xxx",uri="sip:proxy01.sipphone.com",response="yyy",algorithm=MD5 Content-Length: 0 and response: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-bbc1e2712d417352-1--d87543-;rport=10251;received=93.148.135.54 To: <sip:17474492586@proxy01.sipphone.com>;tag=92390300a369f0d75803e369c733575e.c70f From: <sip:17474492586@proxy01.sipphone.com>;tag=f43be43c Call-ID: b1b4a060512d4b0c1250955005@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 2 REGISTER P-Behind-NAT: Yes Contact: <sip:17474492586@93.148.135.54:10251>;expires=1800 Content-Length: 0 in the second case, the client gets the P-Behind-NAT flag set to Yes, and for a good reason. But then why is this all i get when trying to call the first ua? : SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.1.100:64064;branch=z9hG4bK-d87543-7fb07a028db5c137-1--d87543-;rport=10251;received=93.148.135.54 To: <sip:asymmetric@proxy01.sipphone.com> From: <sip:17474492586@proxy01.sipphone.com>;tag=bc135372 Call-ID: 88d41c746d51146e1250955219@YXN5bW1ldHJpYy5sb2NhbA.. CSeq: 1 INVITE P-Behind-NAT: Yes Content-Length: 0 whereas the inverse works: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 198.65.166.131;branch=z9hG4bKf3b1.bc014575.0 Via: SIP/2.0/UDP 93.148.135.54:5060;rport=13171;branch=z9hG4bK82654 Record-Route: <sip:198.65.166.131;lr;ftag=z9hG4bK90319798> Contact: <sip:17474492586@proxy01.sipphone.com> To: <sip:17474492586@proxy01.sipphone.com>;tag=6ae0e759 From: "asymmetric"<sip:asymmetric@proxy01.sipphone.com>;tag=z9hG4bK90319798 Call-ID: 593542951976(a)93.148.135.54 CSeq: 1 INVITE User-Agent: MacGizmo/2.0.02 (Gizmo-s2n1) Content-Length: 0 CQBM: 244 -- thanks a lot! asymmetric

3 3

Presence over NAT
by Andrew O. Zhukov 25 Aug '09

25 Aug '09

Is there any samples let's forwarded Subscribe fix the Contact: I mean Proxy URI not the customer mapped. Let's the followed NOTIFY go over proxy not P2P. Thanks in avance. Andrew O. Zhukov

5 9

[Kamailio-Users] kamailio 3.0 - the time before freezing
by Daniel-Constantin Mierla 25 Aug '09

25 Aug '09

Hello everybody, during the last IRC devel meeting, we pinned end of summer to be the time to enter testing phase for the new major release - codenamed kamailio 3.0. There was quite a lot of brand new features, considering that most of devel effort was directed to integration. A try to collect new items is available at: http://sip-router.org/wiki/features/new-in-devel Of course, all ser features are available as well, resulting a large amount of new stuff in kamailio 3.0 vs kamailio 1.5.x. See: http://sip-router.org/benefits/ http://www.iptel.org/ser/features I propose to start the testing phase sometime next week. Meanwhile, let's see if something important was forgotten from kamailio core or tm. To my knowledge: - path support in tm - Andrei has it on his todo afaik - in a way or another, it must get in next release - drop compatibility - drop reply in reply_route - internally drop and exit are now differentiated by core, needs review and update of handling after running the routes - on my to-do Apart seas, all Kamailio modules are fully ugraded to new core - I started but lack of test env + busy summer resulted in delays - can be done during testing period, being just api updates. If you have discovered something else, let us know. Thanks, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] exchanging CDR data between Asterisk servers
by Klaus Darilion 24 Aug '09

24 Aug '09

Hi! I have the following setup: PSTN<------>Asterisk<-----SIP-->Asterisk GW/LCR \ \ ... \ \ ... \ --SIP-->Asterisk \ ... ------->Asterisk The GW-Asterisk just does the gatewaying stuff and writes the CDRs for the billing system. The other Asterisk servers handle all the services (IVR, REGISTER, ...) Some scenarios require to write some additional data to the CDRs. For outgoing calls this is not a problem (I signal the extra data in a SIP header and set a CDR() variable in the GW asterisk). My problem are incoming calls. The "extra data" is only known to the service Asterisk, but the CDR is written by the GW Asterisk. Does anybody know a method how to signal the "extra data" during the call from the server Asterisk back to the GW Asterisk und put it into a CDR() variable? Regards Klaus

3 4

[Kamailio-Users] TLS TCP error
by shehzad p 24 Aug '09

24 Aug '09

Hello, My Kamailio (tls version) is configured to handle TCP as well as TLS traffic. Making calls from softphone using TCP only works fine. But my TLS traffic has been stucked. I can see lots of error as below in messages file: ====================================================================== Aug 24 06:05:01 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Aug 24 06:05:01 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_send: connect failed Aug 24 06:05:01 ns110 /usr/local/sbin/kamailio[24273]: ERROR:tm:msg_send: tcp_send failed Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_blocking_connect: poll error: flags 18 Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (110) Connection timed out Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcpconn_connect: tcp_blocking_connect failed Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:core:tcp_send: connect failed Aug 24 06:08:10 ns110 /usr/local/sbin/kamailio[24273]: ERROR:tm:msg_send: tcp_send failed Aug 24 06:08:25 ns110 /usr/local/sbin/kamailio[24279]: ERROR:core:_tls_read: something wrong in SSL: 5 Aug 24 06:08:25 ns110 /usr/local/sbin/kamailio[24279]: ERROR:core:tcp_read_req: failed to read Aug 24 06:09:00 ns110 /usr/local/sbin/kamailio[24277]: INFO:core:tls_accept: client did not present a certificate ================================================================================= I am very confused, because with the same configuration, another kamailio server works very well without any error. I am not able to know what could be the issue in this server with same configuration. What could be the cause and resolution of this errors? Any help will be appreciated Thanks in advance, Mohammed Shehzad -- View this message in context: http://www.nabble.com/TLS-TCP-error-tp25114961p25114961.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 3

[Kamailio-Users] new PV class: $sel(name)
by Daniel-Constantin Mierla 24 Aug '09

24 Aug '09

Hello, there is a new PV class that provides access to SER select variables. A list with available selects can be found at: http://sip-router.org/wiki/cookbooks/selects/devel Therefore, all Kamailio modules relying on PV framework can get access to select values without any change. See example of usage from config at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#selects Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Kamailio-Users] sip phone behind the nat
by aiphie 24 Aug '09

24 Aug '09

Hi,body. I'm using the kamailio as a sip server,If in the local network ,It works very well ,But when i set the sip server in the internet, When starting voice chat,Two sip phone will receive the behind net ip address(sip phone's local ip address). How i can configure sip server(kamailio) let sip server send sip phone's peer ip address using front nat ip address(internet ip address)? Any body can give me how to realize it? Thank you! --Aiphie 2009-08-21 “欢乐家庭总动员”--第七届全球通演出季网络互动有惊喜，千元大奖送给您！移动新推GPRS流量优惠系列新套餐，手机上网省钱又省心！

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Re: [Kamailio-Users] sip phone behind the nat
by Daniel-Constantin Mierla 24 Aug '09

24 Aug '09

On 24.08.2009 9:38 Uhr, aiphie wrote: > hi,all > you probably forgot to cc to the mailing list. please do so all the time - there might be people that can help faster and better. > > I have use sed command to clear the comment which in front of regarding nat,But it's always use local network IP to fill session description protocol->connection information->connection address,I know sip server can rewrite this message item,Can you tell me how to let sip server rewrite it? > annex is my config file,please look what wrong in this? > It is very unlikely that one will debug a whole config file for you. You just need to investigate yourself and come to mailing list with specific questions. The media IP is updated with what RTPPproxy is listening on. What is the command used to start rtpproxy? Cheers, Daniel > thank you! > > --aiphie > > > > ----- Original Message ----- > From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> > To: "aiphie" <chengwu8(a)139.com> > Cc: <users(a)lists.kamailio.org> > Sent: Friday, August 21, 2009 6:55 PM > Subject: Re: [Kamailio-Users] sip phone behind the nat > > > >> Hello, >> >> On 21.08.2009 12:52 Uhr, aiphie wrote: >> >>> Hi,body. >>> I'm using the kamailio as a sip server,If in the local network ,It >>> works very well ,But when i set the sip server in the internet, When >>> starting voice chat,Two sip phone will receive the behind net ip >>> address(sip phone's local ip address). >>> How i can configure sip server(kamailio) let sip server send sip >>> phone's peer ip address using front nat ip address(internet ip address)? >>> Any body can give me how to realize it? >>> >> you need to enable nathelper module + appropriate lines in config file >> and install rtpptoxy. >> >> If you check the default config file you will see at the top some >> comments regarding nat traversal and what sed command you can use to >> enable it. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> * http://www.asipto.com/ >> > > -- Daniel-Constantin Mierla * http://www.asipto.com/

1 0

[Serusers] TLS module load error
by qiulei212 24 Aug '09

24 Aug '09

hello friends: I crashed when i compile the SER source and use the tls.so! I have install the openssl 0.9.8 in the default folder :/usr/local and install the ser-2.0.0-rc1 in the default too. run the command below: make group_include="standard" include_modules="tls" all make group_include="standard" include_modules="tls" installthe tls.so in the /usr/local/lib/ ser/modules folderthe I add the loadmodule "/usr/local/lib/ser/modules/tls.so" in the ser.cfg and start /usr/local/sbin/ serbut the ser start failed!I run the ser -c and found:ERROR: load_module: could not open module </usr/local/lib/ser/modules/tls.so>: /usr/local/lib/ser/modules/tls.so: undefined symbol: deflate I don't know Why and ask help for you ! Thanks a lot ! qiulei212 2009-08-21

2 2

[Kamailio-Users] Choosing IP address when relaying request
by mayamatakeshi 20 Aug '09

20 Aug '09

Hello, I have 2 ip addresses being listened in my cfg file. Is it possible to set which ip address will be used when relaying a request? regards, takeshi

3 4

[Kamailio-Users] Problem with BYE when using TCP or TLS between two Kamilio proxies
by Dubravko Caric 20 Aug '09

20 Aug '09

Hi all, I have a problem with routing BYE messages (when called UAC sends BYE) when connection between two Kamailio (1.5.1 with identical configurations) proxies is done by TLS/TCP. WORKING: UAC#1 <-- UDP/TCP/TLS --> Proxy#1 <--UDP--> Proxy#2 <-- UDP/TCP/TLS --> UAC#2 NOT WORKING: UAC#1 <-- UDP/TCP/TLS --> Proxy#1 <-- TCP or TLS --> Proxy#2 <-- UDP/TCP/TLS --> UAC#2 By looking at ngrep log I found out that when “UAC#1” sends INVITE, “Proxy#1” sends right “Contact” header (Contact:<sip:bob@IP_ADDRESS_OF_UAC#1:6126;transport=udp>) to “Proxy#2” , but “Proxy#2” sends wrong “Contact” header (Contact:<sip:bob@IP_ADDRESS_OF_PROXY#1:39879;transport=udp>) to “UAC#2”. When UAC#2 sends BYE, it is sends message with this wrong header and when it arrives on “Proxy#1” proxy doesn’t sends BYE message to UAC#1. Or when UACs with TCP, this are “Proxy#1” errors: DBG:core:tcp_send: no open tcp connection found, opening new one ERROR:core:tcp_blocking_connect: poll error: flags 18 ERROR:core:tcp_blocking_connect: failed to retrieve SO_ERROR (111) Connection refused ERROR:core:tcpconn_connect: tcp_blocking_connect failed ERROR:core:tcp_send: connect failed ERROR:tm:msg_send: tcp_send failed ERROR:tm:t_forward_nonack: sending request failed DBG:tm:t_relay_to: t_forward_nonack returned error What bothers me is that when I call the other UAC, he can hangup the call (even when he is CALLED UAC), then I restart both proxies and it is other way around. I do not know if I described problem clearly but I am lost in where to look next so any help would be appreciated. Why proxy change Contact header when I switch from UDP to TCP between proxies? I'm not rewriting Contact header and only thing that I change in configuration when switching from UDP to TCP is this: if($rd=="example.com") { t_relay("tcp: example.com:5063"); exit; } Thanks a lot /dubravko

2 4

[Kamailio-Users] limits per subscriber and per gateway
by Dmitri G. 20 Aug '09

20 Aug '09

Hello, First I would like to say thank you for the helpful members here, I have got useful tips and suggestions here. I am trying to implement some kind of "protection" for me, let me describe what is my problem at the moment. I know I ask a bit much questions here .) I would like to limit the number of incoming calls from my customers, but on a customer basis. I have a subscriber with an Asterisk, so it will be a bad idea to set a limit to 1 concurrent call for this customer, but limit 1 will work for a customer with an IP phone (remember the 100rel thing what I posted here? that was a workaround for this subscriber... I mean to remove 100rel from the supported header). So basically I would like to limit every subscriber, on a per customer basis. The next big thing is the lcr gateways, so where I send the calls. I have 2 peers, one of then have a limit for me, it's 20 concurrent calls. They don't limit me, so I can send out 30-40-50 calls to them, but they will charge me a big amount of Lei (Romanian currency) for every calls above 20 at the same time. They have freeswitch.... doesn't matter, I'm just saying they have freeswitch :) But they have very very good price(for 20 calls what i mentioned), so I would like to fill that "20 calls" space. I have asked them to give some limits for me, so refuse calls above 20 (so this way lcr can send out to the another gateway), but they don't want to do it (well, they need money I think) I would like to store values for these limits in mysql (column "limit" in subscriber table, and column "calllimit" in gw table, or I can store it in a separate table). Right now I store these values in a mysql table called "iplimits" I am not perfectly sure about how I can select these values from the table, and how can I "play" with these values. Even don't know how to retrieve the current number of concurrent calls per gateway or per subscriber. And what to do with these selected values? Store it in an avp and then do some checks against it? I am reading the module docs, especially sqlops, but I think I can't see the wood for the trees. I hope somebody can give me suggestions (or an example) about this. Kind regards, Dmitri

3 5

[Kamailio-Users] inbound "failover"
by Dmitri G. 20 Aug '09

20 Aug '09

Sorry, I have accidentally sent this message to opensips mailing list, I am sending it now to the right place Hello, I would like to implement some kind of failover for my Asterisk, let me describe how I would like to see it. I registered 2 sip users with my Kamailio, 1020(a)domain.com and 1030(a)domain.com. I have added aliases to 1020 (aliases from 1021@domain com to 1029(a)domain.com). Right now calls to 1020-1029 goes well to 1020, it works fine. But I would like to do the following: If 1020 isn't registered with Kamailio (let's say if registration for 1020 is down in Kamailio, so AOR not found for 1020), it is possible to route calls to 1030? So route calls to 1030 only when registration for 1020 isn't active in Kamailio. I have tried manipulating with faillure_route, but without any luck. I hope it is clear what I would like to do :) Any help would be much appreciated. Kind regards, Dmitri

2 2

Re: [Kamailio-Users] kamailio syslog logging
by Henning Westerholt 19 Aug '09

19 Aug '09

On Mittwoch, 19. August 2009, Jason Penton wrote: > Hey Henning, > > no, none of my log messages from the cfg file come into the log file. > However I do get the following: > > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 797031 local0.debug] > DBG:core:fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 596959 local0.debug] > DBG:core:fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16560 > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 362601 local0.debug] > DBG:core:fm_malloc_init: params (fbe00000, 33554432), start=fbe00000 > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 551434 local0.debug] > DBG:core:shm_mem_init_mallocs: success > Aug 19 17:35:38 ug2s02-zone2 kamailio[12273]: [ID 123545 local0.info] > INFO:core:init_tcp: using /dev/poll as the TCP io watch method (auto > detected) > > weird that I get some messages but not the others. I have alot of LM_INFO > messages in my code that I dont see too????? Hi Jason, so it seems that you indeed get only DEBUG messages in the log file, so try the suggestion Alex gave to you - local0.* This should catch all. Cheers, Henning

2 1

[Kamailio-Users] kamailio syslog logging
by Jason Penton 19 Aug '09

19 Aug '09

Hi All, I am having some problems logging Kamailio in debug mode using syslog. All is fine if I log to stderr. However, when I log to syslog I get a few dbug messages and then it stops -- I dont get any of my routing messages or anything???? i am suing solaris? Any suggestions Cheers Jason

3 6

[Serusers] new_t: out of mem
by Roger Tan 19 Aug '09

19 Aug '09

hi. I tested my ser server which version is 0.9.7+cvs20080424 used sipps sending 1000 invite per sec. after about 10 sec, there was an out of mem error. Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: new_t: out of mem: Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: t_newtran: new_t failed Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sip_msg_cloner: cannot allocate memory and my memlog is: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- Aug 18 10:11:02 ubuntu /sbin/ser[20787]: Memory status (shm): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: qm_status (0xb5acf000): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: heap size= 33554432 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: used= 1914592, used+overhead=2224736, free=31329696 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: max used (+overhead)= 33534720 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping all alloc'ed. fragments: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 0. N address=0xb5ad41a8 frag=0xb5ad41a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 1. N address=0xb5ad41c8 frag=0xb5ad41c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 2. N address=0xb5ad41e8 frag=0xb5ad41e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 3. N address=0xb5ad4208 frag=0xb5ad4200 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 4. N address=0xb5ad4228 frag=0xb5ad4220 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 5. N address=0xb5ad5238 frag=0xb5ad5230 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 6. N address=0xb5ad6248 frag=0xb5ad6240 size=2800 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 7. N address=0xb5ad6d48 frag=0xb5ad6d40 size=1680 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 8. N address=0xb5ad73e8 frag=0xb5ad73e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 9. N address=0xb5ad7408 frag=0xb5ad7400 size=1572864 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 10. N address=0xb5c57418 frag=0xb5c57410 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 11. N address=0xb5c57438 frag=0xb5c57430 size=320 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 12. N address=0xb5c57588 frag=0xb5c57580 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 13. N address=0xb5c575c8 frag=0xb5c575c0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 14. N address=0xb5c57628 frag=0xb5c57620 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 15. N address=0xb5c57688 frag=0xb5c57680 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 16. N address=0xb5c576e8 frag=0xb5c576e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 17. N address=0xb5c57708 frag=0xb5c57700 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 18. N address=0xb5c57728 frag=0xb5c57720 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 19. N address=0xb5c57748 frag=0xb5c57740 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 20. N address=0xb5c57768 frag=0xb5c57760 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 21. N address=0xb5c57798 frag=0xb5c57790 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 22. N address=0xb5c777a8 frag=0xb5c777a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 23. N address=0xb5c777c8 frag=0xb5c777c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 24. N address=0xb5c777e8 frag=0xb5c777e0 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 25. N address=0xb5c77818 frag=0xb5c77810 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 60. N address=0xb5ca7948 frag=0xb5ca7940 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 61. N address=0xb5ca7988 frag=0xb5ca7980 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 62. N address=0xb5ca79a8 frag=0xb5ca79a0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 63. N address=0xb5ca7a08 frag=0xb5ca7a00 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 64. N address=0xb5ca7a48 frag=0xb5ca7a40 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 65. N address=0xb5ca7a78 frag=0xb5ca7a70 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 66. N address=0xb5ca7a98 frag=0xb5ca7a90 size=65808 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping free list stats : Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 1. fragments no.: 1232, unused: 0 ^I^I bucket size: 16 - 16 (first 16) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 38. fragments no.: 54, unused: 0 ^I^I bucket size: 608 - 608 (first 608) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 43. fragments no.: 54, unused: 0 ^I^I bucket size: 688 - 688 (first 688) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 44. fragments no.: 3991, unused: 0 ^I^I bucket size: 704 - 704 (first 704) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 57. fragments no.: 114, unused: 0 ^I^I bucket size: 912 - 912 (first 912) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 58. fragments no.: 4004, unused: 0 ^I^I bucket size: 928 - 928 (first 928) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 68. fragments no.: 46, unused: 0 ^I^I bucket size: 1088 - 1088 (first 1088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 82. fragments no.: 109, unused: 0 ^I^I bucket size: 1312 - 1312 (first 1312) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 115. fragments no.: 34, unused: 0 ^I^I bucket size: 1840 - 1840 (first 1840) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 174. fragments no.: 3795, unused: 0 ^I^I bucket size: 2784 - 2784 (first 2784) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 186. fragments no.: 790, unused: 0 ^I^I bucket size: 2976 - 2976 (first 2976) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 187. fragments no.: 3820, unused: 0 ^I^I bucket size: 2992 - 2992 (first 2992) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 262. fragments no.: 1, unused: 0 ^I^I bucket size: 4192 - 4192 (first 4192) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 318. fragments no.: 1, unused: 0 ^I^I bucket size: 5088 - 5088 (first 5088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- I think it's not a memory leak but the so many small memory fragments, and those fragments were not combinated. How I solve this problem? Thanks. -- MSN: miao__tan(a)hotmail.com Email: m.roger.t(a)gmail.com

3 2

[Kamailio-Users] SIP Project for Beginner
by Sangmane 19 Aug '09

19 Aug '09

Hi all, I am new to SIP. I've downloaded Netbeans 6.7 and installed related plugins. When I tried to create an SIP Project, I couldn't find any SIP Categories in NB 6.7. Is there any basic guidance for SIP in Netbeans ? thank you very much. Sangmane -- View this message in context: http://www.nabble.com/SIP-Project-for-Beginner-tp25010377p25010377.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[Kamailio-Users] remove 100rel from Supported header
by Dmitri G. 19 Aug '09

19 Aug '09

Hello, Is there a way to remove 100rel from Supported header (only this, and keep the others) by Kamailio? I am using Kamailio 1.5.1 notls. Kind regards, Dmitri

4 7

[Kamailio-Users] T38
by BERGANZ François 19 Aug '09

19 Aug '09

Hello, In a SIP Frame, how to see if it is a T38 frame? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

3 2

[Serusers] SER with a PSTN Gateway (pstn, cfg) vs called-party-BYE or Re-INVITE
by Frank Durda IV 19 Aug '09

19 Aug '09

I have noted that the pstn.cfg example (or anything built from its premise) doesn't work right for one very important situation, that hopefully someone has already discovered and has a good fix for. The documentation isn't helping. The situation: I have a call initiated by some SIP client, with the INVITE passing through SER and sent on to a PSTN gateway (the SIP to SS7/ISUP conversion occurs there). The call is in progress (post 200 OK) and both parties are chatting away. So far, so good. The problem: Trouble begins when the called-party (somewhere on the PSTN) hangs up first. What happens: This event triggers a SS7 REL message, and the SIP/SS7 switch providing the VOIP gateway service emits a BYE message, which gets sent to SER. SER receives the message and doesn't treat it as a reply (because it isn't), but instead it comes in the door at route {}. The pstn.cfg-like script quickly determines that this BYE message is coming from the gateway (correct) and skips all the authentication checks (good), and does a rewritehostport() using the PSTN gateway IP address (bad), and finally does a t_relay() (very bad). The BYE is now emitted by SER and sent back to the PSTN gateway switch, rather than on to wherever the INVITE originally came from, the device that needs to know about this BYE. This quickly degenerates into a small battle between PSTN gateway switch and SER, as the PSTN gateway isn't getting the answer back it is expecting in response to the BYE it sent to SER, and the PSTN gateway certainly doesn't want to see the message it just sent out come back over and over. Eventually various looping checks or timers kick in and the argument ends, but the calling system remains in the dark over the fact that the call is now over (other than getting a lot of silence). So that behavior is obviously wrong, but that's what the stock copy of pstn.cfg does here. SER obviously wants to be involved, because it added a Record-Route, but the script really isn't able to deal with the consequences. If the calling network was all SIP this would be merely annoying, but if it is another ISUP network (with a SIP leg in the middle), the originating ISUP legs of the call just rack up the toll charges, unaware that the other legs of the call are gone. Bad things also happen if the original INVITE included a Session-Expires: header, which causes the PSTN gateway to emit its own INVITE back to the calling network (via the SER) at intervals during the call, starting at 50% of the time specified in that header. These packets also die a horrible fate in the PSTN-Gateway and SER packet loop, which causes the call to collapse 32 seconds later. (That's unbelieveably bad.) Recognizing this apparent-shortcoming in the pstn.cfg example, and using the fact that the script could determine that this called-party-BYE or INVITE message came from the PSTN gateway and not somewhere else, I added tests to not perform the rewritehost that sends the message back to the gateway, but instead sets the address to that of the known place where the INVITE originally came from, because in this super simple example, there is only one place, eg phone-asterisk-SER-PSTN-SS7network-... Forcing BYE/INVITEs coming from the PSTN direction sort of fixes the problem of called-party BYEs, and the PSTN gateway device emitting re-INVITEs. However, it only works for ONE source, which is lousy as you have to hard-code the right place for the call in both directions! And of course, someone has too many calls for their one SBC (aka an asterisk box) and they have more than one now. In fact, many more, and any of them can send me the same range of calls from the same sources and to the same PSTN gateway. And it would be really nice if I sent the called-party-BYE or re-INVITEs back to the right SBC. But how? An examination of the numerous variables that SER has (as viewed via xlog() when the called-party-BYE or called-network-reINVITE comes in) shows that SER actually has the right IP address available that the called-party-BYE or re-INVITE should be sent back to (apparently saved as part of the transaction details from the original INVITE), but I can't figure out how to reference this value in "rewritehost()". Like most SER functions, rewritehost() doesn't appear to accept variable names as parameters, and as xlog uses cryptic abbreviations for what it displays, I probably wouldn't know what variable name to specify in rewritehost() anyway. (Did two different programmers do these parts?) Now, if you don't use "rewritehost()" in the send-back-to-caller direction and are hoping that SER will do the right thing on its own, you will be disappointed. The t_relay() merely sends the message back itself and you get a much smaller SIP message loop going. (This is because SER puts its IP address all over the original INVITE message so that the PSTN gateway would talk to SER rather than the true source of the call.) So that appears to means that you must set rewritehost() to a reasonable destination to get the message to move beyond SER, and somehow I have to choose a destination from many possible destinations the original INVITE came from, based on some piece of trivia that was in the original INVITE that I can somehow trigger on. That means if there are 250 different SBCs out there sending me calls, and one of them was where the original INVITE came from for the call I just got a called-party-BYE from, how do I force that called-party-BYE (or a called-network re-INVITE) to go back to the right place? Must I build a giant if statement that tests the addresses of all 250 sources and leaves a flag or something that can be used in 250 if statements later when the called-party-BYE or called-network-reINVITE shows up? Sounds horrible and a giant kludge! Any other solutions? Thanks in advance for thinking about this one!

2 1

[Serusers] Is there any way to validate the RTP/media address?
by Frank Durda IV 19 Aug '09

19 Aug '09

I am receiving INVITEs from a specific source, and I have a need to test the media IP address specified in the SDP payload against a list of approved netblocks. I'd like to catch this in the INVITE and reject the call if a RTP address in the SDP payload is out of a pre-defined range for that given source. Is there a way to get this address into a variable that can be used in "if" statements? As in something like this? if (src_ip==17.18.19.20 && method=="INVITE") { if (rtp_addr!=17.0.0.0/8) { sl_reply("502", "Rejected - Unauthorized RTP source"); drop; } } with rtp_addr being the made-up, hope-something-like-it-exists thing. Thanks in advance.

1 0

[Kamailio-Users] Kamailio installed on other dir issues
by Uriel Rozenbaum 18 Aug '09

18 Aug '09

Guys, I've installed all Kamailio in /opt/kamailio. When I execute some kamctl using cr module I get the following: /opt/kamailio/sbin/kamctl cr reload ERROR: This command requires a database engine - none was loaded Do I have to configure something? If i just run /opt/kamailio/sbin/kamctl I get the help lines and running /opt/kamailio/sbin/kamctl fifo cr_reload_routes makes effect as well. Thanks! Uriel

2 4

[Serusers] new_t: out of mem
by Roger Tan 18 Aug '09

18 Aug '09

hi. I tested my ser server which version is 0.9.7+cvs20080424 used sipps sending 1000 invite per sec. after about 10 sec, there was an out of mem error. Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: new_t: out of mem: Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: t_newtran: new_t failed Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Aug 18 09:40:12 ubuntu /sbin/ser[20805]: ERROR: sip_msg_cloner: cannot allocate memory and my memlog is: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- Aug 18 10:11:02 ubuntu /sbin/ser[20787]: Memory status (shm): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: qm_status (0xb5acf000): Aug 18 10:11:02 ubuntu /sbin/ser[20787]: heap size= 33554432 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: used= 1914592, used+overhead=2224736, free=31329696 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: max used (+overhead)= 33534720 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping all alloc'ed. fragments: Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 0. N address=0xb5ad41a8 frag=0xb5ad41a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 1. N address=0xb5ad41c8 frag=0xb5ad41c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 2. N address=0xb5ad41e8 frag=0xb5ad41e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 3. N address=0xb5ad4208 frag=0xb5ad4200 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 4. N address=0xb5ad4228 frag=0xb5ad4220 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 5. N address=0xb5ad5238 frag=0xb5ad5230 size=4096 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 6. N address=0xb5ad6248 frag=0xb5ad6240 size=2800 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 7. N address=0xb5ad6d48 frag=0xb5ad6d40 size=1680 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 8. N address=0xb5ad73e8 frag=0xb5ad73e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 9. N address=0xb5ad7408 frag=0xb5ad7400 size=1572864 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 10. N address=0xb5c57418 frag=0xb5c57410 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 11. N address=0xb5c57438 frag=0xb5c57430 size=320 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 12. N address=0xb5c57588 frag=0xb5c57580 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 13. N address=0xb5c575c8 frag=0xb5c575c0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 14. N address=0xb5c57628 frag=0xb5c57620 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 15. N address=0xb5c57688 frag=0xb5c57680 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 16. N address=0xb5c576e8 frag=0xb5c576e0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 17. N address=0xb5c57708 frag=0xb5c57700 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 18. N address=0xb5c57728 frag=0xb5c57720 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 19. N address=0xb5c57748 frag=0xb5c57740 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 20. N address=0xb5c57768 frag=0xb5c57760 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 21. N address=0xb5c57798 frag=0xb5c57790 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 22. N address=0xb5c777a8 frag=0xb5c777a0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 23. N address=0xb5c777c8 frag=0xb5c777c0 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 24. N address=0xb5c777e8 frag=0xb5c777e0 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 25. N address=0xb5c77818 frag=0xb5c77810 size=131072 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 60. N address=0xb5ca7948 frag=0xb5ca7940 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 61. N address=0xb5ca7988 frag=0xb5ca7980 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 62. N address=0xb5ca79a8 frag=0xb5ca79a0 size=80 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 63. N address=0xb5ca7a08 frag=0xb5ca7a00 size=48 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 64. N address=0xb5ca7a48 frag=0xb5ca7a40 size=32 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 65. N address=0xb5ca7a78 frag=0xb5ca7a70 size=16 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: 66. N address=0xb5ca7a98 frag=0xb5ca7a90 size=65808 used=1 Aug 18 10:11:02 ubuntu /sbin/ser[20787]: dumping free list stats : Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 1. fragments no.: 1232, unused: 0 ^I^I bucket size: 16 - 16 (first 16) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 38. fragments no.: 54, unused: 0 ^I^I bucket size: 608 - 608 (first 608) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 43. fragments no.: 54, unused: 0 ^I^I bucket size: 688 - 688 (first 688) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 44. fragments no.: 3991, unused: 0 ^I^I bucket size: 704 - 704 (first 704) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 57. fragments no.: 114, unused: 0 ^I^I bucket size: 912 - 912 (first 912) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 58. fragments no.: 4004, unused: 0 ^I^I bucket size: 928 - 928 (first 928) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 68. fragments no.: 46, unused: 0 ^I^I bucket size: 1088 - 1088 (first 1088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 82. fragments no.: 109, unused: 0 ^I^I bucket size: 1312 - 1312 (first 1312) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 115. fragments no.: 34, unused: 0 ^I^I bucket size: 1840 - 1840 (first 1840) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 174. fragments no.: 3795, unused: 0 ^I^I bucket size: 2784 - 2784 (first 2784) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 186. fragments no.: 790, unused: 0 ^I^I bucket size: 2976 - 2976 (first 2976) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 187. fragments no.: 3820, unused: 0 ^I^I bucket size: 2992 - 2992 (first 2992) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 262. fragments no.: 1, unused: 0 ^I^I bucket size: 4192 - 4192 (first 4192) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: hash= 318. fragments no.: 1, unused: 0 ^I^I bucket size: 5088 - 5088 (first 5088) Aug 18 10:11:02 ubuntu /sbin/ser[20787]: ----------------------------- I think it's not a memory leak but the so many small memory fragments, and those fragments were not combinated. How I solve this problem? Thanks. -- MSN: miao__tan(a)hotmail.com Email: m.roger.t(a)gmail.com

1 0

[Serusers] tcpconn_new: mem. allocation failure
by Roger Tan 18 Aug '09

18 Aug '09

Hi:

1 0

[Kamailio-Users] Generating SIP /TLS Certificate
by Meftah Tayeb 17 Aug '09

17 Aug '09

hello all please how i can generate a certificate to do sip / tls? and how to configure my kamailio? try to help me please thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4341 (20090817) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

2 1

Re: [Kamailio-Users] presence_dialoginfo context
by David 17 Aug '09

17 Aug '09

Hey, I definitely have an exit(), but even if I didn't most of the message is changed, an exit would not change that would it? David Klaus Darilion wrote: > Very strange. > > Maybe exit(); is missing after t_relay() loop and both PUBLISHs are > processed? I think I forgot the exit() in the config snippet. > > Yes, on the server. And please attach the trace as attachment - inline > is difficult to read > > klaus > > David schrieb: >> Hey, >> >> Here is an expert from the /var/log/messsages : >> >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request - >> M=PUBLISH RURI=sip:group1.105@mydomain.tld >> F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld >> IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish >> rewritten forwarded for ext=group1.105 >> ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml >> version="1.0"?> <dialog-info >> xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" >> entity="sip:group1.105@mydomain.tld"> <dialog >> id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> local-tag="4fbrxu9548" remote-tag="as4fb63283" >> direction="recipient"> <state>early</state> <remote> >> <identity>sip:103@mydomain.tld</identity> <target >> uri="sip:103@mydomain.tld"/> </remote> <local> >> <identity>sip:group1.105@mydomain.tld</identity> <target >> uri="sip:group1.105@mydomain.tld"/> </local> </dialog> >> </dialog-info> >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request - >> M=PUBLISH RURI=sip:group1.105@mydomain.tld >> F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld >> IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish >> accepted for ext=group1.105 ruri=sip:group1.105@mydomain.tld >> duri=<null> body=<?xml version="1.0"?> <dialog-info >> xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" >> entity="sip:105@mydomain.tld"> <dialog >> id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" >> local-tag="4fbrxu9548" remote-tag="as4fb63283" >> direction="recipient"> <state>early</state> <remote> >> <identity>sip:103@mydomain.tld</identity> <target >> uri="sip:103@mydomain.tld"/> </remote> <local> >> <identity>sip:105@mydomain.tld</identity> <target >> uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info> >> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: >> INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld >> via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld >> for event dialog >> >> But the phone's syslog shows this : >> >> Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 >> Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 >> Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 >> SIP/2.0^M Via: SIP/2.0/UDP >> 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To: >> sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From: >> sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M >> CSeq: 12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M >> Content-Length: 655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M >> Max-Forwards: 70^M Event: dialog^M Contact: <sip:mydomain.tld:5060>^M >> Subscription-State: active;expires=1470^M Content-Type: >> application/dialog-info+xml^M ^M <?xml version="1.0"?> <dialog-info >> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" >> state="full" entity="group1.105(a)mydomain.tld"> <dialog >> id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" >> call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" >> local-tag="loto7h3i9a" remote-tag="as4f284f05" >> direction="recipient"> <state>early</state> <remote> >> <identity>sip:103@mydomain.tld</identity> <target uri=" >> >> Everything is good, but notice the entity is somehow back to group1.105. >> >> Where do you want me to run the traceroute? On the server? >> >> David >> >> Klaus Darilion wrote: >>> >>> >>> David schrieb: >>>> Hey, >>>> >>>> I tried that, but it did not seem to change anything. I saw the >>>> message change on the second loop, but it seems that the presence >>>> module figure out my tactic and fixed the message. >>> >>> That's not possible. The presence module does not alter the received >>> body. I think the problem is somewhere else. >>> >>> Can you send me a trace? >>> >>> ngrep -t -P "" -W byline -d any port 5060 >>> >>> regards >>> klaus >>> >>> >>>> >>>> David >>>> >>>> Klaus Darilion wrote: >>>>> That does not work: presence_server always uses the original >>>>> received body, the modifications applied by you are not seen by >>>>> the presence server. >>>>> >>>>> The workaround would be to "loop" the modified PUBLISH to Kamailio >>>>> again, e.g.: >>>>> >>>>> listen=.......:5060 >>>>> listen=.......:5050 >>>>> >>>>> route{ >>>>> ... >>>>> if (is_method("PUBLISH")) { >>>>> if (dst_port==5060) { >>>>> subst_body('/group1[.]//'); >>>>> t_relay("udp:127.0.0.1:5050); >>>>> } else if (dst_port==5050) { >>>>> handle_publish(); >>>>> t_release(); >>>>> } >>>>> } >>>>> ... >>>>> } >>>>> >>>>> >>>>> klaus >>>>> David schrieb: >>>>>> Hey, >>>>>> >>>>>> I tried doing a subst or replace_body_all to change the entity="" >>>>>> value, >>>>>> >>>>>> if (is_method("PUBLISH") ) >>>>>> { >>>>>> subst_body('/group1[.]//'); >>>>>> handle_publish(); >>>>>> t_release(); >>>>>> } >>>>>> >>>>>> It did not work, the $rb variable still shows the old body and >>>>>> the NOTIFY received on the phone is the old body as well. >>>>>> >>>>>> I thought about doing a string replace on the outbound NOTIFY but >>>>>> it never hits my routing script, so I can't. >>>>>> >>>>>> I read textops and it does not list any limitations relating to me. >>>>>> >>>>>> Any ideas? >>>>>> >>>>>> David >>>>>> >>>>>> Klaus Darilion wrote: >>>>>>> I think the problem lies within the dialog module. Dialog module >>>>>>> stores from/to URI. Both, from and to URI are derived from the >>>>>>> headers. We could extend the dialog module to store request-URI >>>>>>> in the dialog-info structure too, and add an option to >>>>>>> pua_dialoginfo to use RURI instead of To-URI. >>>>>>> >>>>>>> regards >>>>>>> klaus >>>>>>> >>>>>>> David schrieb: >>>>>>>> Hey, >>>>>>>> >>>>>>>> I think I am doing something wrong, because here is the NOTIFY >>>>>>>> that comes to the phone : >>>>>>>> >>>>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0 >>>>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0 >>>>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d >>>>>>>> From: >>>>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>>>> >>>>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104 >>>>>>>> Content-Length: 298 >>>>>>>> User-Agent: MyServer 1.0 >>>>>>>> Max-Forwards: 70 >>>>>>>> Event: dialog >>>>>>>> Contact: <sip:my.kamailio.domain.name:5060> >>>>>>>> Subscription-State: active;expires=1570 >>>>>>>> Content-Type: application/dialog-info+xml >>>>>>>> >>>>>>>> <?xml version="1.0"?> <dialog-info >>>>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" >>>>>>>> version="11" state="full" >>>>>>>> entity="group1.102(a)my.kamailio.domain.name"> <dialog >>>>>>>> id="7aaa32e84362a246716009175ad670be(a)domain.tld" >>>>>>>> direction="recipient"> <state>early</state> </dialog> >>>>>>>> </dialog-info> >>>>>>>> >>>>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + >>>>>>>> SPA932) does not flash lights or anything. Since you suggested >>>>>>>> that a solid kamailio would work out of the box, I suspect that >>>>>>>> either I miscommunicated my setup or did something really >>>>>>>> wrong. The notify definitely gets to the phone and the phone >>>>>>>> replies 200/OK when it receives the NOTIFY. But I think that >>>>>>>> the telephone is not understanding the request because it >>>>>>>> subscribed to '102' but received a notification for >>>>>>>> 'group1.102'. The funny thing is, the From header matches the >>>>>>>> subscribe To header, it's just the XML has the full name >>>>>>>> instead of the shortened name. >>>>>>>> >>>>>>>> To: >>>>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>>>> >>>>>>>> >>>>>>>> I see this in my ( on kamailio ) /var/log/messages : >>>>>>>> >>>>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]: >>>>>>>> INFO:presence:send_notify_request: NOTIFY >>>>>>>> sip:group1.101@my.kamailio.domain.name via >>>>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of >>>>>>>> sip:102@my.kamailio.domain.name for event dialog >>>>>>>> >>>>>>>> I should also mention that the NOTIFY sent out by presence >>>>>>>> bypasses my routing scripts. So I have the PUBLISH come through >>>>>>>> ( which I leave alone ) and the NOTIFY is sent according to the >>>>>>>> location table without ever consulting my routing script. >>>>>>>> >>>>>>>> So everything amazingly worked out well, except that the lights >>>>>>>> are not changing status which I think is related to the XML >>>>>>>> document dialog-info entity attribute containing the group >>>>>>>> name, sent from my server. >>>>>>>> >>>>>>>> Any ideas or suggestions ? >>>>>>>> >>>>>>>> Thanks, >>>>>>>> >>>>>>>> David >>>>>>>> >>>>>>>> Other information : >>>>>>>> >>>>>>>> if ( is_method("SUBSCRIBE") ) >>>>>>>> { >>>>>>>> avp_db_query("select groupname from sometable ", >>>>>>>> "$avp(s:groupname)"); >>>>>>>> avp_printf("$ru", >>>>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name"); >>>>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to >>>>>>>> $ru - M=$rm \n"); >>>>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU - >>>>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>>>>>>> handle_subscribe(); >>>>>>>> t_release(); >>>>>>>> exit; >>>>>>>> } >>>>>>>> >>>>>>>> Kamailio compiled from sources : >>>>>>>> >>>>>>>> Path: /usr/src/kamailio >>>>>>>> URL: >>>>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 >>>>>>>> Repository Root: >>>>>>>> https://openser.svn.sourceforge.net/svnroot/openser >>>>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 >>>>>>>> Revision: 5910 >>>>>>>> Node Kind: directory >>>>>>>> Schedule: normal >>>>>>>> Last Changed Author: henningw >>>>>>>> Last Changed Rev: 5910 >>>>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009) >>>>>>>> >>>>>>>> >>>>>>>> Klaus Darilion wrote: >>>>>>>>> notifies are generated by the presence module, based on the >>>>>>>>> subscription. So, there is nothing you have to do with NOTIFY. >>>>>>>>> Also PUBLISHs are created internally. Depending on your >>>>>>>>> concrete setup (e.g. format of the SIP user names) it should >>>>>>>>> work out of the box. >>>>>>>>> >>>>>>>>> klaus >>>>>>>>> >>>>>>>>> dlublink schrieb: >>>>>>>>>> What about the notifies? >>>>>>>>>> >>>>>>>>>> Klaus Darilion wrote: >>>>>>>>>>> dlublink wrote: >>>>>>>>>>>> Hello, >>>>>>>>>>>> >>>>>>>>>>>> I have three different groups of extensions on my kamailio >>>>>>>>>>>> I want to be able to separate them, so I prefixed a name to >>>>>>>>>>>> the extensions, so I have : >>>>>>>>>>>> >>>>>>>>>>>> 1. group1.101 >>>>>>>>>>>> 2. group1.102 >>>>>>>>>>>> 3. group2.101 >>>>>>>>>>>> 4. group2.102 >>>>>>>>>>>> 5. group3.102 >>>>>>>>>>>> 6. group3.103. >>>>>>>>>>>> >>>>>>>>>>>> The phones from different groups can not call each other, I >>>>>>>>>>>> found a pseudo variable that I use to rewrite the >>>>>>>>>>>> destination url, so if user group1.101 dials 102 I rewrite >>>>>>>>>>>> it to group1.102. >>>>>>>>>>>> >>>>>>>>>>>> I want to do the same thing for presence_dialog info, how >>>>>>>>>>>> can I rewrite the subscribe, presence and and notify >>>>>>>>>>>> messages to append the appropriate prefix ? >>>>>>>>>>> >>>>>>>>>>> Just apply the same rewrite you have already done for the >>>>>>>>>>> INVITE also for the SUBSCRIBE >>>>>>>>>>> >>>>>>>>>>> regards >>>>>>>>>>> klaus >>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Thanks, >>>>>>>>>>>> >>>>>>>>>>>> David >>>>>>>>>>>> >>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>> Kamailio (OpenSER) - Users mailing list >>>>>>>>>>>> Users(a)lists.kamailio.org >>>>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>>>>>>>> >>>>>>>>>> >>>>>>>> >>>>>> >>>> >>

2 1

[Serusers] 200 OK not being ACKnowledged
by Steven C. Blair 14 Aug '09

14 Aug '09

I'm testing SIP trunking connectivity to an ITSP through a Covergence (CXC)/ACME Packet session border controller. Calls from an IP phone to the PSTN through SER and the CXC are being dropped 15 seconds after the initial call is setup and just after a cal l is answered. We do not have this problem with calls which reach the PSTN via local gateways. It seems that SER is not relaying an ACK which it receives from the phone which causes the SBC to resend a 200 OK message until finally the connection times out. I've checked the call routing and do not see any issue however I do notice the following in the SER logs. Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: --ACK--: Time: [Thu Aug 13 13:07:24 2009] From: <sip:89390@vidar.net.isc.upenn.edu> To <sip:92155738396@vidar.net.isc.upenn.edu;user=phone> Src IP <128.91.2.223> Method:<ACK> Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: [Thu Aug 13 13:07:24 2009] [453a4bbf-90848c86-ea3a8091(a)128.91.56.18] <sip:92155738396@vidar.net.isc.upenn.edu:5060;user=phone;maddr=128.91.56.242;transport=udp> Starting PRESENCE/BLA section Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: [Thu Aug 13 13:07:24 2009] [453a4bbf-90848c86-ea3a8091(a)128.91.56.18] <sip:92155738396@vidar.net.isc.upenn.edu:5060;user=phone;maddr=128.91.56.242;transport=udp> End PRESENCE/BLA section Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: [Thu Aug 13 13:07:24 2009] [453a4bbf-90848c86-ea3a8091(a)128.91.56.18] --URI match--. Method: <ACK> R-uri: <sip:92155738396@vidar.net.isc.upenn.edu:5060;user=phone;maddr=128.91.56.242;transport=udp> Contact Header: <<sip:89390@128.91.56.18:5060>> From: <sip:89390@vidar.net.isc.upenn.edu> To <sip:92155738396@vidar.net.isc.upenn.edu;user=phone> IP source address <128.91.2.223> Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: --ACK--: Time: [Thu Aug 13 13:07:24 2009] From: <sip:89390@vidar.net.isc.upenn.edu> To <sip:92155738396@vidar.net.isc.upenn.edu;user=phone> Src IP <128.91.2.223> Method:<ACK> Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: [SER]: Too many hops Aug 13 13:07:24 vidar /usr/local/sbin/ser[19053]: Warning: sl_send_reply: I won't send a reply for ACK!! User 89390 is a local subscriber and the phone initiating the call. The recipient of the call is my desk phone 215-573-8396 which is on a different, unrelated SER service. We do record_route traffic through SER for call control and accounting reasons. It seems like the ACK is looping through SER and therefore never being sent to the SBC but I cannot see why. Does anyone have any thoughts on this issue? Thanks,Steve

3 5

Re: [Kamailio-Users] presence_dialoginfo context
by David 14 Aug '09

14 Aug '09

Hey, Apparently the list did not accept the previous message... Here is the code you suggested with my modifications, it'll help put the log extracts in context : else if ( dst_port == 5060 ) { xlog("L_INFO", "Publish rewritten forwarded for ext=$tU ruri=$ru duri=$du body=$rb"); subst_body('/group1[.]//g' ) ; t_relay("udp:127.0.0.1:5050"); exit ; } else { xlog("L_INFO", "Publish accepted for ext=$tU ruri=$ru duri=$du body=$rb"); handle_publish(); t_release(); exit; } Here is an expert from the /var/log/messsages : Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish rewritten forwarded for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:group1.105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:group1.105@mydomain.tld</identity> <target uri="sip:group1.105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish accepted for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:105@mydomain.tld</identity> <target uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld for event dialog But the phone's syslog shows this : Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 SIP/2.0^M Via: SIP/2.0/UDP 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To: sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From: sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M CSeq: 12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M Content-Length: 655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M Max-Forwards: 70^M Event: dialog^M Contact: <sip:mydomain.tld:5060>^M Subscription-State: active;expires=1470^M Content-Type: application/dialog-info+xml^M ^M <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="group1.105(a)mydomain.tld"> <dialog id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" local-tag="loto7h3i9a" remote-tag="as4f284f05" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri=" Everything is good, but notice the entity is somehow back to group1.105. Where do you want me to run the traceroute? On the server? David Klaus Darilion wrote: > > > David schrieb: >> Hey, >> >> I tried that, but it did not seem to change anything. I saw the >> message change on the second loop, but it seems that the presence >> module figure out my tactic and fixed the message. > > That's not possible. The presence module does not alter the received > body. I think the problem is somewhere else. > > Can you send me a trace? > > ngrep -t -P "" -W byline -d any port 5060 > > regards > klaus > > >> >> David >> >> Klaus Darilion wrote: >>> That does not work: presence_server always uses the original >>> received body, the modifications applied by you are not seen by the >>> presence server. >>> >>> The workaround would be to "loop" the modified PUBLISH to Kamailio >>> again, e.g.: >>> >>> listen=.......:5060 >>> listen=.......:5050 >>> >>> route{ >>> ... >>> if (is_method("PUBLISH")) { >>> if (dst_port==5060) { >>> subst_body('/group1[.]//'); >>> t_relay("udp:127.0.0.1:5050); >>> } else if (dst_port==5050) { >>> handle_publish(); >>> t_release(); >>> } >>> } >>> ... >>> } >>> >>> >>> klaus >>> David schrieb: >>>> Hey, >>>> >>>> I tried doing a subst or replace_body_all to change the entity="" >>>> value, >>>> >>>> if (is_method("PUBLISH") ) >>>> { >>>> subst_body('/group1[.]//'); >>>> handle_publish(); >>>> t_release(); >>>> } >>>> >>>> It did not work, the $rb variable still shows the old body and the >>>> NOTIFY received on the phone is the old body as well. >>>> >>>> I thought about doing a string replace on the outbound NOTIFY but >>>> it never hits my routing script, so I can't. >>>> >>>> I read textops and it does not list any limitations relating to me. >>>> >>>> Any ideas? >>>> >>>> David >>>> >>>> Klaus Darilion wrote: >>>>> I think the problem lies within the dialog module. Dialog module >>>>> stores from/to URI. Both, from and to URI are derived from the >>>>> headers. We could extend the dialog module to store request-URI in >>>>> the dialog-info structure too, and add an option to pua_dialoginfo >>>>> to use RURI instead of To-URI. >>>>> >>>>> regards >>>>> klaus >>>>> >>>>> David schrieb: >>>>>> Hey, >>>>>> >>>>>> I think I am doing something wrong, because here is the NOTIFY >>>>>> that comes to the phone : >>>>>> >>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0 >>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0 >>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d >>>>>> From: >>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104 >>>>>> Content-Length: 298 >>>>>> User-Agent: MyServer 1.0 >>>>>> Max-Forwards: 70 >>>>>> Event: dialog >>>>>> Contact: <sip:my.kamailio.domain.name:5060> >>>>>> Subscription-State: active;expires=1570 >>>>>> Content-Type: application/dialog-info+xml >>>>>> >>>>>> <?xml version="1.0"?> <dialog-info >>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" >>>>>> state="full" entity="group1.102(a)my.kamailio.domain.name"> >>>>>> <dialog id="7aaa32e84362a246716009175ad670be(a)domain.tld" >>>>>> direction="recipient"> <state>early</state> </dialog> >>>>>> </dialog-info> >>>>>> >>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + SPA932) >>>>>> does not flash lights or anything. Since you suggested that a >>>>>> solid kamailio would work out of the box, I suspect that either I >>>>>> miscommunicated my setup or did something really wrong. The >>>>>> notify definitely gets to the phone and the phone replies 200/OK >>>>>> when it receives the NOTIFY. But I think that the telephone is >>>>>> not understanding the request because it subscribed to '102' but >>>>>> received a notification for 'group1.102'. The funny thing is, the >>>>>> From header matches the subscribe To header, it's just the XML >>>>>> has the full name instead of the shortened name. >>>>>> >>>>>> To: >>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> >>>>>> I see this in my ( on kamailio ) /var/log/messages : >>>>>> >>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]: >>>>>> INFO:presence:send_notify_request: NOTIFY >>>>>> sip:group1.101@my.kamailio.domain.name via >>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of >>>>>> sip:102@my.kamailio.domain.name for event dialog >>>>>> >>>>>> I should also mention that the NOTIFY sent out by presence >>>>>> bypasses my routing scripts. So I have the PUBLISH come through ( >>>>>> which I leave alone ) and the NOTIFY is sent according to the >>>>>> location table without ever consulting my routing script. >>>>>> >>>>>> So everything amazingly worked out well, except that the lights >>>>>> are not changing status which I think is related to the XML >>>>>> document dialog-info entity attribute containing the group name, >>>>>> sent from my server. >>>>>> >>>>>> Any ideas or suggestions ? >>>>>> >>>>>> Thanks, >>>>>> >>>>>> David >>>>>> >>>>>> Other information : >>>>>> >>>>>> if ( is_method("SUBSCRIBE") ) >>>>>> { >>>>>> avp_db_query("select groupname from sometable ", >>>>>> "$avp(s:groupname)"); >>>>>> avp_printf("$ru", >>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name"); >>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to $ru >>>>>> - M=$rm \n"); >>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU - >>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>>>>> handle_subscribe(); >>>>>> t_release(); >>>>>> exit; >>>>>> } >>>>>> >>>>>> Kamailio compiled from sources : >>>>>> >>>>>> Path: /usr/src/kamailio >>>>>> URL: >>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 >>>>>> Repository Root: https://openser.svn.sourceforge.net/svnroot/openser >>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 >>>>>> Revision: 5910 >>>>>> Node Kind: directory >>>>>> Schedule: normal >>>>>> Last Changed Author: henningw >>>>>> Last Changed Rev: 5910 >>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009) >>>>>> >>>>>> >>>>>> Klaus Darilion wrote: >>>>>>> notifies are generated by the presence module, based on the >>>>>>> subscription. So, there is nothing you have to do with NOTIFY. >>>>>>> Also PUBLISHs are created internally. Depending on your concrete >>>>>>> setup (e.g. format of the SIP user names) it should work out of >>>>>>> the box. >>>>>>> >>>>>>> klaus >>>>>>> >>>>>>> dlublink schrieb: >>>>>>>> What about the notifies? >>>>>>>> >>>>>>>> Klaus Darilion wrote: >>>>>>>>> dlublink wrote: >>>>>>>>>> Hello, >>>>>>>>>> >>>>>>>>>> I have three different groups of extensions on my kamailio I >>>>>>>>>> want to be able to separate them, so I prefixed a name to the >>>>>>>>>> extensions, so I have : >>>>>>>>>> >>>>>>>>>> 1. group1.101 >>>>>>>>>> 2. group1.102 >>>>>>>>>> 3. group2.101 >>>>>>>>>> 4. group2.102 >>>>>>>>>> 5. group3.102 >>>>>>>>>> 6. group3.103. >>>>>>>>>> >>>>>>>>>> The phones from different groups can not call each other, I >>>>>>>>>> found a pseudo variable that I use to rewrite the destination >>>>>>>>>> url, so if user group1.101 dials 102 I rewrite it to group1.102. >>>>>>>>>> >>>>>>>>>> I want to do the same thing for presence_dialog info, how can >>>>>>>>>> I rewrite the subscribe, presence and and notify messages to >>>>>>>>>> append the appropriate prefix ? >>>>>>>>> >>>>>>>>> Just apply the same rewrite you have already done for the >>>>>>>>> INVITE also for the SUBSCRIBE >>>>>>>>> >>>>>>>>> regards >>>>>>>>> klaus >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> >>>>>>>>>> David >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (OpenSER) - Users mailing list >>>>>>>>>> Users(a)lists.kamailio.org >>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>>>> >>>>>> >>>> >>

1 0

Re: [Kamailio-Users] presence_dialoginfo context
by David 14 Aug '09

14 Aug '09

Hey, Here is an expert from the /var/log/messsages : Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28924]: Publish rewritten forwarded for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:group1.105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:group1.105@mydomain.tld</identity> <target uri="sip:group1.105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: New request - M=PUBLISH RURI=sip:group1.105@mydomain.tld F=sip:group1.105@mydomain.tld T=sip:group1.105@mydomain.tld IP=72.55.182.125 ID=7bddec01-28924(a)72.55.182.125 Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: Publish accepted for ext=group1.105 ruri=sip:group1.105@mydomain.tld duri=<null> body=<?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:105@mydomain.tld"> <dialog id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" call-id="5a9207de129fc16f49cafaa425e312f1(a)mydomain.tld" local-tag="4fbrxu9548" remote-tag="as4fb63283" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri="sip:103@mydomain.tld"/> </remote> <local> <identity>sip:105@mydomain.tld</identity> <target uri="sip:105@mydomain.tld"/> </local> </dialog> </dialog-info> Aug 14 10:16:42 kamailio-dev /usr/sbin/kamailio[28932]: INFO:presence:send_notify_request: NOTIFY sip:group1.101@mydomain.tld via sip:group1.101@phone.ip:57234 on behalf of sip:105@mydomain.tld for event dialog But the phone's syslog shows this : Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys [0:57234]<<server.ip:5060 Aug 14 10:16:21 linksys NOTIFY sip:group1.101@phone.ip:57234 SIP/2.0^M Via: SIP/2.0/UDP 72.55.182.125;branch=z9hG4bK0b0a.9725e23.0^M To: sip:group1.101@mydomain.tld;tag=1c32db3af4d889ac^M From: sip:105@mydomain.tld;tag=6d6077e15fecee48b2721a5e94bf8883-91a8^M CSeq: 12 NOTIFY^M Call-ID: 984d4cd2-37b04f4(a)192.168.1.104^M Content-Length: 655^M User-Agent: Kamailio (1.5.2-tls (i386/linux))^M Max-Forwards: 70^M Event: dialog^M Contact: <sip:mydomain.tld:5060>^M Subscription-State: active;expires=1470^M Content-Type: application/dialog-info+xml^M ^M <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="group1.105(a)mydomain.tld"> <dialog id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" call-id="0b5776fb2ea599c429038414054eb383(a)mydomain.tld" local-tag="loto7h3i9a" remote-tag="as4f284f05" direction="recipient"> <state>early</state> <remote> <identity>sip:103@mydomain.tld</identity> <target uri=" Everything is good, but notice the entity is somehow back to group1.105. Where do you want me to run the traceroute? On the server? David Klaus Darilion wrote: > > > David schrieb: >> Hey, >> >> I tried that, but it did not seem to change anything. I saw the >> message change on the second loop, but it seems that the presence >> module figure out my tactic and fixed the message. > > That's not possible. The presence module does not alter the received > body. I think the problem is somewhere else. > > Can you send me a trace? > > ngrep -t -P "" -W byline -d any port 5060 > > regards > klaus > > >> >> David >> >> Klaus Darilion wrote: >>> That does not work: presence_server always uses the original >>> received body, the modifications applied by you are not seen by the >>> presence server. >>> >>> The workaround would be to "loop" the modified PUBLISH to Kamailio >>> again, e.g.: >>> >>> listen=.......:5060 >>> listen=.......:5050 >>> >>> route{ >>> ... >>> if (is_method("PUBLISH")) { >>> if (dst_port==5060) { >>> subst_body('/group1[.]//'); >>> t_relay("udp:127.0.0.1:5050); >>> } else if (dst_port==5050) { >>> handle_publish(); >>> t_release(); >>> } >>> } >>> ... >>> } >>> >>> >>> klaus >>> David schrieb: >>>> Hey, >>>> >>>> I tried doing a subst or replace_body_all to change the entity="" >>>> value, >>>> >>>> if (is_method("PUBLISH") ) >>>> { >>>> subst_body('/group1[.]//'); >>>> handle_publish(); >>>> t_release(); >>>> } >>>> >>>> It did not work, the $rb variable still shows the old body and the >>>> NOTIFY received on the phone is the old body as well. >>>> >>>> I thought about doing a string replace on the outbound NOTIFY but >>>> it never hits my routing script, so I can't. >>>> >>>> I read textops and it does not list any limitations relating to me. >>>> >>>> Any ideas? >>>> >>>> David >>>> >>>> Klaus Darilion wrote: >>>>> I think the problem lies within the dialog module. Dialog module >>>>> stores from/to URI. Both, from and to URI are derived from the >>>>> headers. We could extend the dialog module to store request-URI in >>>>> the dialog-info structure too, and add an option to pua_dialoginfo >>>>> to use RURI instead of To-URI. >>>>> >>>>> regards >>>>> klaus >>>>> >>>>> David schrieb: >>>>>> Hey, >>>>>> >>>>>> I think I am doing something wrong, because here is the NOTIFY >>>>>> that comes to the phone : >>>>>> >>>>>> NOTIFY sip:group1.101@my.home.ip:57234 SIP/2.0 >>>>>> Via: SIP/2.0/UDP my.kamailio.ip;branch=z9hG4bK3cab.78e494a4.0 >>>>>> To: sip:group1.101@my.kamailio.domain.name;tag=b2514519391e8b4d >>>>>> From: >>>>>> sip:102@my.kamailio.domain.name;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> CSeq: 12 NOTIFY^M Call-ID: 24db3209-611c031d(a)192.168.1.104 >>>>>> Content-Length: 298 >>>>>> User-Agent: MyServer 1.0 >>>>>> Max-Forwards: 70 >>>>>> Event: dialog >>>>>> Contact: <sip:my.kamailio.domain.name:5060> >>>>>> Subscription-State: active;expires=1570 >>>>>> Content-Type: application/dialog-info+xml >>>>>> >>>>>> <?xml version="1.0"?> <dialog-info >>>>>> xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" >>>>>> state="full" entity="group1.102(a)my.kamailio.domain.name"> >>>>>> <dialog id="7aaa32e84362a246716009175ad670be(a)domain.tld" >>>>>> direction="recipient"> <state>early</state> </dialog> >>>>>> </dialog-info> >>>>>> >>>>>> The phone ( I tested with a GXP2000, GXP2020 and SPA962 + SPA932) >>>>>> does not flash lights or anything. Since you suggested that a >>>>>> solid kamailio would work out of the box, I suspect that either I >>>>>> miscommunicated my setup or did something really wrong. The >>>>>> notify definitely gets to the phone and the phone replies 200/OK >>>>>> when it receives the NOTIFY. But I think that the telephone is >>>>>> not understanding the request because it subscribed to '102' but >>>>>> received a notification for 'group1.102'. The funny thing is, the >>>>>> From header matches the subscribe To header, it's just the XML >>>>>> has the full name instead of the shortened name. >>>>>> >>>>>> To: >>>>>> <sip:102@my.kamailio.domain.name>;tag=6d6077e15fecee48b2721a5e94bf8883-f712 >>>>>> >>>>>> >>>>>> I see this in my ( on kamailio ) /var/log/messages : >>>>>> >>>>>> Aug 13 09:55:24 kamailio-dev /usr/sbin/kamailio[25449]: >>>>>> INFO:presence:send_notify_request: NOTIFY >>>>>> sip:group1.101@my.kamailio.domain.name via >>>>>> sip:group1.101@my.home.ip.where.phone.is:57234 on behalf of >>>>>> sip:102@my.kamailio.domain.name for event dialog >>>>>> >>>>>> I should also mention that the NOTIFY sent out by presence >>>>>> bypasses my routing scripts. So I have the PUBLISH come through ( >>>>>> which I leave alone ) and the NOTIFY is sent according to the >>>>>> location table without ever consulting my routing script. >>>>>> >>>>>> So everything amazingly worked out well, except that the lights >>>>>> are not changing status which I think is related to the XML >>>>>> document dialog-info entity attribute containing the group name, >>>>>> sent from my server. >>>>>> >>>>>> Any ideas or suggestions ? >>>>>> >>>>>> Thanks, >>>>>> >>>>>> David >>>>>> >>>>>> Other information : >>>>>> >>>>>> if ( is_method("SUBSCRIBE") ) >>>>>> { >>>>>> avp_db_query("select groupname from sometable ", >>>>>> "$avp(s:groupname)"); >>>>>> avp_printf("$ru", >>>>>> "sip:$avp(s:zone).$tU@my.kamailio.domain.name"); >>>>>> xlog("L_INFO", "Subscribe rewritten from $tU to $ru >>>>>> - M=$rm \n"); >>>>>> xlog("L_INFO", "Handling SUBSCRIBE - $fU - >>>>>> $avp(s:zone) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>>>>> handle_subscribe(); >>>>>> t_release(); >>>>>> exit; >>>>>> } >>>>>> >>>>>> Kamailio compiled from sources : >>>>>> >>>>>> Path: /usr/src/kamailio >>>>>> URL: >>>>>> https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 >>>>>> Repository Root: https://openser.svn.sourceforge.net/svnroot/openser >>>>>> Repository UUID: 689a6050-402a-0410-94f2-e92a70836424 >>>>>> Revision: 5910 >>>>>> Node Kind: directory >>>>>> Schedule: normal >>>>>> Last Changed Author: henningw >>>>>> Last Changed Rev: 5910 >>>>>> Last Changed Date: 2009-08-06 13:08:30 -0400 (Thu, 06 Aug 2009) >>>>>> >>>>>> >>>>>> Klaus Darilion wrote: >>>>>>> notifies are generated by the presence module, based on the >>>>>>> subscription. So, there is nothing you have to do with NOTIFY. >>>>>>> Also PUBLISHs are created internally. Depending on your concrete >>>>>>> setup (e.g. format of the SIP user names) it should work out of >>>>>>> the box. >>>>>>> >>>>>>> klaus >>>>>>> >>>>>>> dlublink schrieb: >>>>>>>> What about the notifies? >>>>>>>> >>>>>>>> Klaus Darilion wrote: >>>>>>>>> dlublink wrote: >>>>>>>>>> Hello, >>>>>>>>>> >>>>>>>>>> I have three different groups of extensions on my kamailio I >>>>>>>>>> want to be able to separate them, so I prefixed a name to the >>>>>>>>>> extensions, so I have : >>>>>>>>>> >>>>>>>>>> 1. group1.101 >>>>>>>>>> 2. group1.102 >>>>>>>>>> 3. group2.101 >>>>>>>>>> 4. group2.102 >>>>>>>>>> 5. group3.102 >>>>>>>>>> 6. group3.103. >>>>>>>>>> >>>>>>>>>> The phones from different groups can not call each other, I >>>>>>>>>> found a pseudo variable that I use to rewrite the destination >>>>>>>>>> url, so if user group1.101 dials 102 I rewrite it to group1.102. >>>>>>>>>> >>>>>>>>>> I want to do the same thing for presence_dialog info, how can >>>>>>>>>> I rewrite the subscribe, presence and and notify messages to >>>>>>>>>> append the appropriate prefix ? >>>>>>>>> >>>>>>>>> Just apply the same rewrite you have already done for the >>>>>>>>> INVITE also for the SUBSCRIBE >>>>>>>>> >>>>>>>>> regards >>>>>>>>> klaus >>>>>>>>> >>>>>>>>>> >>>>>>>>>> Thanks, >>>>>>>>>> >>>>>>>>>> David >>>>>>>>>> >>>>>>>>>> _______________________________________________ >>>>>>>>>> Kamailio (OpenSER) - Users mailing list >>>>>>>>>> Users(a)lists.kamailio.org >>>>>>>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>>>>>>>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>>>>>>> >>>>>> >>>> >>

1 0

[Kamailio-Users] Generate INVITE from Openser
by ALAEDDINE abbech 14 Aug '09

14 Aug '09

Hello all How can i generate an INVITE from OpenSER. That means that Openser make the role of UAC. Who have an idea for this pb. Thanks Regards Aladin

6 9

[Kamailio-Users] presence_dialoginfo context
by dlublink 14 Aug '09

14 Aug '09

Hello, I have three different groups of extensions on my kamailio I want to be able to separate them, so I prefixed a name to the extensions, so I have : 1. group1.101 2. group1.102 3. group2.101 4. group2.102 5. group3.102 6. group3.103. The phones from different groups can not call each other, I found a pseudo variable that I use to rewrite the destination url, so if user group1.101 dials 102 I rewrite it to group1.102. I want to do the same thing for presence_dialog info, how can I rewrite the subscribe, presence and and notify messages to append the appropriate prefix ? Thanks, David

4 10

[Kamailio-Users] Load Balancing Asterisk using Kamailio
by Kurt Weasel 14 Aug '09

14 Aug '09

Hi, I searched around the web to load balance asterisk servers and found Kamailio for possible solution. Let's say I have two identical asterisk servers with same dialplan and configuration and I want both servers look like they have same IP address from clients. Based on my understanding, we need 1 Kamailio as load balancer and 2 Asterisk servers as the real servers. Let's say the setup is : Kamailio load balancer 192.168.2.1 Asterisk Server #1 192.168.2.2 Asterisk Server #2 192.168.2.3 My question is, X-Lite softphone Configuration should be set to domain 192.168.2.1, right? I also want to know the step by step configuration to set kamailio as load balancer. I have not used Kamailio before. However after searching the documentation, the step (based on my understanding) is somewhat like this : 1. Install Kamailio. I will use the step by step here : http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn 2. Then, using dipatcher module, I will configure it using this guide here : http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha Then modify the dispatcher.list file to match the IP address of my asterisk servers : *1 sip:192.168.2.2:5060 1 sip:192.168.2.3:5060* Am I missing some steps? Do I also need to configure dialplan or any other file at Kamailio load balancer? Or those two steps basically done it all for simple load balancing configuration? Thanks for your responses. Best Regard, Kurt Weasel.

5 8

[Kamailio-Users] OpenSER without MYSQL
by Ratheendran R 13 Aug '09

13 Aug '09

Hello everyone, I want to port openSER on an embedded board,I need configure the openser without MySQL due to space constraint. So can you let me know whether this is possible also can you give any reference document on how to port on ARM 9 based boad. Regards, Rathndran <users(a)lists.kamailio.org>

7 9

[Kamailio-Users] Fwd: dialog info + Grandstreams = freezing
by Daniel-Constantin Mierla 13 Aug '09

13 Aug '09

.. fwd to list ... reply all button mismatch ... ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed, Aug 12, 2009 at 9:20 AM Subject: Re: [Kamailio-Users] dialog info + Grandstreams = freezing To: David <kamailio.org(a)spam.lublink.net> Hello, On Tue, Aug 11, 2009 at 2:18 PM, David <kamailio.org(a)spam.lublink.net>wrote: > Hey, > > I am not good at diagrams, but here goes : > > > [ Asterisk] [ Asterisk ] ================> [ MySQL ] > / \ / \ > [ Kamailio ] ============/\ > / \ / \ / \ > [ Extension ] [ Extension ] [ Extension ] > > Here is a simple flow : > > 1. Extension 1 is plugged in > 2. It registers to Kamailio and Kamailio stores to Location > 3. Extension 1 subscribes to all the other extensions > 4. Kamailio calls handle_publish() and t_release() > 5. Extension 1 dials '102' > 6. Kamailio receives the INVITE chooses an asterisk server and relays the > INVITE ( t_relay ) > 7. Asterisk does some checking, starts monitor ( if needed ) and some other > stuff > 8. Asterisk sends INVITE to Kamailio for testspace.102 > 9. Kamailio finds testspace.102 in database and forwards INVITE to the > appropriate extension > 10. Extension replies 'OK' > 11. Kamailio, using reply-route, sends it back to asterisk > 12. Asterisk bridges the call with 101 > 13. Asterisk sends an OK back through Kamailio to Extension 1. > 14. Asterisk tries to reinvite the audio ( if not monitoring ) > 15. Audio bypasses Asterisk, but the SIP still is making the full trip. > > Does this answer your question? Do you do call tracking with dialog for both legs (inbond: phone to asterisk; outbound: asterisk to phone)? You should do it for one, outbound is the best, in case asterisk auto-answers the inbound leg. Cheers, Daniel > > David > > > > Daniel-Constantin Mierla wrote: > >> Hello, >> >> I am using the module with snom 370 and works ok. The BLF is on for >> Polycoms but they do not implement the call pickup properly (they have >> direct call pickup as far as I could get). >> >> Please take latest kamailio 1.5 from svn branch 1.5 as there were >> committed some updates (maybe they don't affect you, being mainly to rls, >> but is better to have the latest stable so we refer to same source code). >> >> What would be good is to have a diagram of the signaling, since you say >> there are lot of messages sent around. >> http://callflow.sourceforge.net/ >> >> Cheers, >> Daniel >> >> >> On 11.08.2009 5:03 Uhr, David wrote: >> >>> Hello, >>> >>> I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, >>> 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS >>> 1.5.2-tls compilde on Ubuntu 8.04 >>> >>> I am trying to setup presence_dialoginfo with my Grandstreams, Snom and >>> Linksys. I have a 4 phones on the server. >>> >>> 101 - Linksys SPA962 ( 6.1.5a ) >>> 102 - Grandstream GXP2020 ( 1.2.1.4 ) >>> 103 - Grandstream GXP2000 ( 1.2.1.4 ) >>> 104 - Grandstream GXP2000 ( 1.1.6.46 ) >>> 105 - Snom 360 ( 7.3.23 ) >>> >>> My Kamailio deals with registrations, NAT and BLF everything else is sent >>> to one of two asterisk boxes. I use the dispatcher module for this. This >>> means that when I call one extension to the other, both call legs from >>> asterisk are going through Kamailio as separate calls. But to divide my >>> customers, the usernames are different from the URL that the user types. For >>> example the customer dials '101' but it is changed to testspace.101 when it >>> comes back from asterisk. So Kamailio would have two calls in the event that >>> 101 dials 102. >>> >>> sip:testspace.101@myserver to 102 ( this is sent to asterisk ) >>> sip:testspace.101@myserver to testspace.102 ( this is coming back from >>> asterisk ) >>> >>> Something is horribly wrong. I have the following problems : >>> >>> 1. If 102 calls 103, when 103 answers both phones hang for about 2 >>> minutes >>> 2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on >>> the Linksys and dark on the Snom), but the orange light stays on on the Snom >>> and it thinks the call is still active ( the light is on, but the call is >>> over ) >>> 3. If any extension calls any extension and I try a call pickup, it >>> fails. It looks like the Linksys is sending a NOTIFY to pickup the call ( I >>> thought it was supposed to send an invite... ? ) >>> >>> Looking at the logs it looks like Kamailio is sending out so many NOTIFYs >>> that it is crashing the Grandstreams, and causing the Snom to act funny. >>> >>> Here are some experts from my config file : >>> >>> root@kamailio-dev:/etc/kamailio# grep dialog * >>> kamailio.cfg:# * avp value for dialogs is still not correct >>> kamailio.cfg:loadmodule "dialog.so" >>> kamailio.cfg:loadmodule "presence_dialoginfo.so" >>> kamailio.cfg:loadmodule "pua_dialoginfo.so" >>> kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0) >>> kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0) >>> kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0) >>> kamailio.cfg:modparam("dialog", "dlg_flag", 4) >>> kamailio.cfg:modparam("dialog", "db_mode", 1) >>> kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)") # I still >>> haven't figured out how to set $avp(i:10) >>> kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300) >>> kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog", 1) >>> kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1) >>> kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog", >>> "db_url", >>> kamailio.cfg:# Flag 4 = Mark the current request for a dialog >>> kamailio.cfg: # sequential request withing a dialog should >>> >>> the set flag looks like this : >>> if ( ds_is_from_list() ) >>> { >>> xlog("L_INFO", "Coming from asterisk"); >>> if ( is_method("INVITE")) >>> { >>> setflag(4); >>> } >>> } >>> >>> So the dialog flag is only set for the leg coming back from asterisk. >>> >>> When a notify comes in : >>> >>> if(is_method("NOTIFY") ) >>> { >>> if (! t_newtran()) >>> { >>> sl_reply_error(); >>> exit; >>> }; >>> >>> t_reply("200", "OK"); >>> t_release(); >>> exit ; >>> } >>> >>> Publish and subscribe are like this : >>> >>> >>> if( is_method("PUBLISH") || is_method("SUBSCRIBE") ) >>> { >>> route(5); >>> exit; >>> } >>> >>> route[5] >>> { >>> # absorb retransmissions >>> if (! t_newtran()) >>> { >>> xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition - >>> M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >>> sl_reply_error(); >>> exit; >>> }; >>> >>> append_to_reply("Contact: <sip:myserver.tld:5060>\r\n"); >>> >>> if(is_method("PUBLISH")) >>> { >>> handle_publish(); >>> t_release(); >>> >>> } else if( is_method("SUBSCRIBE")) { >>> handle_subscribe(); >>> t_release(); >>> } >>> else >>> { >>> } >>> >>> exit; >>> } >>> >>> I also have NAT checking for those telephones where stun isn't enough. >>> Before I reach publish/subscribe/invite/notify, I also call setbflag() and >>> sometimes call fix_nated_contact(). Additionnally, I have a block if code >>> before my presence stuff if ( has_totag() && loose_route()) { t_relay(); }. >>> >>> If sip.conf:canreinvite=yes, the grandstreams freeze so long that the >>> server times out, and the BLFs get really messed up. if sip:canreinvite=no >>> the grandstreams only freeze for about 30 seconds. >>> >>> Obviously I am doing something wrong, but despite having searched google >>> for endless hours, and poured over documentation, I can not seem to find >>> what I did wrong. >>> >>> I would really appreciate if someone could shed light on my problem. >>> >>> I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, >>> 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS >>> 1.5.2-tls compilde on Ubuntu 8.04 >>> >>> Thanks, >>> >>> David >>> >>> _______________________________________________ >>> Kamailio (OpenSER) - Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >> > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

2 2

[Kamailio-Users] acc doesn't write to database and to a log file
by Koffedor 13 Aug '09

13 Aug '09

Hello to all I enabled the acc module but it isn't writing to "acc" or "missed_calls" table in mysql, and to a log file too. my config: (...) loadmodule "db_mysql.so" loadmodule "tm.so" loadmodule "rr.so" (...) # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://xxx:xxxx@localhost/kamailio") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") Flags 1 and 2 set just like in a sample configuration file. (...) if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); (...) # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } Mysql tables are created: | acc | CREATE TABLE `acc` ( `id` int(10) unsigned NOT NULL auto_increment, `method` varchar(16) NOT NULL default '', `from_tag` varchar(64) NOT NULL default '', `to_tag` varchar(64) NOT NULL default '', `callid` varchar(128) NOT NULL default '', `sip_code` char(3) NOT NULL default '', `sip_reason` varchar(32) NOT NULL default '', `time` datetime NOT NULL default '0000-00-00 00:00:00', `src_ip` varchar(64) NOT NULL default '', `dst_user` varchar(64) NOT NULL default '', `dst_domain` varchar(128) NOT NULL default '', `src_user` varchar(64) NOT NULL default '', `src_domain` varchar(128) NOT NULL default '', `cdr_id` int(11) NOT NULL default '0', `dst_ouser` varchar(64) NOT NULL default '', PRIMARY KEY (`id`), KEY `acc_callid` (`callid`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 | | missed_calls | CREATE TABLE `missed_calls` ( `id` int(10) unsigned NOT NULL auto_increment, `method` varchar(16) NOT NULL default '', `from_tag` varchar(64) NOT NULL default '', `to_tag` varchar(64) NOT NULL default '', `callid` varchar(128) NOT NULL default '', `sip_code` char(3) NOT NULL default '', `sip_reason` varchar(32) NOT NULL default '', `time` datetime NOT NULL default '0000-00-00 00:00:00', `src_ip` varchar(64) NOT NULL default '', `dst_user` varchar(64) NOT NULL default '', `dst_domain` varchar(128) NOT NULL default '', `src_user` varchar(64) NOT NULL default '', `src_domain` varchar(128) NOT NULL default '', `cdr_id` int(11) NOT NULL default '0', `dst_ouser` varchar(64) NOT NULL default '', PRIMARY KEY (`id`), KEY `mc_callid` (`callid`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 There are no errors in log, and nothing about acc module in debug mode. Cant get where is my mistake.

2 3

BUG: tls [tls_server.c:98]: Invalid connection state (bug in TCP code)
by MÉSZÁROS Mihály 12 Aug '09

12 Aug '09

Hello! Another problem I am using sip-router git version with the attached configuration files. I am getting the error mentioned in subject. Please see the debug log and my configuration files. Can anyone help me figure out what i a configured wrong? Misi Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:alice@atlanta.test> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-260c1648321add52-1---d8754z->; state=6 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: PUBLISH sip:alice@atlanta.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-260c1648321add52-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: "alice"<sip:alice@atlanta.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 16 PUBLISH#015#012Expires: 1200#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/pidf+xml#015#012Proxy-Authorization: Digest username="alice",realm="atlanta.test",nonce="4a81babd0000000bdc0c130f92356ce11fc7ea51a926c644",uri="sip:alice@atlanta.test",response="527ee7c53e6f621ef9ee161426da4afd",cnonce="9f418c76f9b91cd9c3d71fbb4d36f415",nc=00000002,qop=auth,algorithm=MD5#015#012SIP-If-Match: a.1250015022.397.2.8#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Event: presence#015#012Content-Length: 517#015#012#015#012<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:alice@atlanta.test'><tuple id='tae2c271a'><status><basic>open</basic></status></tuple><dm:person id='p7a66b539'><rpid:activities><rpid:busy/><rpid:on-the-phone/></rpid:activities><dm:note>On the phone</dm:note></dm:person></presence> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [33]; uri=[sip:alice@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body ["alice"<sip:alice@atlanta.test>#015#012] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=3 global id=2 T start=0xffffffff Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <16> <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: beginning of SIP-If-Match: yet=0x2d706973 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: middle of SIP-If-Match: yet=0x6d2d6669 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: end of SIP-If-Match Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=3518, isACK=0 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=3 global id=3 T end=(nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=6470be16 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [nonce.c:186]: comparing [4a81babd0000000bdc0c130f92356ce11fc7ea51a926c644] and [4a81babd0000000bdc0c130f92356ce11fc7ea51a926c644] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 3 columns returned from the query Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 12 bytes for result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 12 bytes for result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4e4c)[0]=[password] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82dbf88)[1]=[email_address] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[2] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f5768)[2]=[rpid] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 60 bytes for row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [bakker] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:003631001001@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:123]: HA1 string calculated: 67bb4b4b59a70b7f246682dfdd188df2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:239]: our result = '527ee7c53e6f621ef9ee161426da4afd' Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:246]: authorization is OK Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:194]: nonce index= 11 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [index.c:187]: nonce already used Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:198]: nonce index not valid Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 3 columns Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [index.c:128]: second= 16, sec_monit= 8, index= 12 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:107]: nonce index= 12 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:151]: 'Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2", qop="auth"#015#012' Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: sl [sl.c:314]: reply in stateless mode (sl) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 742 bytes written Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=742 fd=8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 407 Proxy Authentication Required#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-260c1648321add52-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: "alice"<sip:alice@atlanta.test>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.a62c#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 16 PUBLISH#015#012Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2", qop="auth"#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:alice@atlanta.test out_uri=sip:alice@atlanta.test via_cnt==1"#015#012#015#012 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:alice@atlanta.test> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-8b6841568301eb55-1---d8754z->; state=6 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:38 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: PUBLISH sip:alice@atlanta.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-8b6841568301eb55-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: "alice"<sip:alice@atlanta.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 17 PUBLISH#015#012Expires: 1200#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/pidf+xml#015#012Proxy-Authorization: Digest username="alice",realm="atlanta.test",nonce="4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2",uri="sip:alice@atlanta.test",response="cc8d63baac587662386d887e507bf60b",cnonce="1eeeeb8bf845982d90f185276c67bc78",nc=00000001,qop=auth,algorithm=MD5#015#012SIP-If-Match: a.1250015022.397.2.8#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Event: presence#015#012Content-Length: 517#015#012#015#012<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:alice@atlanta.test'><tuple id='tae2c271a'><status><basic>open</basic></status></tuple><dm:person id='p7a66b539'><rpid:activities><rpid:busy/><rpid:on-the-phone/></rpid:activities><dm:note>On the phone</dm:note></dm:person></presence> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [33]; uri=[sip:alice@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body ["alice"<sip:alice@atlanta.test>#015#012] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=4 global id=3 T start=(nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <17> <PUBLISH> Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: beginning of SIP-If-Match: yet=0x2d706973 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: middle of SIP-If-Match: yet=0x6d2d6669 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_hname2.c:182]: end of SIP-If-Match Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=517 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=3519, isACK=0 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=4 global id=4 T end=(nil) Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=6470be16 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [nonce.c:186]: comparing [4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2] and [4a81bacc0000000cc8f1349a189042236344c2b25f33aeb2] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 3 columns returned from the query Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 12 bytes for result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 12 bytes for result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4ebc)[0]=[password] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f5768)[1]=[email_address] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[2] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82dbf88)[2]=[rpid] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 60 bytes for row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [bakker] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:003631001001@atlanta.test] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:123]: HA1 string calculated: 67bb4b4b59a70b7f246682dfdd188df2 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:239]: our result = 'cc8d63baac587662386d887e507bf60b' Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:246]: authorization is OK Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:194]: nonce index= 12 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:176]: set string AVP ""/13 = "sip:003631001001@atlanta.test" Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 3 columns Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82f4ebc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82f5768 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82dbf88 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f55d8 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f52fc Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5390 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f4e4c Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82bd628 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: uri_db [checks.c:137]: Digest username and URI username match Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:108]: No Route headers found Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:907]: There is no Route HF Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:38 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1383]: DEBUG: t_newtran: msg id=4 , global msg id=4 , T on entrance=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=3519, isACK=0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_hooks.c:335]: DBG: trans=0xb585090c, callback type 1, id 0 entered Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [event_list.c:349]: start event= [presence/1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [publish.c:369]: SIP-If-Match header found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [publish.c:378]: existing etag = a.1250015022.397.2.8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [publish.c:390]: Expires header found, value= 1200 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82f56e4 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 2 columns returned from the query Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 8 bytes for result names at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 8 bytes for result types at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82f4e84 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4e84)[0]=[body] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:121]: use DB1_BLOB result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82f5730 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f5730)[1]=[sender] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f4e10 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 40 bytes for row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:168]: converting BLOB [<?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' xmlns:lt='urn:ietf:params:xml:ns:location-type' entity='sip:alice@atlanta.test'><tuple id='tae2c271a'><status><basic>open</basic></status></tuple><dm:person id='p7a66b539'><dm:note>Available</dm:note></dm:person></presence>] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 2 columns Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82f4e84 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82f5730 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f4e10 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82f56e4 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [presentity.c:88]: etag= a.1250015022.390.2.9 / 20#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [presentity.c:102]: send 200OK reply Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [presentity.c:103]: etag= a.1250015022.390.2.9 - len= 20 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=4 global id=4 T start=0xb585090c Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1152]: DEBUG: t_check_msg: T already found! Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [mem/shm_mem.c:117]: WARNING:vqm_resize: resize(0) called Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:1355]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 643 bytes written Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=643 fd=8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 200 OK#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-8b6841568301eb55-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: "alice"<sip:alice@atlanta.test>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2965#015#012From: "alice"<sip:alice@atlanta.test>;tag=6470be16#015#012Call-ID: MTM5NDE0YjRjMmU2ZWNlOThjNDI1MGM2ZjAzMDBjZjA.#015#012CSeq: 17 PUBLISH#015#012Expires: 1200#015#012SIP-ETag: a.1250015022.390.2.9#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:alice@atlanta.test out_uri=sip:alice@atlanta.test via_cnt==1"#015#012#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:610]: DEBUG: reply sent out. buf=0x82f5c80: SIP/2.0 2..., shmem=0xb584f674: SIP/2.0 2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:620]: DEBUG: _reply_light: finished Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: sl [sl.c:309]: reply in stateful mode (tm) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [notify.c:1302]: found 0 dialogs( 0 in database and 0 in hash_table) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: presence [notify.c:1321]: Could not find subs_dialog Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z->; state=6 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: INVITE sip:bob@biloxi.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 1 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:bob@biloxi.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>#015#012] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffff Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5d194242 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=47015, isACK=0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=5 global id=5 T end=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:126]: credentials with given realm not found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [index.c:128]: second= 18, sec_monit= 8, index= 13 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:107]: nonce index= 13 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [challenge.c:151]: 'Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0", qop="auth"#015#012' Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: sl [sl.c:314]: reply in stateless mode (sl) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 724 bytes written Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=724 fd=8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 407 Proxy Authentication Required#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d643#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 1 INVITE#015#012Proxy-Authenticate: Digest realm="atlanta.test", nonce="4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0", qop="auth"#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <ACK> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-0b3389781d1b817e-1---d8754z->; state=6 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d643 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [65]; uri=[sip:bob@biloxi.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: sl [sl_funcs.c:327]: local ACK found -> dropping it! Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-2a767f38e361686d-1---d8754z->; state=6 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:39 a1 /usr/sbin/ser[390]: ERROR: <script>: Misi: INVITE sip:bob@biloxi.test SIP/2.0#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport#015#012Max-Forwards: 70#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Proxy-Authorization: Digest username="alice",realm="atlanta.test",nonce="4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0",uri="sip:bob@biloxi.test",response="60ca1ab3ccd52ae04fded05e2056a49a",cnonce="41e9f5aabf671596242bb02343062a1b",nc=00000001,qop=auth,algorithm=MD5#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=9 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:bob@biloxi.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>#015#012] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: siputils [checks.c:65]: no totag Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1080]: DEBUG: t_check_msg: msg id=7 global id=5 T start=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field: cseq <CSeq>: <2> <INVITE> Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_body : content_length=544 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:89]: found end of header Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5d194242 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=47012, isACK=0 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1149]: DEBUG: t_check_msg: msg id=7 global id=7 T end=(nil) Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==9 && [atlanta.test] == [127.0.0.1] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 12==14 && [atlanta.test] == [195.111.158.41] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [nonce.c:186]: comparing [4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0] and [4a81bacd0000000dda4c36dd574a1b80048af63c44e36fd0] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:116]: allocate 28 bytes for result set at 0x82bd628 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:68]: 3 columns returned from the query Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:153]: allocate 12 bytes for result names at 0x82f55d8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:163]: allocate 12 bytes for result types at 0x82f52b8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[0] at 0x82dbf88 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82dbf88)[0]=[password] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[1] at 0x82f4e4c Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4e4c)[1]=[email_address] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:85]: allocate 8 bytes for RES_NAMES[2] at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:92]: RES_NAMES(0x82f4ebc)[2]=[rpid] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: db_mysql [km_res.c:136]: use DB1_STRING result type Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:182]: allocate 8 bytes for rows at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:119]: allocate 60 bytes for row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [bakker] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_val.c:117]: converting STRING [sip:003631001001@atlanta.test] Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:123]: HA1 string calculated: 67bb4b4b59a70b7f246682dfdd188df2 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:239]: our result = '60ca1ab3ccd52ae04fded05e2056a49a' Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:246]: authorization is OK Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth [api.c:194]: nonce index= 13 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: auth_db [authorize.c:176]: set string AVP ""/13 = "sip:003631001001@atlanta.test" Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:79]: freeing 3 columns Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[0] at 0x82dbf88 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[1] at 0x82f4e4c Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:83]: freeing RES_NAMES[2] at 0x82f4ebc Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:92]: freeing result names at 0x82f55d8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:97]: freeing result types at 0x82f52b8 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:52]: freeing 1 rows Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x82f5960 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:60]: freeing rows at 0x82f5768 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: <core> [db_res.c:134]: freeing result set at 0x82bd628 Aug 11 20:38:39 a1 /usr/sbin/ser[390]: DEBUG: uri_db [checks.c:137]: Digest username and URI username match Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:108]: No Route headers found Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: rr [loose.c:907]: There is no Route HF Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5061 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==9 && [biloxi.test] == [127.0.0.1] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:484]: grep_sock_info - checking if host==us: 11==14 && [biloxi.test] == [195.111.158.41] Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [socket_info.c:487]: grep_sock_info - checking if port 5060 matches port 5060 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [forward.c:318]: check_self: host != me Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1383]: DEBUG: t_newtran: msg id=7 , global msg id=7 , T on entrance=(nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=47012, isACK=0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:485]: DEBUG: RFC3261 transaction matching failed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_hooks.c:335]: DBG: trans=0xb584cf18, callback type 1, id 0 entered Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_funcs.c:348]: SER: new INVITE Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [mem/shm_mem.c:117]: WARNING:vqm_resize: resize(0) called Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 572 bytes written Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=572 fd=8 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 100 trying -- your call is important to us#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:610]: DEBUG: reply sent out. buf=0x82f5c80: SIP/2.0 1..., shmem=0xb58505ec: SIP/2.0 1 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:620]: DEBUG: _reply_light: finished Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(biloxi.test(11), 35), h=599 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2674]: naptr_iterate: found a valid sip NAPTR rr _sip._tcp.biloxi.test, proto 2 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1091]: naptr_choose(o: -1 w: -1 p:0 , o: 20 w:50 p:2) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1107]: naptr_choose: changed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2674]: naptr_iterate: found a valid sip NAPTR rr _sip._udp.biloxi.test, proto 1 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1091]: naptr_choose(o: 20 w: 50 p:2 , o: 30 w:50 p:1) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1102]: naptr_choose: no change Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2674]: naptr_iterate: found a valid sip NAPTR rr _sips._tcp.biloxi.test, proto 3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1091]: naptr_choose(o: 20 w: 50 p:2 , o: 10 w:50 p:3) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [resolve.c:1107]: naptr_choose: changed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2688]: naptr_iterate: choosed NAPTR rr _sips._tcp.biloxi.test, proto 3 tried: 0x0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(_sips._tcp.biloxi.test(22), 33), h=208 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2246]: dns_srv_get_nxt_rr(0xb5864a2c, 0, 0, 559080590): selected 0/1 in grp. 0 (rand_w=0, rr=0xb5864a70 p=10 w=0 rsum=0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(b1.biloxi.test(14), 1), h=386 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b1.biloxi.test, 0) returning 0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3115]: dns_srv_resolve_ip("_sips._tcp.biloxi.test", 0, 0), ret=0, ip=195.111.158.43 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3350]: dns_naptr_sip_resolve(biloxi.test, 0, 0), srv0, ret=0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: <script>: new branch at sip:bob@biloxi.test Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:2379]: create_via_hf: id added: <;i=2>, rcv proto=3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1778]: tcp_send: no open tcp connection found, opening new one Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: 195.111.158.43 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1050]: tcpconn_new: on port 5061, type 3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, 2, fd 68 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: BUG: tls [tls_server.c:98]: Invalid connection state (bug in TCP code) Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:1349]: tcpconn_add: hashes: 1306:736:1196, 9 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tls [tls_server.c:145]: Delayed init failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8228560, 68, 2, 0xb5865b20), fd_no=55 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb5865b20 n=-1 fd=13 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012INVITE sip:bob@biloxi.test SIP/2.0#015#012Record-Route: <sip:195.111.158.41:5061;transport=tls;lr=on;ftag=5d194242>#015#012Via: SIP/2.0/TLS 195.111.158.41:5061;branch=z9hG4bK4a7b.1036c875.0;i=2#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;received=92.249.143.13;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783#015#012Max-Forwards: 69#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012P-hint: outbound#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: <core> [tcp_main.c:2166]: ERROR: tcp_send: failed to send on 0xb5865b20 (195.111.158.41:59267->195.111.158.43:5061): Operation now in progress (115) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tm [../../forward.h:163]: msg_send: ERROR: tcp_send failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, -2, fd -1 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_fwd.c:961]: t_send_branch: send to 195.111.158.43:5061 (3) failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: ERROR: <core> [tcp_main.c:3027]: handle_ser_child: ERROR: received CON_ERROR for 0xb5865b20 (id 9), refcnt 2 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b1.biloxi.test, 1) returning -1 Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8228560, 68, -1, 0x10) fd_no=56 called Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2246]: dns_srv_get_nxt_rr(0xb5864a2c, 0, 0, 559080593): selected 0/1 in grp. 1 (rand_w=0, rr=0xb5864a98 p=20 w=0 rsum=0) Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: tls [tls_server.c:689]: Closing SSL connection Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:560]: dns_hash_find(b2.biloxi.test(14), 1), h=281 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b2.biloxi.test, 0) returning 0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3115]: dns_srv_resolve_ip("_sips._tcp.biloxi.test", 1, 0), ret=0, ip=195.111.158.44 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3256]: dns_sip_resolve(biloxi.test, 1, 0), srv, ret=0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:2379]: create_via_hf: id added: <;i=2>, rcv proto=3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_fwd.c:974]: t_send_branch: send on branch 0 failed, adding another branch with another ip Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1778]: tcp_send: no open tcp connection found, opening new one Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [ip_addr.c:116]: tcpconn_new: new tcp connection: 195.111.158.44 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1050]: tcpconn_new: on port 5061, type 3 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: BUG: tls [tls_server.c:98]: Invalid connection state (bug in TCP code) Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, 2, fd 68 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tls [tls_server.c:145]: Delayed init failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:1349]: tcpconn_add: hashes: 1693:359:2697, 10 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb5865b20 n=-1 fd=13 Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:361]: DBG: io_watch_add(0x8228560, 68, 2, 0xb5865b20), fd_no=55 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012INVITE sip:bob@biloxi.test SIP/2.0#015#012Record-Route: <sip:195.111.158.41:5061;transport=tls;lr=on;ftag=5d194242>#015#012Via: SIP/2.0/TLS 195.111.158.41:5061;branch=z9hG4bK4a7b.1036c875.1;i=2#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;received=92.249.143.13;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783#015#012Max-Forwards: 69#015#012Contact: <sip:alice@92.249.143.13:2783;transport=TLS>#015#012To: <sip:bob@biloxi.test>#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO#015#012Content-Type: application/sdp#015#012Supported: replaces#015#012User-Agent: Bria Professional release 2.4 stamp 49381#015#012Content-Length: 544#015#012P-hint: outbound#015#012#015#012v=0#015#012o=- 6 2 IN IP4 92.249.143.13#015#012s=CounterPath Bria Professional#015#012c=IN IP4 92.249.143.13#015#012t=0 0#015#012m=audio 4242 RTP/AVP 107 119 100 106 0 98 8 18 101#015#012a=alt:1 2 : bg/abBbq 9nirlFAG 192.168.1.7 4242#015#012a=alt:2 1 : wCTfkbfT 6n6JS41z 92.249.143.13 4242#015#012a=fmtp:18 annexb=yes#015#012a=fmtp:101 0-15#015#012a=rtpmap:107 BV32/16000#015#012a=rtpmap:119 BV32-FEC/16000#015#012a=rtpmap:100 SPEEX/16000#015#012a=rtpmap:106 SPEEX-FEC/16000#015#012a=rtpmap:98 iLBC/8000#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=sendrecv#015#012a=x-rtp-session-id:C5EF9F1EE4C64633A625BEC3FF808160#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: <core> [tcp_main.c:2166]: ERROR: tcp_send: failed to send on 0xb5865b20 (195.111.158.41:55898->195.111.158.44:5061): Operation now in progress (115) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tm [../../forward.h:163]: msg_send: ERROR: tcp_send failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [tcp_main.c:3014]: handle_ser_child: read response= b5865b20, -2, fd -1 from 36 (390) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_fwd.c:961]: t_send_branch: send to 195.111.158.44:5061 (3) failed Aug 11 20:38:40 a1 /usr/sbin/ser[398]: ERROR: <core> [tcp_main.c:3027]: handle_ser_child: ERROR: received CON_ERROR for 0xb5865b20 (id 10), refcnt 2 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:2861]: dns_a_resovle(b2.biloxi.test, 1) returning -1 Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: <core> [io_wait.h:588]: DBG: io_watch_del (0x8228560, 68, -1, 0x10) fd_no=56 called Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3115]: dns_srv_resolve_ip("_sips._tcp.biloxi.test", 2, 1), ret=-1, ip= Aug 11 20:38:40 a1 /usr/sbin/ser[398]: DEBUG: tls [tls_server.c:689]: Closing SSL connection Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [dns_cache.c:3256]: dns_sip_resolve(biloxi.test, 2, 1), srv, ret=-1 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: ERROR: tm [t_fwd.c:980]: ERROR: t_send_branch: sending request on branch 1 failed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_funcs.c:358]: ERROR:tm:t_relay_to: t_forward_nonack returned error Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_funcs.c:366]: -477 error reply generation delayed Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: sl [sl_funcs.c:283]: error text is Unfortunately error on sending to next hop occurred (477/SL) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 636 bytes written Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=636 fd=8 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/SL)#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.573f#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_lookup.c:1536]: t_unref: delayed error reply generation(-477) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [msg_translator.c:200]: check_via_address(92.249.143.13, 192.168.1.7, 0) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:1355]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:1998]: tcp_send: send from reader (390 (36)), reusing fd Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2049]: tcp_send: sending... Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tls [tls_server.c:738]: write finished, 636 bytes written Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2085]: tcp_send: after real write: c= 0xb584b448 n=636 fd=8 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_main.c:2086]: tcp_send: buf=#012SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM)#015#012Via: SIP/2.0/TLS 192.168.1.7:4241;branch=z9hG4bK-d8754z-2a767f38e361686d-1---d8754z-;rport=2783;received=92.249.143.13#015#012To: <sip:bob@biloxi.test>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-573f#015#012From: "alice"<sip:alice@atlanta.test>;tag=5d194242#015#012Call-ID: ZTczYjE5ZThhZjhkNmYyMDY0NGYxM2ExOTdkYjg5ZDM.#015#012CSeq: 2 INVITE#015#012Server: SIP Router (2.99.0-dev01-serk (i386/linux))#015#012Content-Length: 0#015#012Warning: 392 195.111.158.41:5061 "Noisy feedback tells: pid=390 req_src_ip=92.249.143.13 req_src_port=2783 in_uri=sip:bob@biloxi.test out_uri=sip:bob@biloxi.test via_cnt==1"#015#012#015#012 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_hooks.c:252]: DBG: trans=0xb584cf18, callback type 128, id 0 entered Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:610]: DEBUG: reply sent out. buf=0x82f6088: SIP/2.0 4..., shmem=0xb58505ec: SIP/2.0 4 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: tm [t_reply.c:620]: DEBUG: _reply_light: finished Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 0 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <ACK> Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:bob@biloxi.test> Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-2a767f38e361686d-1---d8754z->; state=6 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:40 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:127]: After parse_msg... Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:166]: preparing to run routing scripts... Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=f8f2ab2c1295e90ed7dbb499b30f44b2.573f Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_to.c:753]: end of header reached, state=29 Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:173]: DEBUG: get_hdr_field: <To> [65]; uri=[sip:bob@biloxi.test] Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:175]: DEBUG: to body [<sip:bob@biloxi.test>] Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: sl [sl_funcs.c:327]: local ACK found -> dropping it! Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [usr_avp.c:629]: DEBUG:destroy_avp_list: destroying list (nil) Aug 11 20:38:41 a1 /usr/sbin/ser[390]: DEBUG: <core> [receive.c:254]: receive_msg: cleaning up Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [tcp_read.c:713]: tcp_read_req: content-length= 447 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:599]: SIP Request: Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:601]: method: <PUBLISH> Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:603]: uri: <sip:alice@atlanta.test> Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:605]: version: <SIP/2.0> Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 232, <branch> = <z9hG4bK-d8754z-0d1d4d59c544e42e-1---d8754z->; state=6 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:1283]: Found param type 235, <rport> = <n/a>; state=17 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/parse_via.c:2296]: end of header reached, state=5 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:495]: parse_headers: Via found, flags=2 Aug 11 20:38:44 a1 /usr/sbin/ser[390]: DEBUG: <core> [parser/msg_parser.c:497]: parse_headers: this is the first via # # $Id$ # # This a very basic config file w aliases and anamed route but # w/o authentication, accounting, database, multi-domain support etc. # Please refer to sip-router.cfg for a more complete example # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) memdbg=6 # memory debug message level memlog=6 # memory statistics log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 user=ser group=ser #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #TLS enable_tls=yes #TCP settings disable_tcp=no tcp_async=no tcp_send_timeout=2 #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) tcp_delayed_ack=no tcp_fd_cache=no # Alias alias=atlanta.test:5061 alias=atlanta.test:5060 #Tells how long to keep negative DNS responses in cache. If set to 0, disables caching of negative responses. #Default is 60 (seconds). dns_cache_negative_ttl=5 #Number of dns retransmissions before giving up. dns_retr_no=3 #Time in seconds before retrying a dns request. dns_retr_time=3 #dns_srv_loadbalancing Enable dns srv weight based load balancing dns_srv_lb = yes #If it is set to 'yes' and a DNS lookup fails, it will retry it for ipv6 (AAAA record) dns_try_ipv6=yes #Enable NAPTR support according to RFC 3263 dns_try_naptr = yes #Set preference for each protocol when doing naptr lookups. #By default dns_udp_pref=30, dns_tcp_pref=20, dns_tls_pref=10 and dns_sctp_pref=20. #To completely ignore NAPTR records for a specific protocol, set the corresponding protocol preference to -1 (or any other negative number). dns_udp_pref = 1 dns_tcp_pref = 1 dns_tls_pref = 1 dns_sctp_pref = 1 #If set to 'no', the search list in '/etc/resolv.conf' will be ignored. #(-> fewer lookups -> gives up faster). Default value is 'yes'. dns_use_search_list=no #default off use_dns_failover=yes # ------------------ module loading ---------------------------------- #loadpath "modules:modules_s" #loadpath "/usr/lib/ser/modules:/usr/lib/ser/modules_s" # common modules loadpath "/usr/lib/ser/modules" loadmodule "tm" loadmodule "utils" loadmodule "db_mysql" loadmodule "avpops.so" loadmodule "tls" # Kamailio modules loadpath "/usr/lib/ser/modules_k" loadmodule "sl" loadmodule "rr" loadmodule "usrloc" loadmodule "registrar" loadmodule "maxfwd" loadmodule "textops" loadmodule "uri_db.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "alias_db.so" loadmodule "pv.so" loadmodule "siputils.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so" loadmodule "presence_dialoginfo.so" loadmodule "xlog.so" # ser modules loadpath "/usr/lib/ser/modules_s" loadmodule "ctl" loadmodule "cfg_rpc" loadmodule "auth_identity" #loadmodule "sl" #loadmodule "sanity" #loadmodule "registrar" # ----------------- setting module-specific parameters --------------- # -- tls params -- modparam("tls","config","/etc/ser/tls.cfg") # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ctl params # by default ctl listens on unixs:/tmp/sip-router_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/sip-router_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/sip-router_fifo") # listen on tcp, localhost #modparam("ctl", "binrpc", "tcp:localhost:2046") # ----- registrar params ----- #modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- auth params ----- modparam("auth", "rpid_avp", "$avp(i:13)") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") modparam("auth_db", "load_credentials", "$avp(i:13)=rpid;email_address") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url", "mysql://opensips:opensipsrw@localhost/opensips") # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence", "db_url","mysql://opensips:opensipsrw@localhost/opensips") modparam("presence", "server_address", "sip:atlanta.test") modparam("presence_xml", "db_url","mysql://opensips:opensipsrw@localhost/opensips") modparam("presence_xml", "force_active", 1) # ----- auth identity params ----- modparam("auth_identity","cainfo_path","/etc/ssl/certs/ca-certificates.crt") #TODO web server publicating CERTIFICATE. modparam("auth_identity","certificate_url","https://www.atlanata.test/mycert.der") modparam("auth_identity","certificate_path","/etc/ser/tls/cert.pem") modparam("auth_identity","privatekey_path","/etc/ser/tls/privkey.pem") ####### Routing Logic ######## # main request routing logic route{ xlog("L_ERR", "Misi: $mb\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (msg:len >= 4096 ) { sl_send_reply("513", "Message too big"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } else if (is_method("INVITE")) { # even if in most of the cases is useless, do RR for # re-INVITEs alos, as some buggy clients do change route set # during the dialog. record_route(); } # route it out to whatever destination was set by loose_route() # in $du (destination URI). route(1); } else { /* uncomment the following lines if you want to enable presence */ if (is_method("SUBSCRIBE") && $rd == "atlanta.test") { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after # a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction -> # ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber (uncomment to enable auth) # authenticate all initial non-REGISTER request that pretend to be # generated by local subscriber (domain from FROM URI is local) if (!(method=="REGISTER") && from_uri==myself) /*no multidomain version*/ ##if (!(method=="REGISTER") && is_from_local()) /*multidomain version*/ { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # preloaded route checking if (loose_route()) { xlog("L_ERR", "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]"); if (!is_method("ACK")) sl_send_reply("403","Preload Route denied"); exit; } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } if (!uri==myself) ## replace with following line if multi-domain support is used ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain ## uncomment this if you want to enable presence server ## and comment the next 'if' block ## NOTE: uncomment also the definition of route[2] from below if( is_method("PUBLISH|SUBSCRIBE")) route(2); #if (is_method("PUBLISH")) #{ # sl_send_reply("503", "Service Unavailable"); # exit; #} if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU=="") { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) alias_db_lookup("dbaliases"); if(uri=~"sip:0.*@.*"){ sethost("195.111.158.3"); strip(1); # Append Remote-Party-ID header field append_rpid_hf("<", ">;party=calling;id-type=subscriber;screen=yes;privacy=off"); /* Remote-Party-ID: <sip:003646565111@195.111.104.13>;party=calling;screen=no;privacy=off */ route(1); exit; } if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } # Presence route /* uncomment the whole following route for enabling presence NOTE: do not forget to enable the call of this route from the main route */ route[2] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); # Removed command. Not Supported anymore. # At the end of script the transaction is automaticaly relased by TM. #t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); # Removed command. Not Supported anymore. # At the end of script the transaction is automaticaly relased by TM. #t_release(); } exit; } branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("L_ERR", "Reply Misi: $mb\n"); xlog("incoming reply\n"); } failure_route[1] { if (t_is_canceled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## # do not set the missed call flag again ## t_relay(); ##} } ####################################Ser########################################## # ## ------------------------- request routing logic ------------------- # ## main routing logic # #route{ # # # initial sanity checks -- messages with # # max_forwards==0, or excessively long requests # if (!mf_process_maxfwd_header("10")) { # sl_send_reply("483","Too Many Hops"); # break; # } # if (msg:len >= 4096 ) { # sl_send_reply("513", "Message too big"); # break; # } # # # we record-route all messages -- to make sure that # # subsequent messages will go through our proxy; that's # # particularly good if upstream and downstream entities # # use different transport protocol # if (!method=="REGISTER") record_route(); # # # subsequent messages withing a dialog should take the # # path determined by record-routing # if (loose_route()) { # # mark routing logic in request # append_hf("P-hint: rr-enforced\r\n"); # route(FORWARD); # break; # } # # if (!uri==myself) { # # mark routing logic in request # append_hf("P-hint: outbound\r\n"); # route(FORWARD); # break; # } # # # if the request is for other domain use UsrLoc # # (in case, it does not work, use the following command # # with proper names and addresses in it) # if (uri==myself) { # # if (method=="REGISTER") { # save_contacts("location"); ## break; # } # # # native SIP destinations are handled using our USRLOC DB # if (!lookup_contacts("location")) { # sl_send_reply("404", "Not Found"); # break; # } ## append_hf("P-hint: usrloc applied\r\n"); # } # route(FORWARD); #} # #route[FORWARD] #{ # # send it out now; use stateful forwarding as it works reliably # # even for UDP2TCP # if (!t_relay()) { # sl_reply_error(); ## } #} # # $Id$ # # Example SER TLS Configuration File # # This is the default server domain, settings # in this domain will be used for all incoming # connections that do not match any other server # domain in this configuration file. # # We do not enable anything else than TLSv1 # over the public internet. Clients do not have # to present client certificates by default. # [server:default] method = TLSv1 verify_certificate = yes require_certificate = no private_key = /etc/ser/tls/privkey.pem certificate = /etc/ser/tls/cert.pem ca_list = /etc/ser/tls/calist.pem # This is the default client domain, settings # in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] verify_certificate = yes require_certificate = yes # This is example server domain for TLS connections # received from the loopback interface. We allow # the use of SSLv2 and SSLv3 protocols here, we do # not require that clients present client certificates # but if they present it it must be valid. We also use # a special certificate and CA list for loopback # interface. # #[server:127.0.0.1:5061] #method = SSLv23 #verify_certificate = yes #require_certificate = no #private_key = local_key.pem #certificate = local_cert.pem #verify_depth = 3 #ca_list = local_ca.pem # Special settings for the iptel.org public SIP # server. We do not verify the certificate of the # server because it can be expired. The server # implements authentication using SSL client # certificates so configure the client certificate # that was given to use by iptel.org staff here. # #[client:195.37.77.101:5061] #verify_certificate = no #certificate = iptel_client.pem #private_key = iptel_key.pem #ca_list = iptel_ca.pem

2 3

sip-router select
by MÉSZÁROS Mihály 12 Aug '09

12 Aug '09

Hi! Can any of you explain how select framework is working? Where can i find any documentation about it. I am experiencing problem when i am try using it. Any help highly appreciated! Regards, Misi

2 4

[Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 12 Aug '09

12 Aug '09

here is correct reqwest and reqwest via kamailio is there any way to rewrite To: field in kamailio because now it nto work correct reqwest(asterisk) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >;tag=as366a0ba0 To: <sip:34249@193.110.78.12:8484> Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as6075df60 Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 -- Merkulov Alexander

6 21

[Kamailio-Users] kamailio auth with service provider
by Lance Evans 12 Aug '09

12 Aug '09

Hey Guys, Apparently Kamailio cannot authenticate with my service provider. Can this not be done with a sip trunk or peering!? if so how do i setup a trunk with kamailio!? and how do i setup peering?! Thanks

2 1

[Kamailio-Users] Fwd: Getting Error 477 while registering Softphones
by Daniel-Constantin Mierla 12 Aug '09

12 Aug '09

... fwd to list ... reply all button mismatch ... ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed, Aug 12, 2009 at 8:27 AM Subject: Re: [Kamailio-Users] Getting Error 477 while registering Softphones To: Meftah Tayeb <tayeb.meftah(a)gmail.com> Hello, add: alias=tayeb.blogdns.com in your config file or use the domain module: http://kamailio.org/docs/modules/1.5.x/domain.html http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#alias Cheers, Daniel On Wed, Aug 12, 2009 at 1:15 AM, Meftah Tayeb <tayeb.meftah(a)gmail.com>wrote: > hello, > i'm getting Error 477 while registering a Softphone from outside my domain > i added my domain, tayeb.blogdns.com to the domain name variable, > sip_domain in kamctlrc.conf > created the DB, started, and added user 1000 and 1001 > in registration: > for internal use, i use 192.168.1.3 (Kamailio IP Address) > and for external use i use my domain > for internal, is working corectly but while i register a external softphone > is getting Error 477 > what is this error message? > how i can fix it? > please try to help me, any help is welcome > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4327 (20090811) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Getting Error 477 while registering Softphones
by Meftah Tayeb 11 Aug '09

11 Aug '09

hello, i'm getting Error 477 while registering a Softphone from outside my domain i added my domain, tayeb.blogdns.com to the domain name variable, sip_domain in kamctlrc.conf created the DB, started, and added user 1000 and 1001 in registration: for internal use, i use 192.168.1.3 (Kamailio IP Address) and for external use i use my domain for internal, is working corectly but while i register a external softphone is getting Error 477 what is this error message? how i can fix it? please try to help me, any help is welcome __________ Information from ESET NOD32 Antivirus, version of virus signature database 4327 (20090811) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

1 0

[Kamailio-Users] dialog info + Grandstreams = freezing
by David 11 Aug '09

11 Aug '09

Hello, I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 I am trying to setup presence_dialoginfo with my Grandstreams, Snom and Linksys. I have a 4 phones on the server. 101 - Linksys SPA962 ( 6.1.5a ) 102 - Grandstream GXP2020 ( 1.2.1.4 ) 103 - Grandstream GXP2000 ( 1.2.1.4 ) 104 - Grandstream GXP2000 ( 1.1.6.46 ) 105 - Snom 360 ( 7.3.23 ) My Kamailio deals with registrations, NAT and BLF everything else is sent to one of two asterisk boxes. I use the dispatcher module for this. This means that when I call one extension to the other, both call legs from asterisk are going through Kamailio as separate calls. But to divide my customers, the usernames are different from the URL that the user types. For example the customer dials '101' but it is changed to testspace.101 when it comes back from asterisk. So Kamailio would have two calls in the event that 101 dials 102. sip:testspace.101@myserver to 102 ( this is sent to asterisk ) sip:testspace.101@myserver to testspace.102 ( this is coming back from asterisk ) Something is horribly wrong. I have the following problems : 1. If 102 calls 103, when 103 answers both phones hang for about 2 minutes 2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on the Linksys and dark on the Snom), but the orange light stays on on the Snom and it thinks the call is still active ( the light is on, but the call is over ) 3. If any extension calls any extension and I try a call pickup, it fails. It looks like the Linksys is sending a NOTIFY to pickup the call ( I thought it was supposed to send an invite... ? ) Looking at the logs it looks like Kamailio is sending out so many NOTIFYs that it is crashing the Grandstreams, and causing the Snom to act funny. Here are some experts from my config file : root@kamailio-dev:/etc/kamailio# grep dialog * kamailio.cfg:# * avp value for dialogs is still not correct kamailio.cfg:loadmodule "dialog.so" kamailio.cfg:loadmodule "presence_dialoginfo.so" kamailio.cfg:loadmodule "pua_dialoginfo.so" kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0) kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0) kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0) kamailio.cfg:modparam("dialog", "dlg_flag", 4) kamailio.cfg:modparam("dialog", "db_mode", 1) kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)") # I still haven't figured out how to set $avp(i:10) kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300) kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog", 1) kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1) kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog", "db_url", kamailio.cfg:# Flag 4 = Mark the current request for a dialog kamailio.cfg: # sequential request withing a dialog should the set flag looks like this : if ( ds_is_from_list() ) { xlog("L_INFO", "Coming from asterisk"); if ( is_method("INVITE")) { setflag(4); } } So the dialog flag is only set for the leg coming back from asterisk. When a notify comes in : if(is_method("NOTIFY") ) { if (! t_newtran()) { sl_reply_error(); exit; }; t_reply("200", "OK"); t_release(); exit ; } Publish and subscribe are like this : if( is_method("PUBLISH") || is_method("SUBSCRIBE") ) { route(5); exit; } route[5] { # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; }; append_to_reply("Contact: <sip:myserver.tld:5060>\r\n"); if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } else { } exit; } I also have NAT checking for those telephones where stun isn't enough. Before I reach publish/subscribe/invite/notify, I also call setbflag() and sometimes call fix_nated_contact(). Additionnally, I have a block if code before my presence stuff if ( has_totag() && loose_route()) { t_relay(); }. If sip.conf:canreinvite=yes, the grandstreams freeze so long that the server times out, and the BLFs get really messed up. if sip:canreinvite=no the grandstreams only freeze for about 30 seconds. Obviously I am doing something wrong, but despite having searched google for endless hours, and poured over documentation, I can not seem to find what I did wrong. I would really appreciate if someone could shed light on my problem. I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 Thanks, David

2 3

[Kamailio-Users] problem regarding openser+sip+raduis project
by Sanyog Kale 10 Aug '09

10 Aug '09

sir i am working on project Build SIP-based VOIP Service With RADIUS AAA Using Kamailio (OpenSER) And FreeRadius as per tutorial given by you. but the problem is 1.i am not understanding how to intialize modules for freeradius 2.while testing radius server where to make file named digest 3.the openser configuration which is given by you should be stored in which file? 4.and after all configurations done how to test all the parameters configured hoping for reply regards sanyog

4 3

[Kamailio-Users] Traffic parsing
by michel freiha 09 Aug '09

09 Aug '09

Hi all, I would like to ask you please how to configure kamailio in order to send signaling traffic to a specific gateway and rtp traffic to another gateway? Kindly note that I'm using media proxy for rtp and carrierroute for sending traffic to different gateways Regards

4 5

[Kamailio-Users] Siremis accounting
by Lance Evans 07 Aug '09

07 Aug '09

Hey guys Nothing seems to be showing in the Accounting tabs (in any of the subtabs) Any ideas!?

3 2

[Kamailio-Users] Siremis not displaying anything in Online SIP Users
by Lance Evans 07 Aug '09

07 Aug '09

Hey Guys, Hope someone has time for a real newbie to VOIP, kamailio and siremis!? I got Kamailio and Siremis work ok. Fighting with Siremis accounting still... but for i cant seem see anything in the Online SIP Users when i'm making calls etc. ubuntu server 8.10 mysql v5.0.67 php v5.2.6-2 Kamailio 1.5.2 notls (built from source) + siremis 0.9.3 let me know if you guys need anything else!? Thanks

3 2

sip-router + tls
by MÉSZÁROS Mihály 07 Aug '09

07 Aug '09

Hi All! I have on question regarding the TLS module. loadpath "/usr/lib/ser/modules" loadmodule "avpops.so" loadmodule "tls" _but_ i can use only tlsops setting parameters. modparam("tlsops","config","/etc/ser/tls.cfg") tls module name in tlsops? modules/tls_mod.c /* * Module interface */ struct module_exports exports = { "tlsops", Is this a typo or some work in progress? Is it a bug, or a feature? Thanks, Misi

3 4

sip-router + tls
by MÉSZÁROS Mihály 07 Aug '09

07 Aug '09

Hi All! I have on question regarding the TLS module. loadpath "/usr/lib/ser/modules" loadmodule "avpops.so" loadmodule "tls" _but_ i can use only tlsops setting parameters. modparam("tlsops","config","/etc/ser/tls.cfg") tls module name in tlsops? modules/tls_mod.c /* * Module interface */ struct module_exports exports = { "tlsops", Is this a typo or some work in progress? Is it a bug, or a feature? Thanks, Misi

1 0

[Kamailio-Users] Local routing
by Martin Werne 06 Aug '09

06 Aug '09

Hi! I'm planning a new platform based on Kamailio, to replace an existing Cisco-platform. - A total of appr. 2000 ATAs/IP Phones (today appr. 300). - Routing to various providers/gateways via custom made OSP-server. - Two Kamailio-servers on different geographic locations. - Load-balancing/Failover of ATA's by use of DNS-SRV. The main "problem" I see, is how local calls between ATA's registered on each of the Kamailio-servers is routed. How can Kamailio1 route a call to an ATA registered on Kamailio2? What is best in this scenario? 1. If user not found on local Kamailio-server, forward call to next Kamailio-server. 2. Let the Kamailio-servers share the same database (same location-table)? 3. Let the Kamailio-servers copy the REGISTER to the other Kamailio-server(s) so that the other Kamailio-server(s) update their own location-table. I would prefer not to use alternative 1... If alternative 3 is the best, is 't_replicate' the way to do it? What if I add a third Kamailio-server, will that change the recommendation? Martin

4 4

[Kamailio-Users] cfgutils module
by karhu 06 Aug '09

06 Aug '09

Hello, I'm trying something but doesn't work :s Here is my idea : In my cfg file i added the following lines : if (is_gflag("10")) xlog("L_INFO", "gflag is 10"); I'm using MI Commands to set gflag to 10 so : $ kamctl fifo set_gflag 10 After that if i look on my log file there is no "gflag is 10" !! Am i using the module and MI commands in the right way ? Thanks -- View this message in context: http://www.nabble.com/cfgutils-module-tp24823254p24823254.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 11

nat_uac_test mode parameter 16
by Andrew O. Zhukov 05 Aug '09

05 Aug '09

I just spend a lot of time to fix a trouble with Cisco AS5300 (AS5350) BYE message disapearing. The same not resolved case here: http://lists.iptel.org/pipermail/serusers/2005-July/021698.html The source of trouble was in the nat_uac_test parameter 16: Here is the part of cisco log: Jul 31 12:16:00.726 UKR: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 193.28.184.13:5060, local_port 50058 --Skipped--- Jul 31 12:16:00.738 UKR: Sent: INVITE sip:3039100@193.28.184.13:5060 SIP/2.0 Via: SIP/2.0/UDP 193.28.184.21:5060 From: <sip:2308228@193.28.184.21>;tag=12E601F8-10C7 To: <sip:3039100@193.28.184.13> Via header the same in the all availible IOS-es. As the result openser considerate the Cisco is out of the nat box. The question is: Can someone show me the case when this parameter is necessary, before I'll switch it of on the production server? Thanks in advance. Andrew O. Zhukov.

3 6

[Kamailio-Users] $hdr error if used with an avp
by catalina oancea 04 Aug '09

04 Aug '09

Hello I need to get the value of a header using a dynamic variable. I have the name of the header in an avp and I get the value using $hdr($avp(s:headername))). This worked in kamailio 1.4.0, I am sure of it because I tried it now on 1.4.0. But now in kamailio 1.5.1 I get an error when trying to retrieve the value: /usr/sbin/kamailio[32761]: ERROR:core:pv_get_spec_name: name type is PV_NAME_OTHER - cannot resolve /usr/sbin/kamailio[32761]: ERROR:pv:pv_get_hdr: invalid name Does anybody have any idea what the problem might be? Thanks

2 4

[Kamailio-Users] QjSimple 0.6.3 - sip phone for linux, macosx and windows
by Daniel-Constantin Mierla 03 Aug '09

03 Aug '09

Thanks to Klaus Darilion, QjSimple 0.6.3 works on most of desktop OS: Linux(es), Windows and Mac OS X. It is a very handy application, especially for developers, but not olny, with simple UI and debugger window. A perfect replacement (for my very old good Linux testing tool) KPhone since QjSimple has SIMPLE presence support, tls ... more at: http://www.ipcom.at/index.php?id=560/ Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

3 6

[Kamailio-Users] dokuwiki, bug tracker and git
by Daniel-Constantin Mierla 03 Aug '09

03 Aug '09

Hello, I installed two plugins for dokuwiki and flyspray bug tracker to reference easier the git repository and bugs. For FlySpray, when you describe or comment a bug you can reference to a git commit by using: GIT#hash - hash must be at list the first six characters of git commit hash value Unfortunately FlySpray accepts only plain text in the "Close Task" description, so the git reference does not work there, yet. In dokuwiki same syntax can be used. In addition, you can add references to FlySpray items via: FlySpray#itemid - itemid is the integer id of FlySpray tracked item I tried to collect all these plus other options for DW syntax in a wiki page: http://sip-router.org/wiki/tutorials/contribute-to-dokuwiki Feel free to add more details there. Cheers, Daniel -- Daniel-Constantin Mierla * SIP Router Bootcamp * Kamailio (OpenSER) and Asterisk Training * Berlin, Germany, Sep 1-4, 2009 * http://www.asipto.com/index.php/sip-router-bootcamp/

1 0

[Serusers] IPTel web pages still relevant?
by SIP 03 Aug '09

03 Aug '09

A brief look at the IPTel pages show pretty much nothing but comment spam for pages and pages. Does anyone even bother to maintain the pages anymore? I'm looking for more relevant documentation to SER 2.0 (or even the SIP Router project). The docs in the doc subtree of the source for both of these projects is outdated and no longer valid with respect to any information stored in the database. It still references subscriber tables and the like. All 0.9.X format DB data. Is there an update somewhere? N.

3 3

[Kamailio-Users] Kamailio OSP-module compile problem
by Martin Werne 31 Jul '09

31 Jul '09

Hi! I have problem compiling the OSP-module in Kamailio 1.5.x OS: Ubuntu Server 8.04LTS Compiling and installing OSPToolkit is OK. The problem occures when it comes to compiling the OSP-module in Kamailio: make[1]: Entering directory `/usr/src/kamailio/modules/osp' make[1]: Leaving directory `/usr/src/kamailio/modules/osp' make[1]: Entering directory `/usr/src/kamailio/modules/osp' Compiling destination.c Compiling globals.c Compiling orig_transaction.c In file included from orig_transaction.c:40: sipheader.h:48: error: expected declaration specifiers or '...' before 'OSPTCALLID' orig_transaction.c: In function 'ospLoadRoutes': orig_transaction.c:84: error: 'OSPE_DEST_PROT' undeclared (first use in this function) orig_transaction.c:84: error: (Each undeclared identifier is reported only once orig_transaction.c:84: error: for each function it appears in.) orig_transaction.c:84: error: expected ';' before 'protocol' orig_transaction.c:85: error: 'OSPE_DEST_OSP_ENABLED' undeclared (first use in this function) orig_transaction.c:85: error: expected ';' before 'enabled' orig_transaction.c:150: error: 'protocol' undeclared (first use in this function) orig_transaction.c:154: error: 'OSPE_DEST_PROT_SIP' undeclared (first use in this function) orig_transaction.c:157: error: 'OSPE_DEST_PROT_H323_LRQ' undeclared (first use in this function) orig_transaction.c:158: error: 'OSPE_DEST_PROT_H323_SETUP' undeclared (first use in this function) orig_transaction.c:159: error: 'OSPE_DEST_PROT_IAX' undeclared (first use in this function) orig_transaction.c:163: error: 'OSPE_DEST_PROT_UNDEFINED' undeclared (first use in this function) orig_transaction.c:164: error: 'OSPE_DEST_PROT_UNKNOWN' undeclared (first use in this function) orig_transaction.c:170: error: 'enabled' undeclared (first use in this function) orig_transaction.c:174: error: 'OSPE_OSP_FALSE' undeclared (first use in this function) orig_transaction.c:189: error: 'OSPC_SOURCE' undeclared (first use in this function) orig_transaction.c: In function 'ospRequestRouting': orig_transaction.c:260: error: 'OSPTCALLID' undeclared (first use in this function) orig_transaction.c:260: error: 'callids' undeclared (first use in this function) orig_transaction.c:282: error: too many arguments to function 'ospGetCallId' orig_transaction.c:325: error: 'OSPC_E164' undeclared (first use in this function) make[1]: *** [orig_transaction.o] Error 1 make[1]: Leaving directory `/usr/src/kamailio/modules/osp' make: *** [modules] Error 2 Has tried with both OSPToolkit 3.5.2 and 3.3.6. And with both Kamailio 1.5.1 and 1.5.2 Same problem. Does anyone know what is the problem? Regards, Martin

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[Kamailio-Users] About dlg_isflagset in onreply_route
by Min Wang 31 Jul '09

31 Jul '09

Hi all In kamailio 1.5.2, I used dlg_isflagset inside a ONREPLY_ROUTE. ////////////////scripts////////////////// Route[1]{ ... Dlg_manage(); Dlg_setflag("1"); ... } onreply_route[1] { #dlg_get("$ci", "$ft", "$tt"); #xlog("L_INFO", " Dialog: $dlg(h_id) . $dlg(h_entry)\n"); if (dlg_isflagset("1")) { xlog("L_INFO", " here is it \n"); } } ///////////////////////////////////////// The dlg_isflagset seems do not match anything until there is dlg_get("$ci", "$ft", "$tt"); before it. Questions are: (1) Is it the correct usage or a pontential bug. (2) How dialog is cheched/matched in the onreply_route internally? Kind Regards Min Wang

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[Kamailio-Users] Parallel forking problem
by Dubravko Caric 30 Jul '09

30 Jul '09

Hi all, could you please give me suggestions how to simultaneously send call to PSTN destination for UAC that is logged in. So, when call comes for sip:bob@example.com it must also go to its PSTN destination 1234(a)10.0.0.2 (avp in mysql) and ring both destinations simultaneously. Example below doesn't work and I keep ending up with two INVITEs to PSTN GW and proxy doesn't fork call to UAC. in this case, if I understand correctly, this is proper behavior because append_branch() has its function only if UAC isn't logged in or on 480,486... And how to handle this when user is logged in? if (!lookup("location")) { ... } if(...) { avp_pushto("$ru", "$avp(s:PSTN_fwd)"); append_branch(); } Thanks a lot /dubravko

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