- sr-users - kamailio.org

[OpenSER-Users] Problem with loading snmpstats
by Stanisław Pitucha 25 Mar '08

25 Mar '08

Hey I can't get snmpstats module to load. Message from logs is (during config parsing): ERROR:core:sr_load_module: could not open module </usr/local/openser/lib/openser/modules/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader System: CentOS-5.1, 32b, up-to-date. Openser: svn head from 1.3 branch Tried `ldconfig`, recompiling after `make clean`, ... Has anyone got some ideas how to resolve it? Thanks, Stanisław Pitucha

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[OpenSER-Users] 483: Too Many Hops
by Marcio 25 Mar '08

25 Mar '08

Thanks Bogdan..but didn't work....I set the "alias"(alias=" vmw02-src.spo.voip.br <http://vmw02-src.spo.voip.br>" into "openser.cfg" and it`s ok...passed in "vmw02-srv~#openser -c" but with the parameter "listen" didn't accept....(listen=udp:192.168.0.10:5060 < http://192.168.0.10:5060>) and be receiving yet.....483:.... regards Marcio

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[OpenSER-Users] Presence Server
by Stefan Schmidt 25 Mar '08

25 Mar '08

hello, iam new to this list and i have a question about the presence Server in openser. We are currently using asterisk for a hosted pbx solution with the feature so show active and also ringing lines on a snom360 phone. This feature uses the Asterisk hint, subscriber and notify handling. i havent found if i could use the presence server for this solution. The problem is that if i just pass through the subscriber and notify messages from the client via openser to the asterisk server that the server doesnt know if the client which should be watched is registered or not (registrations are made with openser). So i have to find a solution where openser could handle subscriptions for clients which are registered and also send notifys on every invite, ringing and bye of a watched client to the subscriber. could i use the presence server for that or should i search for another solution? Best regards Steve Smith -- Für weitere Fragen stehen wir gerne unter voip(a)sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // sst(a)sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------

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[OpenSER-Users] "always" serial fork with LCR?
by Amit Sharma 25 Mar '08

25 Mar '08

Hi All, I am a newbie to this list so please forgive me if the question below has been discussed before. I could not find anything related so i am sending my query. I have been looking at the LCR module to do serial forking since we want to prioritize contacts based on q values. However, we do not want to fork in parallel to contacts even if they share the same q value. AFAIK,this is currently not possible with the LCR module. Would it be a good idea to have a parameter (e.g "append-branches") in the LCR module which can control the forking behavior when q value of contacts is the same? Thanks, Amit

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[OpenSER-Users] OpenSER sending 483: Too many Hops
by Marcio 25 Mar '08

25 Mar '08

Hi, If anybody can take a look openser.cfg below. I`m working vmware (enviroments virtual)...Whenever I try a "register" a client I get 483: Too many Hops....as if he had a loop (config file) but I am unable to find the problem... I use OpenSER 1.3-notls, client softphone: twincle or xlite... OpenSER - 192.168.0.10 Client - 192.168.0.25 Bind9/named.conf(SRV, A): spo.voip.br Does anyone know how to fix this ? Any hints.....?? thx... *OpenSER File:* ** ####### Global Parameters ######### alias="vmw02-srv.spo.voip.br" debug=6 log_stderror=no log_facility=LOG_LOCAL0 log_name="Openser" fork=yes children=8 tcp_children=4 group="openser" user="openser" /* 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/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 listen=udp:192.168.0.10:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like 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 mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" # ----------------- 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) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- 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", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") 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", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) # ----- 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@192.168.0.10 /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@192.168.0.10 /openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@192.168.0.10 /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@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); 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 accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { 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(); # DESCOMENTEI O BLOCO IF{} # authenticate if from local subscriber (uncomment to enable auth) if (!(method=="REGISTER") && from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } 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")) { # ACRESCENTEI O IP ADDR - MRG # authenticate the REGISTER requests (uncomment to enable auth) #if (!www_authorize("vmw02-srv", "subscriber")) #{ # www_challenge("vmw02-srv", "0"); # exit; #} # #if (!check_to()) ##{ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} 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); } 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(); ## t_release(); ## } ## else ## if( is_method("SUBSCRIBE")) ## { ## handle_subscribe(); ## t_release(); ## } ## ## exit; ##} branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { 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(); ##} } *Ngrep -d eth1 -qt -W byline port 5060* root@vmw02-srv:~# ngrep -d eth1 -qt -W byline port 5060 interface: eth1 (192.168.0.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 2008/03/20 13:57:31.763060 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:32.333159 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:33.321952 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . *U 2008/03/20 13:57:35.309584 192.168.0.25:5060 -> 192.168.0.10:5060* *REGISTER sip:spo.voip.br SIP/2.0.* Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . *U 2008/03/20 13:57:37.026703 192.168.0.10:5060 -> 192.168.0.25:5060* *SIP/2.0 483 Too Many Hops.* Via: SIP/2.0/UDP 192.168.0.25;rport=5060;branch=z9hG4bKignjupsg. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag= 1333af5435552d3677a0ca954c3a6e9f.6889. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Server: OpenSER (1.3.0-notls (i386/linux)). Content-Length: 0.

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[OpenSER-Users] OpenSER Sending 483 Too Many Hops on register
by Marcio 25 Mar '08

25 Mar '08

Hi, If anybody can take a look openser.cfg below. I`m working vmware (enviroments virtual)...Whenever I try a "register" a client I get 483: Too many Hops....as if he had a loop (config file) but I am unable to find the problem... I use OpenSER 1.3-notls, client softphone: twincle or xlite... OpenSER - 192.168.0.10 Client - 192.168.0.25 Bind9/named.conf(SRV, A): spo.voip.br Does anyone know how to fix this ? Any hints.....?? thx... OpenSER File: ####### Global Parameters ######### alias="vmw02-srv.spo.voip.br" debug=6 log_stderror=no log_facility=LOG_LOCAL0 log_name="Openser" fork=yes children=8 tcp_children=4 group="openser" user="openser" /* 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/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.0.10:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like 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 mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" # ----------------- 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) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- 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", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") 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", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) # ----- 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@192.168.0.10 /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@192.168.0.10 /openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@192.168.0.10 /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@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); 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 accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { 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) if (!(method=="REGISTER") && from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) { record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } 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")){ # ACRESCENTEI O IP ADDR - MRG # authenticate the REGISTER requests (uncomment to enable auth) #if (!www_authorize("vmw02-srv", "subscriber")){ # www_challenge("vmw02-srv", "0"); # exit; #} # #if (!check_to()){ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} 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); } 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(); ## t_release(); ## } ## else ## if( is_method("SUBSCRIBE")) ## { ## handle_subscribe(); ## t_release(); ## } ## ## exit; ##} branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { 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(); ##} } Ngrep -d eth1 -qt -W byline port 5060 root@vmw02-srv:~# ngrep -d eth1 -qt -W byline port 5060 interface: eth1 (192.168.0.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 2008/03/20 13:57:31.763060 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:32.333159 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:33.321952 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:35.309584 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:37.026703 192.168.0.10:5060 -> 192.168.0.25:5060 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP 192.168.0.25;rport=5060;branch=z9hG4bKignjupsg. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag= 1333af5435552d3677a0ca954c3a6e9f.6889. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Server: OpenSER (1.3.0-notls (i386/linux)). Content-Length: 0.

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[OpenSER-Users] Info about redirect function
by Valerio Di Marino 25 Mar '08

25 Mar '08

Good morning to all. I am Valerio, and this im my first post. I was never using OPENSER: I discovered it this morning, searcing for a solution. I have this situation: I am using Motorola Canopy wireless antennas to deploy internet connection, but it implements a symmetric NAT which make not operating STUN protocol. I think that one of solution should be to install a proxy server which redirects registering and calls to the public proxy (euteliavoip). I thought that OPENSER should be a good solution: can you confirm this? Thanks for every your suggestions. Valerio

2 1

[OpenSER-Users] Openser fails to load module snmpstats
by Stanislaw Pitucha 25 Mar '08

25 Mar '08

Reposting - for some reason original message got attached as response to another thread on list, so some people might have missed it. --- Hey I can't get snmpstats module to load. Message from logs is (during config parsing): ERROR:core:sr_load_module: could not open module </usr/local/openser/lib/openser/modules/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader System: CentOS-5.1, 32b, up-to-date. Openser: svn head from 1.3 branch Tried `ldconfig`, recompiling after `make clean`, ... Has anyone got some ideas how to resolve it? Google points to problems with linking C extensions for perl only, which I think is not the case here. Thanks, Stanisław Pitucha

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[OpenSER-Users] Openser sending 513 Message Too Big
by Aadilkhan Maniyar 25 Mar '08

25 Mar '08

Hi, Please find my openser.cfg file below. Whenever I try to call a softphone registered with Openser from another softphone (also registered with Openser) i get a 513 Message Too Big. I guess there is a loop in the configuration file but I am unable to find it out. Any help would be appreciated. Regards, Aadil ------------------------------------------------------------------------ ------------------------------------------------------------------------ ---------- ####### Global Parameters ######### debug=9 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* 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 = "/opt/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/opt/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/opt/openser/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:IP1:5060 listen=udp:IP2:5060 ####### Modules Section ######## #set module path mpath="/opt/openser/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like 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 mysql) module must be also loaded */ #loadmodule "presence.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc|auth_db|avpops|domain", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- 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) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- 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) # ----- 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", "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@server/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ 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", "db_url", # "mysql://openser:openserrw@server/openser") #modparam("presence", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") modparam("avpops", "avp_table", "usr_preferences") ####### Routing Logic ######## # main request 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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(1); }; if (!uri==myself) { if (method=="REGISTER") { # if (!www_authorize("", "subscriber")) { #www_challenge("", "0"); #exit; #} # #save("location"); #exit; #xlog("L_DEBUG", "Unknown Domain"); #sl_send_reply("403", "Forbidden"); }; if (method=="INVITE") { xlog("L_DEBUG", "INVITE for other domain"); route(4); }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # 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") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; }; # requests for Media server #if(is_method("INVITE") && !has_totag() ) { if(is_method("INVITE")) { ;route(3); exit; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { # route to Asterisk Media Server prefix("1"); rewritehostport("asterisk:6060"); route(1); } else { sl_send_reply("404", "Not Found"); exit; } }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { xlog("L_INFO","Entered route 1\n"); if(isflagset(2)) t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { xlog("L_INFO","Entered route 3\n"); t_on_reply("3"); # Authenticate calls #if(!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # exit; #} # end Authenticate calls # route to Asterisk Media Server #rewritehostport("asterisk:6060"); route(1); } #Route for calls to other domains route[4] { xlog("L_INFO","Entered route 4"); t_on_reply("4"); rewritehostport("asterisk:6060"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri","$avp(i:10)"); prefix("1"); # route to Asterisk Media Server rewritehostport("asterisk:6060"); resetflag(2); route(1); } failure_route[4] { t_relay(); } onreply_route[3] { if(status =~"180") { t_on_failure("4"); }; } ------------------------------------------------------------------------ ------------------------------------------------------------------------ ----------

2 1

[OpenSER-Users] OpenSER location table timezones
by Adrian A 25 Mar '08

25 Mar '08

I have an interesting problem with an OpenSER setup with 2 proxies in different timezones but using a shared DB with usrloc db_mode 2 (it actually is a master-master replicated MySQL but I don't think that is relevant). The issue is that when clients register to the proxy located in the PST zone, the 'expires' time is much earlier than the current time in EST. As a result, the proxy in EST deletes the entry from the location table since it thinks it is stale. Is there any way around this, other than setting both proxy servers to the same timezone?

3 4

[Serusers] book of asterisk with ser
by aka 25 Mar '08

25 Mar '08

dear sir i want your asterisk and ser ecopy.please send me details about it

1 0

Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 25 Mar '08

25 Mar '08

>Douglas Garstang wrote: >> >Douglas Garstang writes: >> >> The LCR module actually works better. It the gatways had a priority, >> we'd be happy! >> > >You have the source, go add support for priorities... > > >Should be as simple as tweaking the SQL statement ----> 'ORDER BY priority' > Actually, no. I don't think so. If OpenSER picks a gateway randomly, it probably does this in memory. Returning gateways in a specific order will have no effect if OpenSER picks one at random from an in-memory list. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

After spending all day working with carrierroute, I think I am missing something pretty fundamental about it. I see that I can use cr_tree_rewrite_uri() to select a new carrier, but WHY does the carrier have to be hard coded in the cfg file, and why does it require a STRING name of the carrier? I really don't understand that. It makes no sense whatsoever to hard code your carriers in your config file. Doug. ----- Original Message ---- From: Douglas Garstang <dougmig33(a)yahoo.com> To: Ovidiu Sas <osas(a)voipembedded.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 7:59:13 PM Subject: Re: [OpenSER-Users] Carrierroute Ovidiu, Ok... thanks... now... do you know how I can change carrier mid call? Doug. ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 7:24:31 PM Subject: Re: [OpenSER-Users] Carrierroute The database config is right in the README file. Here are the trees: +----+----------+ | id | carrier | +----+----------+ | 1 | carrier1 | | 2 | carrier2 | | 3 | default | +----+----------+ I agree with you that the names are confusing :) You may or may not need to use the subscriber tables, based on what functions are you using. In the example that I gave you, you don't need the location table. Instead, you force a routing tree (in my example is the "default" tree). Hope this helps, Ovidiu Sas On Mon, Mar 24, 2008 at 9:51 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Ovidiu, > > I need database config info.... your example uses the config file. > What is a 'tree'? The documentation keeps talking about it, but I don't see > it defined anywhere. Carrierroute SEEMS to need the subscriber table, but it > doesn't load it from the database.... why not? > > Why should I spend days experimenting to get it to work? If the docs where > adequate, that wouldn't be required. > > Doug. > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: users(a)lists.openser.org > Sent: Monday, March 24, 2008 5:43:53 PM > Subject: Re: [OpenSER-Users] Carrierroute > > Hello Douglas, > > The database sample is available in the README file here: > http://www.openser.org/docs/modules/1.3.x/carrierroute.html#AEN575 > The entire documentation is there, you just need to read it carefully > and experiment. > > You had a similar post about the documentation for the lcr module and > in the end it seems that you figure it out :) > If you think that you can improve documentation, feel free to send > patches for the README file. > > Here you can find a configuration file example for carrierroute: > http://voipembedded.com/resources/openser_cr.cfg > > > Regards, > Ovidiu Sas > > > On Mon, Mar 24, 2008 at 7:52 PM, Douglas Garstang <dougmig33(a)yahoo.com> > wrote: > > > > I have been trying to get the carrierroute module to work ALL day. > > Can someone 'supplement' the openser docs with maybe some sample database > > data? > > > > The docs really aren't making sense. The docs keep talking about a routing > > tree? What is a routing tree? > > When I start openser, I don't see it query the subscriber table. Should > it? > > > > Thanks, > > Doug. > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -----Inline Attachment Follows----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[Serusers] Problems running ser 0.9.7
by Ricardo Martinez 25 Mar '08

25 Mar '08

Hello List. I'm having a problema trying to run ser-0.9.7. The compilation was fine.. but i'm not able to start SER due to this error : Mar 25 00:42:53 edgelcr ser: loading module /usr/local/lib/ser/modules/acc.so Mar 25 00:42:53 edgelcr ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/acc.so>: /usr/local/lib/ser/modules/acc.so: undefined symbol: parse_orig_uri Mar 25 00:42:53 edgelcr ser: parse error (88,13-14): failed to load module Can someone help me here? Thanks in advance Regards, Ricardo Martinez.-

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Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

Ovidiu, Ok... thanks... now... do you know how I can change carrier mid call? Doug. ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 7:24:31 PM Subject: Re: [OpenSER-Users] Carrierroute The database config is right in the README file. Here are the trees: +----+----------+ | id | carrier | +----+----------+ | 1 | carrier1 | | 2 | carrier2 | | 3 | default | +----+----------+ I agree with you that the names are confusing :) You may or may not need to use the subscriber tables, based on what functions are you using. In the example that I gave you, you don't need the location table. Instead, you force a routing tree (in my example is the "default" tree). Hope this helps, Ovidiu Sas On Mon, Mar 24, 2008 at 9:51 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Ovidiu, > > I need database config info.... your example uses the config file. > What is a 'tree'? The documentation keeps talking about it, but I don't see > it defined anywhere. Carrierroute SEEMS to need the subscriber table, but it > doesn't load it from the database.... why not? > > Why should I spend days experimenting to get it to work? If the docs where > adequate, that wouldn't be required. > > Doug. > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: users(a)lists.openser.org > Sent: Monday, March 24, 2008 5:43:53 PM > Subject: Re: [OpenSER-Users] Carrierroute > > Hello Douglas, > > The database sample is available in the README file here: > http://www.openser.org/docs/modules/1.3.x/carrierroute.html#AEN575 > The entire documentation is there, you just need to read it carefully > and experiment. > > You had a similar post about the documentation for the lcr module and > in the end it seems that you figure it out :) > If you think that you can improve documentation, feel free to send > patches for the README file. > > Here you can find a configuration file example for carrierroute: > http://voipembedded.com/resources/openser_cr.cfg > > > Regards, > Ovidiu Sas > > > On Mon, Mar 24, 2008 at 7:52 PM, Douglas Garstang <dougmig33(a)yahoo.com> > wrote: > > > > I have been trying to get the carrierroute module to work ALL day. > > Can someone 'supplement' the openser docs with maybe some sample database > > data? > > > > The docs really aren't making sense. The docs keep talking about a routing > > tree? What is a routing tree? > > When I start openser, I don't see it query the subscriber table. Should > it? > > > > Thanks, > > Doug. > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 25 Mar '08

25 Mar '08

1.3.1... ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Jeremy McNamara <jj(a)nufone.net> Cc: Douglas Garstang <dougmig33(a)yahoo.com>; users(a)lists.openser.org Sent: Monday, March 24, 2008 5:34:30 PM Subject: Re: [OpenSER-Users] LCR Gateway Priority.... On Mon, Mar 24, 2008 at 7:27 PM, Jeremy McNamara <jj(a)nufone.net> wrote: > Douglas Garstang wrote: > > > >Douglas Garstang writes: > > > > The LCR module actually works better. It the gatways had a priority, > > we'd be happy! > > You have the source, go add support for priorities... > > Should be as simple as tweaking the SQL statement ----> 'ORDER BY priority' Which version of openser are you using? The SQL statements are gone since 1.2 afair. Regards, Ovidiu Sas __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

I have been trying to get the carrierroute module to work ALL day. Can someone 'supplement' the openser docs with maybe some sample database data? The docs really aren't making sense. The docs keep talking about a routing tree? What is a routing tree? When I start openser, I don't see it query the subscriber table. Should it? Thanks, Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

Actually, further to that, how can I switch to another carrier in route() ? I can't see any way in the docs to do this. ----- Original Message ---- From: Douglas Garstang <dougmig33(a)yahoo.com> To: users(a)lists.openser.org Sent: Monday, March 24, 2008 4:52:55 PM Subject: [OpenSER-Users] Carrierroute I have been trying to get the carrierroute module to work ALL day. Can someone 'supplement' the openser docs with maybe some sample database data? The docs really aren't making sense. The docs keep talking about a routing tree? What is a routing tree? When I start openser, I don't see it query the subscriber table. Should it? Thanks, Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -----Inline Attachment Follows----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Help with checking header fields
by Steven C. Blair 24 Mar '08

24 Mar '08

I need help constructing an if statement to check for a specific header field. If my proxy receives an INVITE I would like to check further into the message body for a connection address with a value of all zeros (0.0.0.0). Does anyone on this list know the syntax for performing this check? Thanks,Steve

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[Serusers] pls. help me on my installation
by Allan Jardinel 24 Mar '08

24 Mar '08

I'm using ubuntu 7.10 and install ser 0.9.7 and rtp, after the installation i test and it can call both users, when i try to restart the computer xten can not register anymore so i check the settings and its ok. but when i check the /var/run/ser.pid there is no file, also the /tmp/ser_fifo has no file so restart the ser using the /etc/ser restart or force-reload but still pid and fifo has no file. pls. help me to solve this problem.... Thank you.. Allan Send instant messages to your online friends http://uk.messenger.yahoo.com

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[OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 24 Mar '08

24 Mar '08

Can anyone tell me why gateways in LCR don't have a priority? We have multiple POP's where we originate calls from. The ITSP's that we send these calls to ALSO have multiple POP's. Choosing the best gateway isn't just a matter of choosing the least cost... It's also a matter of choosing the closest POP for that ITSP. Being able to set a priority for each gateway means I could do this... mysql> select * from lcr; +----+--------+---------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+---------------+--------+----------+ | 8 | 1303 | @203.84.213.8 | 1 | 1 | <-- (Hong Kong) | 10 | 1303 | @67.151.19.114| 2 | 1 | <-- (USA) +----+--------+---------------+--------+----------+ mysql> select * from gw; +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | id | gw_name | grp_id | ip_addr | port | uri_scheme | transport | strip | prefix | dm | Priority +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | 12 | TELEGLOBE_CH_HKG_01 | 1 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | 1 | 13 | VERIZON_CA_YTO_01 | 1 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | 2 | 14 | ARBINET_CH_HKG_01 | 1 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | 3 | 15 | TELEGLOBE_US_LAX_01 | 2 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | 1 | 16 | VERIZON_CA_YTO_02 | 2 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | 2 | 17 | ARBINET_US_LGA_01 | 2 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | 3 +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ As you can see here, for calls to to prefix 1303 from 203.84.213.8(Hong Kong), we'd send calls to gateways TELEGLOBE_CH_HKG_01, VERIZON_CA_YTO_01 and ARBINET_CH_HKG_01 in that order. For calls to 1303 from 67.151.19.114(the US), we'd send calls to the same our ITSP's in the same order, but use gateways in different POPS, TELEGLOBE_US_LAX_01, VERIZON_CA_YTO_02, ARBINET_US_LGA_01. What's really important here is that the choice of gateway for the cheapest ITSP can be modified to suit the location. Now, WITHOUT the priority, I have to do THIS instead... mysql> select * from lcr; +----+--------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------------+--------+----------+ | 1 | 1303 | @203.84.213.8 | 1 | 1 | | 2 | 1303 | @203.84.213.8 | 2 | 2 | | 3 | 1303 | @203.84.213.8 | 3 | 3 | | 5 | 1303 | @67.151.19.114 | 4 | 1 | | 6 | 1303 | @67.151.19.114 | 5 | 2 | | 7 | 1303 | @67.151.19.114 | 6 | 3 | +----+--------+----------------+--------+----------+ mysql> select * from gw; +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | id | gw_name | grp_id | ip_addr | port | uri_scheme | transport | strip | prefix | dm | +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | 12 | TELEGLOBE_CH_HKG_01 | 1 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | | 13 | VERIZON_CA_YTO_01 | 2 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | | 14 | ARBINET_CH_HKG_01 | 3 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | | 15 | TELEGLOBE_US_LAX_01 | 4 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | | 16 | VERIZON_CA_YTO_02 | 5 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | | 17 | ARBINET_US_LGA_01 | 6 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ As you can see, solution 1 requires LESS columns in the database. What other options do I have? I guess I could create separate lcr and gw tables for every POP we have... yuck. Unfortunately, there is no priority column in the gateway table. :( I see someone else asked about it a while ago... http://www.mail-archive.com/devel@openser.org/msg06757.html Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[OpenSER-Users] LCR Failover
by Douglas Garstang 23 Mar '08

23 Mar '08

I'd like to think I'm not stupid, but I've been dealing with OpenSER for 2-3 years now and it still frustrates me just as much as it did the very first time I looked at it. Can LCR do failover? I have this below. I can see that one of my gateways is selected, seemingly at random, and then when it doesn't reply (because I used a bogus IP), it doesn't time-out after 3s and go to the next gw. Is this possible? How do I do it? Btw... can anyone recommend a book on OpenSER, since the documentation is so lacking? modparam("tm", "fr_inv_timer_avp", "$avp(i:705)") route{ $avp(i:705) = "3"; ... if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); sl_send_reply("100","Trying"); if (!load_gws()) { sl_send_reply("500", "Internal server error, unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); exit; }; route(1); } } route[1] { if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[Serusers] How do you force SDP c= ipaddr to be a specific value that isn't either rtpproxy interface IP?
by Frank Durda IV 23 Mar '08

23 Mar '08

I have a problem with SER 2.0.0-RC1 and rtpproxy that needs some help. I have a PSTN gateway config and a SER box all in a 10.x.x.x address space. Directly connected to two interfaces on the connect to a Cisco router, which performs 1:1 NAT conversion to provide public Internet IPs to those interfaces, one for SIP and one for RTP. (They are separate for bandwidth, debugging and access-list reasons.) Another two interfaces on the SER connect to the PSTN switch, which has separate ports for SIP and RTP traffic. Across the Internet or direct connections to the Cisco router are from Asterisk boxes or similar devices that proxy all SIP/RTP data from their SIP phones. It looks like this: Asterisk SIP SIP SIP Server +-----+ +-+ +------+ 66.1.2.3--internet--208.4.5.6|Cisco|----10.1.1.2|S|10.2.1.1--.2|PSTN | | 1:1 | |E| |Switch| 208.4.5.7| NAT |----10.5.1.2|R|10.6.1.1--.2|TDM | +-----+ +-+ +------+ RTP RTP RTP 208.4.5.6 is converted to 10.1.1.2 and vice versa 208.4.5.7 is converted to 10.5.1.2 and vice versa (Cisco command) ip nat inside source static 10.1.1.2 208.4.5.6 ip nat inside source static 10.5.1.2 208.4.5.7 rtpproxy is in use, and has 10.5.1.2(em5) (toward Cisco) and 10.6.1.1(em4) (toward PSTN switch). SER listens on 10.1.1.2(em1) and 10.2.1.1(em0). Ser.cfg takes care of the small amount of SIP-related tweaking that must be done to make SIP work: if (dst_ip==10.1.1.2) { #Received on em1 force_send_socket("10.2.1.1"); #emit on em0 rewritehost("10.2.1.2"); #Send to PSTN Switch $gw_ip = "10.2.1.2"; #Send to PSTN Switch When the INVITE message finally reaches the PSTN switch, it says: U 2008/03/23 04:05:55.780069 10.2.1.1:5060 -> 10.2.1.2:5060 INVITE sip:4097679102@10.2.1.2 SIP/2.0. Via: SIP/2.0/UDP 10.2.1.1;branch=z9hG4bK7572.3545b131.0. Via: SIP/2.0/UDP 66.1.2.3:5060;branch=z9hG4bK39eee695;rport=5060. From: "VOIP TEST 1000" <sip:4697271700@66.1.2.3>;tag=as571000e7. To: <sip:4097679102@208.4.5.6>. Contact: <sip:4697271700@66.1.2.3>. Call-ID: 2bf456fe3016fd6b62a1649c7cb6cd75(a)66.1.2.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Date: Sun, 23 Mar 2008 04:05:50 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 490. . v=0. o=root 55522 55522 IN IP4 66.1.2.3. s=session. c=IN IP4 10.6.1.1. t=0 0. m=audio 35010 RTP/AVP 0 3 8 112 5 10 7 97 111 101. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:112 AAL2-G726-32/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:10 L16/8000. a=rtpmap:7 LPC/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:111 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. So the invite sent to the PSTN looks wonderful, correctly pointing SIP and RTP back to the SER box. The problem is on the Cisco side of the SER box, and shows up when the 183/200 messages are emitted by the SER. SER/rtpproxy rewrite the c=IN IP4 line to point to 10.5.1.2, which would be fine except that I need the public IP address 208.4.5.7 to appear in this position. I have tried using force_rtp_proxy("flags","208.4.5.7") in the reply_route section, and I know it is getting executed, but nothing changes. I've tried a variety of flag combinations, but obviously haven't tried all 720 combinations (although it feels like I have). I also tried fix_nated_sdp("2") by itself, with an empty force_rtp_proxy() and a force_rtp_proxy("","208.4.5.7") and all that did was create a "none of the above value" for c= of 10.6.1.110.5.1.2, pasting the two IP addresses together from the rtpproxy command line, and ignoring the value right value I'm trying to provide. So, the question is: How do you say to SER "Here is the value that should be used for c= on this call"? How do I set this? For complex reasons eliminated here to make the example simple, extending the public IP address over to the em5 interface so it is public already is not practical. I need a fix that changes the SDP c= value to one I will know based on whom the distant asterisk server (or equivalent) device will be. There may be other fields that also need fixing for RFC compliancy, but c= being wrong is the current problem. Thanks for suggestions.

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[OpenSER-Users] unixODBC and SQLite???
by Douglas Garstang 21 Mar '08

21 Mar '08

/etc/odbc.ini: [sqlite1] Description = SQLite Driver = SQLite Database = /root/test.db Timeout = 9000 Trace = Yes TraceFile = /tmp/sqlite.log /etc/odbcinst.ini: [SQLite] Description = SQLite ODBC Driver Driver = /usr/local/lib/libsqlite3odbc.so Setup = /usr/local/lib/libsqlite3odbc.so openser.cfg: modparam("avpops", "db_url", "unixodbc://localhost/sqlite1") What's the correct format for specifying the DSN when using SQLite with ODBC? SQLite doesn't use a user/password or host. I tried the above, "unixodbc://localhost/sqlite1" and openser barfs with: ERROR:unixodbc:new_connection: failed to connect I also tried: modparam("avpops", "db_url", "unixodbc://sqlite1") and openser barfed with: new_db_id: Error while parsing database URL: unixodbc://sqlite1 What's the correct format? Is it even supported in Openser 1.2.0? Thanks, Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] AVP_TABLE.. Why is it required?
by Douglas Garstang 21 Mar '08

21 Mar '08

I put this in openser.cfg: modparam("avpops","avp_url","mysql://unixodbc:passwd@host/database") When I start openser it complains and then EXITS. ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty Why do I need to specify an avp_table? I plan to use avp_db_query() to retrieve values from another table. That's if using avp_db_query against an odbc source is even possible. People say it is, but given that it's not documented, I'm not sure. Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[OpenSER-Users] AVPops avp_url???
by Douglas Garstang 21 Mar '08

21 Mar '08

I'm trying to use avpops with unixodbc. What's the format of the avp_uri for unixodbc? The documentation has: modparam("avpops","avp_url","mysql://user:passwd@host/database") What format do I use? Is it this? Is it something else? modparam("avpops","avp_url","unixodbc://user:passwd@host/database") Why isn't this documented? Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] O'SER behind http Proxy which requires username and Password
by luzango mfupe 21 Mar '08

21 Mar '08

Hi Mates, IM having my OpenSER in the university campus where all the communication have to go through a http Proxy which always asks for user name and password. Where in the O'SER config file should i place these credentials ( http://user:pass@proxyname:port";) Do i have to do the same in the Mediaproxy config file as well? WBR, LU.

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[OpenSER-Users] UA Get's Disconnected Once a call is initiated
by Ali Jawad 21 Mar '08

21 Mar '08

Dear All I am using Openser with TLS, the client connects successfully however once I press the call button the connection to the client get's closed "Actually the client stays open however I do check with netstat and the connection is not there anymore". As for the openser server The following is noted in the logs Mar 18 15:13:56 [1396] DBG:core:handle_ser_child: read response= b613c8d8, 1, fd -1 from 5 (1378) Mar 18 15:13:56 [1378] DBG:core:tcp_send: after receive_fd: c= 0xb613c8d8 n=4 fd=14 Mar 18 15:13:56 [1378] DBG:core:tcp_send: sending... Mar 18 15:13:56 [1378] DBG:core:tls_update_fd: New fd is 14 Mar 18 15:13:56 [1378] DBG:core:tls_write: write was successful (580 bytes) Mar 18 15:13:56 [1378] DBG:core:tcp_send: after write: c= 0xb613c8d8 n=580 fd=14 Mar 18 15:13:56 [1378] DBG:core:tcp_send: buf= SIP/2.0 408 Request Timeout Via: SIP/2.0/TLS 192.168.0.149:9071;rport=9071;branch=z9hG4bK30231106nh26;received=my.own .public.ip From: 9613041705 <sip:9613041705@server.ip>;tag=30231106 To: 9613041705 <sip:9613041705@server.ip>;tag=22c184cf1a1beaaaab06669b2c48f35d-e4d8 Call-ID: F96E5BE097AAtYP4(a)server.ip CSeq: 2 REGISTER Server: OpenSER (1.3.0-tls (i386/linux)) Content-Length: 0 Warning: 392 xx.yy.yy.yy:443 "Noisy feedback tells: pid=1378 req_src_ip=xx.xx.xx.xx req_src_port=9071 in_uri=sip:87.236.144.13 out_uri=sip:pstn.gw.ip.here:5060 via_cnt==0" Mar 18 15:13:56 [1378] DBG:tm:relay_reply: sent buf=0x816aed8: SIP/2.0 4..., shmem=0xb614fb38: SIP/2.0 4 Mar 18 15:13:56 [1378] DBG:tm:final_response_handler: done Mar 18 15:13:56 [1385] DBG:core:tls_update_fd: New fd is 20 Mar 18 15:13:56 [1385] DBG:core:_tls_read: connection closed cleanly Mar 18 15:13:56 [1385] DBG:core:tcp_read_req: EOF Mar 18 15:13:56 [1385] DBG:core:io_watch_del: io_watch_del (0x813d8c0, 20, -1, 0x10) fd_no=2 called Mar 18 15:13:56 [1385] DBG:core:release_tcpconn: releasing con 0xb613c8d8, state -1, fd=20, id=12 Mar 18 15:13:56 [1385] DBG:core:release_tcpconn: extra_data 0xb614c9f0 Mar 18 15:13:56 [1396] DBG:core:handle_tcp_child: reader response= b613c8d8, -1 from 0 Mar 18 15:13:56 [1396] DBG:core:tcpconn_destroy: destroying connection 0xb613c8d8, flags 0002 Mar 18 15:13:56 [1396] DBG:core:tls_close: closing SSL connection Mar 18 15:13:56 [1396] DBG:core:tls_update_fd: New fd is 24 Mar 18 15:13:56 [1396] DBG:core:tls_shutdown: shutdown successful Mar 18 15:13:56 [1396] DBG:core:tls_tcpconn_clean: entered Mar 18 15:13:58 [1378] DBG:tm:utimer_routine: timer routine:7,tl=0xb615f734 next=(nil), timeout=3056900000 Mar 18 15:14:01 [1378] DBG:tm:timer_routine: timer routine:2,tl=0xb615f630 next=(nil), timeout=3060 Mar 18 15:14:01 [1378] DBG:tm:wait_handler: removing 0xb615f5e8 from table Mar 18 15:14:01 [1378] DBG:tm:delete_cell: delete transaction 0xb615f5e8 Mar 18 15:14:01 [1378] DBG:tm:wait_handler: done Mar 18 15:14:41 [1378] DBG:presence:msg_watchers_clean: cleaning pending subscriptions Mar 18 15:14:41 [1378] DBG:core:db_free_rows: freeing 0 rows Mar 18 15:14:41 [1378] DBG:presence:msg_presentity_clean: cleaning expired presentity information Mar 18 15:14:41 [1378] DBG:core:db_free_rows: freeing 0 rows With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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Re: [OpenSER-Users] How configure Voice mail for multi users.
by Mark Sayer 21 Mar '08

21 Mar '08

Asif - Despite what the documentation says, I'd drop the context portion of your command. i.e. exten => s-NOANSWER,n,VoiceMail(${EXTEN}) Since you are clearly going to VM for different reasons you can also add an optional switch that will play a different vm-message. Check the docs for more info. Mark At 03:04 a.m. 20/03/2008, you wrote: >Hi All, >i want to configure voice mail on Asterisk 1.4 for multiple users. let >me explain you the scenario. > >i have 10 users with the name of >1000,2000,3000,4000,5000,6000,.......and these user can call to each >other. Now i want to configure separate voice mail box for separate >user. > >my extensions.conf ..... settings below.. >[voicemail] >exten => _X.,1,Dial(SIP/${EXTEN}) >exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) >exten => _X.,n,Goto(s-${DIALSTATUS},1) ...........

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[OpenSER-Users] new CDRTool release 6.3.0
by Adrian Georgescu 21 Mar '08

21 Mar '08

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog since 6.1.0: cdrtool (6.3.0) unstable; urgency=low * Boost the speed of the rating engine from 100% to 1000% depending on the complexity of the rating tables * Fixed set of delimiter from global.inc setting when exporting csv files * Added GetEntityProfiles command to rating engine * Return json encoded for GetPrepaidHistory * Added script to create partial billing_rates tables * Moved $CDRTool['rating'] settings to $RatingEngine in global.inc * Added $RatingEngine['split_rating_table'] option to query rating data in multiple tables, each table name is autogenerated from the billing_rates.name field * Fixed set of timeout for each SOAP connection * Adjust some column lengths Must apply setup/mysql/alter_tables.mysql cdrtool (6.2.3) unstable; urgency=low * Print information about the normalize lock in syslog and to stdout * Log if rating engine takes more time to complete. Must set $RatingEngine['log_delay'] to a value in micro seconds in global.inc and restart ratingEngine * Small html changes in sip settings page * Renamed RatingEngine object to differ from the same setting name from global.inc cdrtool (6.2.2) unstable; urgency=low * Added setting in global.inc $RatingEngine['prepaid_lock'] to enable multiple parallel calls using the same prepaid account, this is done by disabling the lock (setting it to 0 or false). Disable the lock only if you accept that the balance might become negative * Added email filter to Sip->getAccounts() * Log the call start time in the head of rating info to eliminate the confusion when rating calls that span multiple profiles * Show total call duration in rate info head * Use <voice-mailbox> instead of absolute URIs for call diversions * Removed unnecessary code used in the past for replication of changes to remote soap engine * Fixed replication of customer account to remote platform * Add active_master to replicatedDatabases in global.inc * Added web link between sip target and sip_accounts * Rewrite of mysql monitoring functions * Added DeleteBalance and DeleteBalance History commands to rating engine * Use colors in mysql replication status * Added posibility to replicate customer entries in Customer Port * Added getPrepaidHistory function to the rating engine * Replaced false with 0 and true with 1 for the return values from rating engine * Show database ip address when replication monitor returns an error * Fixed name of sip_ports settings * Added setup/crontabs directory for non debian installations cdrtool (6.2.1) unstable; urgency=low * Correct update of radius records when mediaproxy field is NULL or !timeout You must reload the sql stored procedures from setup/radius/OpenSER/radius_accounting.proc or update Freeradius with setup/radius/OpenSER/sql.conf when not using the stored procedures * Log add balance to prepaid_history table cdrtool (6.1.9) unstable; urgency=low * Fixed memory leak related to calling the same soap function several times * Added separate access numbers for FUNV, FBUS, FNOL, FNOA * Fixed load of customer properties in ngnpro client * Support multiple clusters for mysql replication monitor * Fixed vulnerability related to missing BYE and MediaProxy radius Update action (reported by Inaki Baz Castil) You must reload the sql stored procedures from setup/radius/OpenSER/radius_accounting.proc or update Freeradius with setup/radius/OpenSER/sql.conf when not using the stored procedures * Specify if output of soap functions is html formatted or not cdrtool (6.1.8) unstable; urgency=low * Fixed mrtg buildStatistics call in cron.d (env LANG=C) * Added web page to show replications status and dynamic instructions about how to fix it * Replace $CDRTool['mysql_clusters'] in global.inc with $replicated_databases (see setup/global.inc.in for syntax) * Delete DB1 and DB2 definitions from global.inc cdrtool (6.1.7) unstable; urgency=low * Set mobile_number per SIP account * Add sip_accounts_lite to soap engine settings * Update voicemail full name when sip account firstname or lastname changes * Create templates_c dir required by smarty * Added templates directory for sending welcome note cdrtool (6.1.6) unstable; urgency=low * Fixed load of customer profiles for prepaid calls when no default customer defined cdrtool (6.1.5) unstable; urgency=low * Added lock icon when SIP UA registered using TLS contact * Added more information about renormalizing prepaid calls * Show port number and transport protocol in sip trace * Group trace message per IP instead of ip:port * Added welcome message per soap engine * Show error message when ngnpro client is not configured * Fixed link from sip domains to sip aliases page * Fixed ENUM soap port authentication in sip settings page * Replaced DB_ser with DB_openser Edit global.inc and replace DB_ser with DB_openser * Increased curl timeout for bulk SOAP operations * Return price in debit balance cdrtool (6.1.4) unstable; urgency=low * Remove sample mrtg files cdrtool (6.1.3) unstable; urgency=high * Fixed calculation of rating profiles when using having specific gateways/domains/subscribers * Fixed default port alocation for NGNPro version < 3 * Allow set of reseller 0.0 (admin) in login account * Show SipThor node in Sip settings title bar * Fixed management of presence rules * Fixed link from Sip accounts to sip settings page cdrtool (6.1.2) unstable; urgency=high * Fixed detection of CDR duration when minimumDuration was set cdrtool (6.1.1) unstable; urgency=low * Corrected default login account sql insert statement Regards, Adrian

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[OpenSER-Users] How configure Voice mail for multi users.
by Mian M Asif 20 Mar '08

20 Mar '08

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) exten => _X.,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Background(vm-nobodyavail) exten => s-NOANSWER,n,VoiceMail(${EXTEN}@usersmail) exten => s-NOANSWER,n,Hangup() exten => s-CONGESTION,1,Background(vm-nobodyavail) exten => s-CONGESTION,n,VoiceMail(${EXTEN}@usersmail) exten => s-CONGESTION,n,Hangup() exten => s-CANCEL,1,Background(vm-nobodyavail) exten => s-CANCEL,n,VoiceMail(${EXTEN}@usersmail) exten => s-CANCEL,n,Hangup() exten => s-BUSY,1,Background(salesrep) exten => s-BUSY,n,VoiceMail(${EXTEN}@usersmail) exten => s-BUSY,n,Hangup() exten => s-CHANUNAVAIL,1,Background(vm-nobodyavail) exten => s-CHANUNAVAIL,n,VoiceMail(${EXTEN}@usersmail) exten => s-CHANUNAVAIL,n,Hangup() my voicemail.conf [usersmail] 1000 => 1212, userm, 1000(a)abc.com 2000 => 1212, userm, 2000(a)abc.com please help me how can i set calling number before send voice mail in users voicemail box. when i dial like VoiceMail(${EXTEN}@usersmail) voice mail not work and when i heard code user, like this VoiceMail(1000@usersmail) voicemail work fine. but i want to set if user dial 2000 or 3000 or 4000 it should be set automatically. thanks for your cooperations. regards, Asif

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[OpenSER-Users] LCR Problems (1.1 -> 1.2)
by Kristian Kielhofner 20 Mar '08

20 Mar '08

Hello everyone, In OpenSER 1.1 the following route block worked perfectly: route[5] { # LCR PSTN handler xlog("$rm $si $ru $ci - LCR - Starting\n"); if (!load_gws()) { xlog("$rm $si $ru $ci - LCR - Error in load_gw, setting engerr 21\n"); avp_printf("$avp(s:engerr)", "21"); route(15); }; if (!next_gw()) { xlog("$rm $si $ru $ci - LCR - Error in next_gw, setting engerr 22\n"); avp_printf("$avp(s:engerr)", "22"); route(15); }; t_on_failure("5"); xlog("$rm $si $ru $ci - LCR - Sending to $td\n"); route(1); } In OpenSER 1.2.3 it fails on next_gw() every time but I am unable to find out why. I used the new convert option to mysqldb.sh and I think my LCR tables are ok. I tried turning up the verbosity but I couldn't see anything relevant to LCR. What could be going on here? Thanks! -- Kristian Kielhofner

2 2

[OpenSER-Users] SQLite with OpenSER
by Douglas Garstang 20 Mar '08

20 Mar '08

How can I use avp_db_query() with SQLite in OpenSER? Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] One way audio calling a -> b, two way audio calling b -> a with two phones behind the same NAT
by Mike Fedyk 20 Mar '08

20 Mar '08

Hi, I am having some trouble with nat traversal. Depending on which phone is originating the call, it may either have one way audio, two way, or even neither. Here is the matrix I have tested so far. Hardphone to hard phone works fine between two spa942 or either of the spa942 phones and the budgetone, the problems come into the picture when soft phones are used. caller callee RTP direction budgetone linphone neither budgetone spa942 both budgetone xlite both linphone budgetone one way linphone spa942 one way linphone xlite neither spa942 budgetone both spa942 linphone neither spa942 xlite one way xlite budgetone both xlite linphone both xlite spa942 both Openser.cfg http://pastebin.ca/946561

2 2

[OpenSER-Users] Quality Metrics with MediaProxy
by Douglas Garstang 20 Mar '08

20 Mar '08

Question.... If I was to use mediaproxy with OpenSER, is there any way to capture RTP quality metrics with it? Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[Serusers] General Question on avp_subst
by Mike Trest - Personal 20 Mar '08

20 Mar '08

Hi, I am confused by documentation with foo bar examples. Can someone please give me a clear ser.cfg example of avp_subst that really works in your running ser.cfg? preferably one that works on and avp called $revision to substitute a substring of numbers (8831) with another set of numbers (5431) where the target digits are surrounded by other stuff. ..mike.. .

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Re: [OpenSER-Users] openser msilo problem
by Daniel-Constantin Mierla 20 Mar '08

20 Mar '08

Hello, On 03/18/08 13:02, Devesh wrote: > Hi Daniel, > Again thanks. > I have commented authentication part from file openser.cfg . But this is not > working for me . there might be other problems. > Where should I change in file. > that is hard to tell :-) I would recommend you to install ngrep, wireshark or other network sniffing tool and watch the sip traffic. It will help to troubleshoot better, you will see where sip messages are sent and where (if) are blocked. Cheers, Daniel PS. please cc to mailing list. > Thanks & Regards > Devesh Bissa > persistent system ltd. > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, March 18, 2008 3:27 PM > To: Devesh > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] openser msilo problem > > Hello, > > seems that you authenticate the MESSAGEs sent by msilo -- filter out > these ones, the source IP should be localhost. > > Cheers, > Daniel > > > On 03/18/08 11:49, Devesh wrote: > >> Hi Daniel, >> Thanks for your reply. >> I am not able to grep any message because this utility is not on my >> > machine. > >> I am sending you MSILO log.(attached) >> Please guide me where I am wrong. >> >> >> Thanks & Regards >> Devesh Bissa >> persistent system ltd. >> >> >> >> -----Original Message----- >> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Sent: Tuesday, March 18, 2008 2:30 PM >> To: Deveshb >> Cc: users(a)lists.openser.org >> Subject: Re: [OpenSER-Users] openser msilo problem >> >> Hello, >> >> On 03/14/08 16:17, Deveshb wrote: >> >> >>> Hi, >>> I have install openser with msilo module . Offline messages stored >>> >>> >> there >> >> >>> but it's not deliver to client( I am using X-Lite). >>> it gives me following error >>> DBG:maxfwd:is_maxfwd_present: max_forwards header not found! >>> DBG:uri:has_totag: no totag >>> DBG:tm:t_lookup_request: start searching: hash=3841, isACK=0 >>> DBG:tm:matching_3261: RFC3261 transaction matching failed >>> DBG:tm:t_lookup_request: no transaction found >>> >>> >>> >> these are not errors, are debug messages. >> >> Grab the SIP traffic for this case and and send here, that will show >> where the message is delivered. Sample command: >> >> ngrep -d any -qt -W byline port 5060 >> >> Cheers, >> Daniel >> >> >> >> >> >>> Could you please help me . >>> >>> Thank you >>> Devesh >>> >>> >>> >> DISCLAIMER >> ========== >> This e-mail may contain privileged and confidential information which is >> > the property of Persistent Systems Ltd. It is intended only for the use of > the individual or entity to which it is addressed. If you are not the > intended recipient, you are not authorized to read, retain, copy, print, > distribute or use this message. If you have received this communication in > error, please notify the sender and delete all copies of this message. > Persistent Systems Ltd. does not accept any liability for virus infected > mails. > >> >> > > > DISCLAIMER > ========== > This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails. >

2 2

[OpenSER-Users] allow_trusted evaluates to false in 1.3.1
by CSB 20 Mar '08

20 Mar '08

I have the following permissions module setup: modparam("auth_db|permissions|uri_db|usrloc|domain|lcr", "db_url", "mysql://user:password@localhost/ser") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") if (!is_uri_host_local()) { if (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; }; When an invite is received the calls is rejected with a 403 and "403 in call type processing" is written to the log file. So allow_trusted must be evaluating to false I can see from the log that the following query is executed: 13 Query select proto,from_pattern,tag from trusted where src_ip='202.180.001.001' When I execute this query directly on the database it returns: +-------+--------------+------+ | proto | from_pattern | tag | +-------+--------------+------+ | any | NULL | NULL | +-------+--------------+------+ 1 row in set (0.00 sec) Could anyone advise what I've done wrong? OpenSER1.3.1 Regards Cameron

1 0

[Serusers] Memory Leak Problems
by Ricardo Martinez 19 Mar '08

19 Mar '08

Hello I'm using SER with the LCR module and i'm having some "memory leak" problems. I have followed the "Problem Reporting" link in the iptel.org site to send you a report. My question is : In the page you mention that i need to use memlog=2 and debug=3. Is this ok? or i need to run SER with more debug to see the problem and report it? Second. How do i know if i'm using the last stable, bug free (in theory) version of SER?. SER -V : version: ser 0.9.6 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 07:54:57 Feb 29 2008 with gcc 4.1.2 i have download this version from the FTP site. I hope someone can help me here. Thanks in advance. Regards, Ricardo Martinez.-

2 4

[OpenSER-Users] openxcap with openser1.2 and postgresql
by Marcelo Rodrigues Leite 19 Mar '08

19 Mar '08

Hello all, I´ve searched around forums and openser and openxcap documentation but I haven´t found any consistent information about the use of openxcap with openser 1.2 and with postgresql. Is it possible to use the 3 of them alltogether? I can overcome any problem with openser 1.2 by installing a newer version, but any mysql restriction is going to be a real problem since I´m using Voip4all architecture, which is built over postgresql. I would really appreciate any information about this. Regards, Marcelo R. Leite

5 5

[OpenSER-Users] About TLS
by Ali Jawad 18 Mar '08

18 Mar '08

Hey All I am trying to setup TLS to use on my openser server this setup allows me to sign in /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = yes disable_tls = no listen = tls:xx.xx.xx.xx:443 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:xx.xx.xx.xx:5060 However removing listen=udp:xx.xx.xx.xx:5060 blocks me from signing in at all. I have checked with the first configuration that I sign in to port 443, however even if I am able to sign in using port 443 I can make calls I always get time out errors from my openser server. It works fine if I disable tls and only use UDP. Any hints ? Thx

2 1

[OpenSER-Users] INVITE issue - 483 Too Many Hops
by Eloi Pecharroman 18 Mar '08

18 Mar '08

Hello all, I have an issue with my OpenSER setup. I am struggling to fix it and to find further information. I am working with the debian package 1.3.0-tls Registration (with and without authentication works fine), however, when a client sends an INVITE, it seems to be looping back the request until the max-forward is reached and hence returning the error. The log throws out an insane amount of *repeated* requests. My configuration file is the standard one, I have only added a few xlogs to see if the uri matched my alias, which strangely seems to do as "URI is not mine" is never output. alias=sipserver [....] if(uri!=myself) { xlog("L_NOTICE","URI is not mine\n"); route(1); exit; } [...] route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Eloi epecharroman(a)aptx.com

2 1

Re: [OpenSER-Users] openser msilo problem
by Daniel-Constantin Mierla 18 Mar '08

18 Mar '08

Hello, seems that you authenticate the MESSAGEs sent by msilo -- filter out these ones, the source IP should be localhost. Cheers, Daniel On 03/18/08 11:49, Devesh wrote: > Hi Daniel, > Thanks for your reply. > I am not able to grep any message because this utility is not on my machine. > I am sending you MSILO log.(attached) > Please guide me where I am wrong. > > > Thanks & Regards > Devesh Bissa > persistent system ltd. > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, March 18, 2008 2:30 PM > To: Deveshb > Cc: users(a)lists.openser.org > Subject: Re: [OpenSER-Users] openser msilo problem > > Hello, > > On 03/14/08 16:17, Deveshb wrote: > >> Hi, >> I have install openser with msilo module . Offline messages stored >> > there > >> but it's not deliver to client( I am using X-Lite). >> it gives me following error >> DBG:maxfwd:is_maxfwd_present: max_forwards header not found! >> DBG:uri:has_totag: no totag >> DBG:tm:t_lookup_request: start searching: hash=3841, isACK=0 >> DBG:tm:matching_3261: RFC3261 transaction matching failed >> DBG:tm:t_lookup_request: no transaction found >> >> > these are not errors, are debug messages. > > Grab the SIP traffic for this case and and send here, that will show > where the message is delivered. Sample command: > > ngrep -d any -qt -W byline port 5060 > > Cheers, > Daniel > > > > >> Could you please help me . >> >> Thank you >> Devesh >> >> > > DISCLAIMER > ========== > This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails. >

1 0

[OpenSER-Users] interconnect with another operator trough sip
by Ion Minzu 18 Mar '08

18 Mar '08

Hi, The architecture looks like this: PSTN <-> AS5300 <-> openser(non_tls) <-> IP-PBX's (asterisk) ^ ^ ^ billing^ (radius) i want to interconnect with another operator trough sip. i think about two methods: -using AS5300 -directly to openser. what method do you recommend for me? and how do you think it's resonable to instal another openser with tls wich will function like session border controller for interconnections with operators through sip. The goal is to not change the architecture of billing and existing SIP proxy. i've read something on the sipforum but i am interested if someone has some experience in such cases, taking into consideration that fact that operators use different equipment that supports sip. i mean that i want in result something like this: | | PSTN <-|-> AS5300 <-> | ^ openser(non_tls) <-> IP-PBX's (asterisk) SIP <-|-> openser<-> ^ | ^ ^ | ^ billing^ (radius)

2 1

[OpenSER-Users] PAI-Question
by Helmut Kuper 18 Mar '08

18 Mar '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is there a way to access two or more P-Asserted-Identity header field directly with PVs? The only way I found to do is via $(hdr(P-Asserted-Identity)[N]). I ask because of performance ... regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3rA+4tZeNddg3dwRAl6xAJ4p4Tv+ih+m46yWB6GRUZuU1wS/8wCaAlOY DtCykNG42Im8cyhVQ1jn2RE= =tKfw -----END PGP SIGNATURE-----

4 3

[OpenSER-Users] openser msilo problem
by Deveshb 18 Mar '08

18 Mar '08

Hi, I have install openser with msilo module . Offline messages stored there but it's not deliver to client( I am using X-Lite). it gives me following error DBG:maxfwd:is_maxfwd_present: max_forwards header not found! DBG:uri:has_totag: no totag DBG:tm:t_lookup_request: start searching: hash=3841, isACK=0 DBG:tm:matching_3261: RFC3261 transaction matching failed DBG:tm:t_lookup_request: no transaction found Could you please help me . Thank you Devesh -- View this message in context: http://www.nabble.com/openser-msilo-problem-tp16048145p16048145.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[OpenSER-Users] Error openser + freeradius
by MagoDaViola 18 Mar '08

18 Mar '08

Untitled DocumentHi. I´m using a server with OpenSER + MySql FreeRadius + + MediaProxy, and it´s working well, but whenever I get this error: / Usr / local / sbin / openser [2610]: ERROR: auth_radius: radius_authorize_sterman: rc_auth failed All customers are authenticated and holding calls. What can that be? Alcindo Schleder.

2 1

[Serusers] ser_ctl and PostgreSQL backend
by Marcus Better 18 Mar '08

18 Mar '08

Hi, I'm trying to use SER 2.0 (daily snapshot) with a PostgreSQL backend, and have a few problems. What is the status of PostgreSQL support in SER? I created the schema with the ser_postgres script. That didn't populate the version table, so I had to do that manually. Now I need to add a domain with ser_ctl, but get this: ERROR: argument of AND must be type boolean, not type integer STATEMENT: SELECT did FROM domain WHERE 1 AND (NOT ((flags & 128)!='0')) AND (did='example.org') LIMIT 1 Is there another way to create domain entries, or should I switch to MySQL? Regards, Marcus Better

3 3

[OpenSER-Users] Your Industry profile
by VOIP Expert 18 Mar '08

18 Mar '08

1 0

[OpenSER-Users] Your Industry profile
by VOIP Expert 18 Mar '08

18 Mar '08

1 0

[OpenSER-Users] Relaying to outbound proxy
by Adrian A 17 Mar '08

17 Mar '08

Hi, I'm setting up multiple OpenSER proxies in different locations to server a SIP domain. The proxies use a shared location database so they always know where to send requests to the client when an incoming call hits either one. The clients have the outbound proxy set to whichever one is closest to them. The problem is that because the clients are NAT they do not receive the request if the proxy IP that sends it is not the same as the one they have set as the outbound proxy. The NAT will reject the request because it does not arrive from the same IP that the connection was established to. How can I make OpenSER relay requests to the "home proxy" of the user so that this can then forward it to the client? I have added an "ob_proxy" column to the subscriber table and tried using avp_db_load to get this value for use in a t_relay() but I'm getting errors in OpenSER. With the following config: modparam("avpops","avp_url","mysql://user:pass@localhost/openser") modparam("avpops", "avp_table", "dummy") modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=ob_proxy;table=subscriber") avp_db_load("$to","$avp(s:ob_proxy)/subscriber"); I'm getting errors in Openser log: /usr/local/sbin/openser[15470]: ERROR:mysql:db_mysql_submit_query: driver error: Unknown column 'value' in 'field list' /usr/local/sbin/openser[15470]: ERROR:mysql:db_mysql_query: error while submitting query /usr/local/sbin/openser[15470]: ERROR:avpops:ops_dbload_avps: db_load failed Does anyone know how to fix this or is there a better way of doing it?

3 8

[OpenSER-Users] Cant connect to local MySQL server through socket
by Megha Saini 17 Mar '08

17 Mar '08

Hi, I have installed OpenSER, openser-1.3.1-notls from: http://openser.org/pub/openser/1.3.1/src/ on Fedora Core 8. I have followed the steps given in: http://www.openser.org/dokuwiki/doku.php/install:openser-from-svn For the make command, I used: make install include_modules="mysql" I have completed all the steps successfully. However, I am not able to create database as said in step 7. To check that MySQL is working on my system, I used the command: /etc/init.d/mysqld start It says: Starting MySQL : [OK] To check openser is running: I used the command, /etc/init.d/openser start I get on command line: Starting openser: [OK] However, when i try to create database: /usr/local/sbin/openserdbctl create I get the error as: database engine 'mysql' loaded INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.socket' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.socket' (2) Usage: egrep [OPTION]... PATTERN [FILE]... Try `egrep --help' for more information. /usr/local/lib/openser/openserctl/openserdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.socket' (2) ERROR: Creating core database and grant privileges failed! I checked the file mysql.socket in /var/lib/mysql, there is only an empty file with this name. As seen in forums, I saw that many people have this file in /tmp/ folder. However, I dont have this file there also! I am unable to create the database and therefore I cannot register the client using X-lite or SJPhone. Kindly help me out with this. I am badly stuck with this problem since 3 days and have been looking in forums a lot. I am not getting any clue of how to proceed further. Please help me out if anybody has any clue. I look forward to this mailing list as my only help. I need to proceed further as I have lot of things ot be done in this project and my deadline is very near. Looking forward to your co-operation. Thanks in advance. Megha

3 3

[OpenSER-Users] R: Error in pua_usrloc: PUBLISH with empty body (bug 1821002)
by Zappasodi Daniele 17 Mar '08

17 Mar '08

Hi, thanks for the reply. I haven't found an expires paramater of usrloc module, maybe you mean the expires of registrar module: modparam("registrar", "max_expires", 3600) If it is so, I have tried with the same value for presence and registrar modparam("presence", "max_expires", 3600) and with presence expires greater than the registrar expires modparam("presence", "max_expires", 4000) I have repeted the test and I have seen that the error messages occur when I deregister and reregister the subscribed client, but I think that the error could be related to the fact that the subscriber (Eyebeam) refuses the first NOTIFY (why?) sended by openser for the first REGISTER from the subscribed. In the attached file you can see the syslog with debug level = 9 ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

1 0

[Serusers] MAC client
by jiri＠iptel.org 16 Mar '08

16 Mar '08

any recommendation for a good Mac client? Thanks! -jiri

3 2

[OpenSER-Users] SIP/2.0 481 Call Leg/Transaction Does Not Exist
by shiebar＠post.tau.ac.il 14 Mar '08

14 Mar '08

Hi, I am new with openser. I was able to configure openser with wesip that both will run. When trying to make a call between two softphone I get "SIP/2.0 481 Call Leg/Transaction Does Not Exist". Both caller and callee are registered. In addition I see that after creating the db I don't have a user openser with read-write permmissions. > openserctl acl show openser Non-existent user 'openser'. Still proceeding? [Y|N] I have used "openserdbctl create" to create the db. And i got no error message when creating it. WeSIP get the message with the openser IP and not with the callee IP. Could it be the reason for the 481 reply? Thanks in advance, Shie

1 0

[Serusers] Moved Temporarily 302 - redirect server
by Ladislav Andel 14 Mar '08

14 Mar '08

Hi all, can I with SER add following in 302 response? 1) add a parameter in Contact header that a UA will read and copy it to the new INVITE and how would I do it in ser config. 2) or add a new header field with in 302 that a UA will read and copy it to the new INVITE The first should be possible as RFC3261 in 20.10 shows in example but I'm wondering if SER can do it too. Thank you, Lada

4 8

[OpenSER-Users] Acceptor peer failed establishing state i came to know about this list today and i thought to send problem. if anybody know the solution or have any idea please do let me know. actually i am authenticating a user using openser and opendiameters NASREQ applica
by bhargav.b＠renovau.com 14 Mar '08

14 Mar '08

hi all, i came to know about this list today and i thought to send problem. if anybody know the solution or have any idea please do let me know. actually i am authenticating a user using openser and opendiameters NASREQ application as the client and server.at present the openser is sending the all needed avp's to opendiameter client.i can see all the avp code and avp data on the client side. my problem is that the on the diameter server side the avp,s are not transferred by the client. the client is giving something like (6372|1698151312) Acceptor peer failed establishing state [3003], closing i am confused about it.before i sent the avp's from openser side there was no such problem. seeking help regards

1 0

[OpenSER-Users] SIP/2.0 481 Call Leg/Transaction Does Not Exist
by bsb093 14 Mar '08

14 Mar '08

Hi, http://www.nabble.com/file/p16047318/WeSIP-debug WeSIP-debug I an trying to make a call between 2 softphones. If I try to make the call directly there is no problem but when trying to use openser with wesip as the server I get: SIP/2.0 481 Call Leg/Transaction Does Not Exist I have tried to check it with the debug but I can not find the problem. OpenSER replace the To header uri with the correct uri: rewrite_uri: rewriting Request-URI with 'sip:shine_sip2@192.168.2.101:5070' Attached are the OpenSER and WeSIP debug output. I will just add that the call is from shine_sip(192.168.2.102) to shine_sip2(192.168.2.101) and if I check openser for online user I see that both users are online: dell110l:/home/mil# openserctl ul show Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=3 max_slot=1 AOR:: shine_sip2 Contact:: sip:shine_sip2@192.168.2.101:5070 Q=0.9 Expires:: 445 Callid:: 1205472816-140-HOME-7C77A26D29(a)192.168.2.101 Cseq:: 97 User-agent:: NCH Swift Sound Express Talk 3.06 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:192.168.2.100:5060 Methods:: 4294967295 AOR:: shine_sip Contact:: sip:shine_sip@10.108.3.84:5070 Q=0.9 Expires:: 455 Callid:: 1205473792-3532-BSB093-31(a)192.168.2.102 Cseq:: 63 User-agent:: NCH Swift Sound Express Talk 3.06 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:192.168.2.100:5060 Methods:: 4294967295 Contact:: sip:shine_sip@10.108.3.84:5061 Q=0.9 Expires:: 1204 Callid:: 1205473792-3532-BSB093-31(a)192.168.2.102 Cseq:: 64 User-agent:: NCH Swift Sound Express Talk 3.06 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:192.168.2.100:5060 Methods:: 4294967295 Please advice. Thanks, Shine http://www.nabble.com/file/p16047318/OpenSER-debug OpenSER-debug -- View this message in context: http://www.nabble.com/SIP-2.0-481-Call-Leg-Transaction-Does-Not-Exist-tp160… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

Re: [OpenSER-Users] Errors
by Bai Shi 14 Mar '08

14 Mar '08

Hi TA, Please check ur config file line 163 as mentioned in the debug log: CRITICAL:core:yyerror: parse error in config file, line 163, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] If you have any problem with it, you can try to post ur config file in the mailing list, so other may be able to help you. Rgds, BS

2 3

[OpenSER-Users] Preserve the original number with alias_db_lookup
by Morten Isaksen 13 Mar '08

13 Mar '08

Hi! I have some challenges with alias_db_lookup and hope someone here can help. I have a PBX that has registered 111@xxx with my OpenSER. I want to attach 222@xxx and 333@xxx to that registration and alias_db_lookup does that fine. If a call comes to 111 the sip header looks like this INVITE 111@xxx .... To: 111 <111@xxx> ... And everything works fine. But if a call comes to 222 the sip header looks like this: INVITE 111@xxx .... To: 222 <222@xxx> ... But now the PBX thinks that the call is for 111 and not 222. On Asterisk I can do something like this to solve the problem: exten => _X.,1,goto(from-siptrunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) [from-siptrunk] exten => 111,1,dial(SIP/1010) exten => 222,1,dial(SIP/1020) But a lot of SIP enabled PBX's cannot do that sort of thing. One solution is the force the PBX to register with OpenSER for every number, but I would prefer not to do that. Another solution is to manually populate the alias tables but this is not very elegant. -- Morten Isaksen http://www.misak.dk/blog/

2 2

Re: [OpenSER-Users] OpenSER + OpenXcap for pres-rules: no NOTIFY
by Mircea Amarascu 13 Mar '08

13 Mar '08

2 1

[OpenSER-Users] Error in pua_usrloc: PUBLISH with empty body (bug 1821002)
by Zappasodi Daniele 13 Mar '08

13 Mar '08

Hello, what about the bug 1821002 on pua_usrloc: PUBLISH with no body? I read that is closed, but in the release 1.3.1 it is still present. Mar 13 14:15:12 localhost ./openser[16477]: ERROR:pua:send_publish: New PUBLISH and no body found- invalid request Mar 13 14:15:12 localhost ./openser[16477]: ERROR:pua_usrloc:ul_publish: while sending publish ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[OpenSER-Users] Is it a bug? TM module is sending CANCEL request regardless what the script do with the From Header since 1.3.0
by Bai Shi 13 Mar '08

13 Mar '08

I'm not sure if it is a bug. Currently our set up need to do masquerading on the From header for all invite requests. It used to work okie when we process for the CANCEL request in the routing logic. But after upgrading to openser 1.3.0, the TM module is sending CANCEL out using local cached From info rather than the info processed by uac_replace_from. May I know is this by design or it is a bug? Rgds, BS

3 12

[Serusers] serweb fifo error
by Adeyeye O. Michael 13 Mar '08

13 Mar '08

How can I fix this error? WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 2(0) INFO: fifo process starting: 6710 2(6710) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1171263842): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1171263842 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1374470966): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1374470966 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1359801488): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1359801488 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_90028122): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_90028122 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1698796794): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1698796794 ------------------------------------------------------------------------------------------------ Below is an excerpt of my ser.cfg ------------------------------------------------------------------------------------------------ # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen=192.168.0.2 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode = 0666 fifo_db_url="mysql://ser:heslo@localhost/ser" --------------------------------------------------------------------------------------------- Yours, Mike.

1 0

[OpenSER-Users] UDP messages size
by Ariadne Ramos 12 Mar '08

12 Mar '08

Hello We're having an issue with the INVITE'S size. The INVITE messages are trunked to 1480 Bytes when they exceed that size. Capturing with ethereal, the package appears like "Fragmented IP protocol" (proto=UDP 0x11, off=0) because is incomplete. Exists a package maximum size?, Can I change it?... If so, Can you tell me how to do this? Could be changing max_len keyword? Ariadne L. Ramos

2 1

[OpenSER-Users] when are AVPs in radius extra evaluated?
by jh＠tutpro.com 12 Mar '08

12 Mar '08

if i put in radius_extra "Some-Known-Attribute=$avp(s:bar)" and in onreply_route if (is_method("INVITE") && t_check_status("200")) { xlog("L_INFO", "Setting <s:bar> to <test>\n"); $avp(s:bar) = "test"; }; i do not get Some-Known-Attribute = "test" to accounting Start request although i do get Mar 12 07:20:32 rautu /usr/sbin/openser[7440]: Setting <s:bar> to <test> in syslog. i didn't find from acc/README information on when the AVPs in radius_extra are evaluated. accounting Start is sent out AFTER final response (200 ok) had been received, because it contains SIP-Response-Code = 200 why doesn't setting $avp(s:bar) in onreply_route cause Some-Known-Attribute to be included in accounting start record? -- juha

2 2

[OpenSER-Users] New Soap-Simple-Proxy release 1.0.3
by Adrian Georgescu 12 Mar '08

12 Mar '08

Hello, There is a new Soap-Simple-Proxy release available. soap-simple-proxy (1.0.3) unstable; urgency=low * Allow items in SOAP policy object to be nillable * Added version replace string to WSDL file The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

1 0

[OpenSER-Users] Increase MAX_BRANCHES value?
by Marcello Lupo 12 Mar '08

12 Mar '08

Hi to all, due to a failover routing policy (serial forking) i need to increase the MAX_BRANCHES value to something like 35. Our gateways have limitation of 30 Channels ---> 1 IP . I have 32 of them available. Can this change have any heavy and dangerous effect on openser ?? Just to know if i will need to increase memory aveilable or something like this. I already use : /*used only if PKG_MALLOC is defined*/ #define PKG_MEM_POOL_SIZE 4*1024*1024 due to my eavy use of AVP. Let me know if i need to increase it more than this. Thanks to all, Bye, Marcello

2 2

[OpenSER-Users] Errors
by TA 12 Mar '08

12 Mar '08

My name is Travis and I currently trying to deploy OpenSer. Out of the box OpenSer works just fine but since I have configured it to work with mediaproxy OpenSer will not longer start. Below you will find the errors that I am recieving. I have been working on this for days trying to get it resolved but no luck. Please advise. Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/mysql.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/sl.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/tm.so Mar 11 22:36:20 [3449] DBG:core:register_module: register_pv: tm Mar 11 22:36:20 [3449] DBG:core:pv_add_extra: extra items list is not initialized Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/rr.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/maxfwd.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/usrloc.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/registrar.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/textops.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/mi_fifo.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/uri_db.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/uri.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/xlog.so Mar 11 22:36:20 [3449] DBG:core:register_module: register_pv: xlog Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/acc.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/nathelper.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/mediaproxy.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/domain.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/auth.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/auth_db.so Mar 11 22:36:20 [3449] DBG:core:yyparse: loading module /usr/lib/openser/modules/alias_db.so Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: mi_fifo matches module mi_fifo Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <fifo_name> in module mi_fifo [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: rr matches module rr Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <enable_full_lr> in module rr [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: rr matches module rr Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <append_fromtag> in module rr [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: registrar matches module registrar Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <method_filtering> in module registrar [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: registrar matches module registrar Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <append_branches> in module registrar [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: registrar matches module registrar Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <max_contacts> in module registrar [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: uri_db matches module uri_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <use_uri_table> in module uri_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: uri_db matches module uri_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <db_url> in module uri_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <early_media> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <report_ack> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <report_cancels> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <detect_direction> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <failed_transaction_flag> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <log_flag> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <log_missed_flag> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <db_flag> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: acc matches module acc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <db_missed_flag> in module acc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: usrloc matches module usrloc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <db_mode> in module usrloc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: usrloc matches module usrloc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <db_mode> in module usrloc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 159, column 1-9: syntax error Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 159, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: auth_db matches module auth_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <calculate_ha1> in module auth_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: auth_db matches module auth_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <password_column> in module auth_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 163, column 1-9: syntax error Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 163, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: auth_db matches module auth_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <load_credentials> in module auth_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 176, column 1-9: syntax error Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 176, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 178, column 1-9: syntax error Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 178, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: domain matches module domain Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <db_mode> in module domain [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: alias_db|auth_db|usrloc|uri_db matches module alias_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <use_domain> in module alias_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: alias_db|auth_db|usrloc|uri_db matches module auth_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <use_domain> in module auth_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: alias_db|auth_db|usrloc|uri_db matches module uri_db Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <use_domain> in module uri_db [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: alias_db|auth_db|usrloc|uri_db matches module usrloc Mar 11 22:36:20 [3449] DBG:core:set_mod_param_regex: found <use_domain> in module usrloc [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 191, column 1-9: syntax error Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 191, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] ERROR:core:set_mod_param_regex: no module matching presence_xml found |Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 191, column 19-20: Can't set module parameter Mar 11 22:36:20 [3449] ERROR:core:set_mod_param_regex: no module matching presence found |Mar 11 22:36:20 [3449] CRITICAL:core:yyerror: parse error in config file, line 192, column 20-21: Can't set module parameter Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <use_media_proxy>(0) in module mediaproxy [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <t_on_reply>(1) in module tm [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <end_media_session>(0) in module mediaproxy [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <t_relay>(0) in module tm [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <sl_reply_error>(0) in module sl [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <use_media_proxy>(0) in module mediaproxy [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] DBG:core:find_cmd_export_t: found <end_media_session>(0) in module mediaproxy [/usr/lib/openser/modules/] Mar 11 22:36:20 [3449] ERROR:core:main: bad config file (12 errors) Mar 11 22:36:20 [3449] DBG:xlog:destroy: destroy module... Mar 11 22:36:20 [3449] DBG:tm:tm_shutdown: tm_shutdown : start Mar 11 22:36:20 [3449] DBG:tm:tm_shutdown: emptying hash table Mar 11 22:36:20 [3449] DBG:tm:tm_shutdown: releasing timers Mar 11 22:36:20 [3449] DBG:tm:tm_shutdown: removing semaphores Mar 11 22:36:20 [3449] DBG:tm:tm_shutdown: destroying callback lists Mar 11 22:36:20 [3449] DBG:tm:tm_shutdown: tm_shutdown : done Mar 11 22:36:20 [3449] DBG:core:shm_mem_destroy: -- "You can educate them but you can't make them smart." ~Unknown~

2 3

[OpenSER-Users] Minor bug in permissions module?
by Helmut Kuper 12 Mar '08

12 Mar '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, during setup of permissions module I found that module parameters "default_allow_file" and "default_deny_file" doesn't seem to work as expected. No mater what filename I configured permissions.so always checks for "permissions.allow" resp. "permissions.deny" files in openser's main config directory. Here is openser's level 3 console output: Mar 11 16:25:49 [3235] WARNING:core:main: no fork mode Mar 11 16:25:49 [3235] INFO:core:init_tcp: using epoll_lt as the TCP io watch method (auto detected) Mar 11 16:25:49 [3235] NOTICE:core:main: version: openser 1.4.0dev0-notls (x86_64/linux) Mar 11 16:25:49 [3235] INFO:core:main: using 32 Mb shared memory Mar 11 16:25:49 [3235] INFO:core:main: using 1 Mb private memory per process Mar 11 16:25:49 [3235] INFO:xlog:mod_init: initializing... Mar 11 16:25:49 [3235] INFO:sl:mod_init: Initializing StateLess engine Mar 11 16:25:49 [3235] INFO:tm:mod_init: TM - initializing... Mar 11 16:25:49 [3235] INFO:maxfwd:mod_init: initializing... Mar 11 16:25:49 [3235] INFO:usrloc:ul_init_locks: locks array size 512 Mar 11 16:25:49 [3235] INFO:registrar:mod_init: initializing... Mar 11 16:25:49 [3235] INFO:textops:mod_init: initializing... Mar 11 16:25:49 [3235] INFO:auth:mod_init: initializing... Mar 11 16:25:49 [3235] INFO:auth_db:mod_init: initializing... Mar 11 16:25:49 [3235] INFO:avpops:avpops_init: initializing... Mar 11 16:25:49 [3235] WARNING:permissions:parse_config_file: file not found: etc/openser/permissions.allow Mar 11 16:25:49 [3235] WARNING:permissions:mod_init: default allow file (etc/openser/permissions.allow) not found => empty rule set Mar 11 16:25:49 [3235] WARNING:permissions:parse_config_file: file not found: etc/openser/permissions.deny Mar 11 16:25:49 [3235] WARNING:permissions:mod_init: default deny file (etc/openser/permissions.deny) not found => empty rule set And here is my permissions config: # -- permissions -- loadmodule "permissions.so" modparam("permissions", "default_allow_file", "permissionddds.allow") modparam("permissions", "default_deny_file", "/opt/app/voip/openser/etc/permissionddds.deny") modparam("permissions", "check_all_branches", 1) modparam("permissions", "allow_suffix", ".allow") modparam("permissions", "deny_suffix", ".deny") modparam("permissions", "address_table", "address") modparam("permissions", "grp_col", "grp") modparam("permissions", "ip_addr_col", "ip_addr") modparam("permissions", "mask_col", "mask") modparam("permissions", "port_col", "port") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "tag_col", "tag") modparam("permissions", "peer_tag_avp", "") I use openser 1.4 build 3384 regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFH1qUP4tZeNddg3dwRAmPbAJiQAKWUzSoUVJx2iPjJEzT+PpKvAJ9Cwgqh Q5z67Xb1/Oze+QK4aqHPNA== =fjgt -----END PGP SIGNATURE-----

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[OpenSER-Users] OpenSer 1.3 + SEMS 0.10: open file problem
by Erich Hohermuth 12 Mar '08

12 Mar '08

Hello I try to use OpenSer 1.3 with SEMS 0.10 for voice announcements and i get a sems error which I don't understand. (49239) ERROR: open (AmAudio.cpp:532): open returned -1 (49239) ERROR: run (AmSession.cpp:319): AnnouncementDialog::onSessionStart: Cannot open file (49239) ERROR: run (AmSession.cpp:327): 500 unexpected exception. (49239) DEBUG: destroy (AmSession.cpp:350): AmSession::destroy() SEMS is compiled with -DOpenSER and the communication between the two seems to work. SER COnfig: =========== listen=212.101.4.3 port=5060 loadmodule "mi_datagram.so" modparam("mi_datagram", "socket_name", "/tmp/openser_sock") modparam("mi_datagram", "unix_socket_mode", 0600) modparam("mi_datagram", "unix_socket_user", "root") modparam("mi_datagram", "unix_socket_group", "wheel") INVITE ROUTE: if (uri=~"^sip:666@") { if(!t_write_unix("/tmp/sems_sock","announcement")) { t_reply("500","error contacting sems"); }; }; SEMS Config: ============ plugin_config_path=/usr/local/etc/sems//etc/ #fork=yes #stderr=no fork=no stderr=yes # - sets log level (error=0, warning=1, info=2, debug=3) loglevel=3 default_application = announcement socket_name=/tmp/sems_sock reply_socket_name=/tmp/sems_resp_sock ser_socket_name=/tmp/openser_sock plugin_path=/usr/local/lib/sems/plug-in/ # - sets lowest for RTP used port rtp_low_port=50000 rtp_high_port=60000 media_processor_threads=1 listen=212.101.4.3 sip_port=5060 SEMS-DEBUG: =========== (49649) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: dispatch it! (49649) DEBUG: getAnnounceFile (Announcement.cpp:79): trying '/usr/local/lib/sems/audio/666.wav' (49649) DEBUG: setLocalTag (AmSession.cpp:205): AmSession::setLocalTag(240BFB44-47D000440002686F-28401100) (49649) DEBUG: start (AmThread.cpp:135): Thread 0 is just created. ... (49649) DEBUG: onSessionStart (Announcement.cpp:142): AnnouncementDialog::onSessionStart (49649) DEBUG: wav_read_header (wav_hdr.c:100): trying to read WAV file (49649) DEBUG: wav_read_header (wav_hdr.c:103): tag = <RIFF> (49649) DEBUG: wav_read_header (wav_hdr.c:111): file size = <2935619840> (49649) DEBUG: wav_read_header (wav_hdr.c:114): tag = <WAVE> (49649) DEBUG: wav_read_header (wav_hdr.c:121): tag = <fmt > (49649) DEBUG: wav_read_header (wav_hdr.c:129): chunk_size = <301989888> (49649) DEBUG: wav_read_header (wav_hdr.c:133): fmt = <100> (49649) DEBUG: wav_read_header (wav_hdr.c:137): channels = <256> (49649) DEBUG: wav_read_header (wav_hdr.c:141): rate = <1075773440> (49649) DEBUG: wav_read_header (wav_hdr.c:148): bits/sample = <4096> (49649) ERROR: open (AmAudio.cpp:532): open returned -1 (49649) ERROR: run (AmSession.cpp:319): AnnouncementDialog::onSessionStart: Cannot open file (49649) ERROR: run (AmSession.cpp:327): 500 unexpected exception. (49649) DEBUG: destroy (AmSession.cpp:350): AmSession::destroy() As you can see the file exists, but nothing happens. :( Full debug on request. Thank you very much for your help Regards Erich -- * Erich Hohermuth IP Engineer - SolNet (AS 9044) PGPKEY-46A08FCB * * phone: +41 32 517 6220 / sip:9044*463@inoc-dba.pch.net *

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[Serusers] sorry -- reply fifo opening error
by Adeyeye O. Michael 12 Mar '08

12 Mar '08

Hi, I get the error below each time I log in to serweb: sorry -- reply fifo opening error Below is my directory permissions for the serweb folder and ser_fifo: prw-rw-rw- 1 root root 0 2008-03-11 19:43 ser_fifo drwxr-xr-x 2 www-data www-data 4096 2008-03-11 19:43 serweb Here is my fifo mode in ser.cfg fifo="/tmp/ser_fifo" fifo_mode = 0666 How can I fix this problem, should I change owner of ser_fifo to www_data also? I have gone through the archive but couldn't find a definite solution. Yours, Michael

1 0

[Serusers] sorry -- reply fifo opening error
by Adeyeye O. Michael 11 Mar '08

11 Mar '08

Hi, I get the error below each time I log in to serweb: sorry -- reply fifo opening error Below is my directory permissions for the serweb folder and ser_fifo: prw-rw-rw- 1 root root 0 2008-03-11 19:43 ser_fifo drwxr-xr-x 2 www-data www-data 4096 2008-03-11 19:43 serweb How can I fix this problem, should I change owner of ser_fifo to www_data also? I have gone through the archive but couldn't find a definite solution. Yours, MIchael

1 0

[OpenSER-Users] Not adding record-route for ack
by Vikas CG. 11 Mar '08

11 Mar '08

Hi all, We have built an SIPALG and using openser server as outbound proxy for testing, When call is made, Ack message contains the route for openser, After receiving the message he is forwarding to next route, but not making it has record-route. Is this behavior is right? Other servers will add record-route after forwarding to next route. So please suggest for this? Thanks, Vikas C.G.

2 3

[OpenSER-Users] OpenSer 1.1 and GW PSTN
by Matteo Abrile 11 Mar '08

11 Mar '08

Hello, I have this problem, I have a pstn gw that accept only 1 ip address, this to security, I setup that my gw accept only ip of openser. So when I try make a call from example my notebook with x-lite, openser redirect my call to pstn gateway but with my ip... example sip:xxxxxxx@notebook-ip.... so my pstn gw don't accept call. How I can doing to send call to pstn gw with ip of openser ??? Thank you Matteo

2 1

[OpenSER-Users] user_agent_header - uac_auth()
by Thomas Gelf 11 Mar '08

11 Mar '08

It seems that uac_auth() doesn't make use of user_agent_header. IMO this would make sense, doesn't it? Regards, Thomas Gelf

2 1

[OpenSER-Users] Dialog module - how to cleanly stop an active dialog
by Thomas Gelf 11 Mar '08

11 Mar '08

Hi list, I'm currently playing around with the dialog module. Current config: > loadmodule "dialog.so" > modparam("dialog", "enable_stats", 1) > modparam("dialog", "hash_size", 4096) > modparam("dialog", "rr_param", "rdlg") > modparam("dialog", "dlg_flag", FLAG_DIALOG) > modparam("dialog", "dlg_match_mode", 1) > modparam("dialog", "db_url", "mysql://DB_USER:DB_PASS@DB_HOST/DB_NAME") > modparam("dialog", "db_mode", 1) I had to comment out the following ones, as 1.3 (SVN rev 3864) doesn't like them (as af http://www.openser.org/docs/modules/1.3.x/dialog.html thy should work): > #modparam("dialog", "default_timeout", "7200") > #modparam("dialog", "db_update_period", "60") However: if I end a dialog with (example): > openserctl fifo dlg_end_dlg 162 1311477520 syslog is telling me: > ... CRITICAL:dialog:next_state_dlg: bogus event 7 in state 5 Both BYE's are sent out correctly, and both clients answer them (OK). Is this "CRITICAL" thingy the normal behaviour? Or may there something be wrong with my config? I'm currently running > setflag(FLAG_DIALOG); for each INVITE request - DB entries are created / modified correctly. Mediaproxy sessions are not terminated, they time out after idling around for a while: > ... session 7bf33c00a5793790@<ip.address>: ended (did timeout). Is there a way to cleanly stop them too? Kind regards, Thomas Gelf

2 4

[OpenSER-Users] OpenSER 1.3.1 is released
by Bogdan-Andrei Iancu 11 Mar '08

11 Mar '08

Hi everybody, OpenSER 1.3.1, the first patch release from the 1.3 seria, is out - it is a minor release, including only bug fixes - no new features or functionalities. V1.3.1 brings some important fixes in critical modules (like TM and USRLOC), so update is recommended. To read more about this release, see http://www.openser.org/mos/view/News/NewsItem/OpenSER-1.3.1-is-released/ Best regards, Bogdan

1 0

[Serusers] syntax explanation
by Valentin Rosic 11 Mar '08

11 Mar '08

hello, I've noticed that syntax is a bit changed in SER 2.0. could someone, please, explain me or give me URL with explanations for parameters with !$ symbols (for example !$t.did and !$f.did) Also... Could someone expain me the difference between did and realm coloumn in credentials? I'm still a beginner and I would really like to know SER to details 0:) thank You in advance for Your responses -- srdacan pozdrav/regards, Valentin ------------------------------------------------ Valentin Rosic SRCE - Sveucilisni racunski centar J. Marohnica 5, 10000 Zagreb, Croatia email: valentin.rosic @ srce.hr web: http://www.srce.hr tel: +385/1/6165 303 fax: +385/1/6165 559 ------------------------------------------------

1 0

[OpenSER-Users] Documentation: user_agent_header
by Thomas Gelf 11 Mar '08

11 Mar '08

user_agent_header behaves likes server_header - I suggest adding a note to http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#user_agent_hea… as in http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#server_header Kind regards, Thomas Gelf

2 2

[OpenSER-Users] Dispatcher module
by Mouncif Benniane 11 Mar '08

11 Mar '08

I noticed on version 1.3 (DISPATCHER MODULE) you can set the gateways off Mysql DB, the question is: will be necessary to restart openser every time I add/change the gateway's IP on the DB table (Mysql)?

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[Serusers] downloaded the latest firmware
by hamdy shibl 11 Mar '08

11 Mar '08

i need to downloaded the latest firmware please sedn the ftp link --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

1 0

[Serusers] 转发： How to Radius of acc modules
by Jerry.Zhao 11 Mar '08

11 Mar '08

Hi I have build SER to follow the install guide, but the radius cannot working, I have modifly the Makefile of acc's moudle too and got error message that "ERROR: load_module: could not open module </usr/local/lib/ser/modules/acc.so>: /usr/local/lib/ser/modules/acc.so: undefined symbol: rc_avpair_add" when ser has been run. I have only want know how to do acc's radius function to work. With best regards; -------------------------- Jerry.Zhao ___________________________________________________________ 雅虎邮箱传递新年祝福，个性贺卡送亲朋！ http://cn.mail.yahoo.com/gc/index.html?entry=5&souce=mail_mailletter_tagline

1 0

[Serusers] Fwd: SIP Express Bundle ready-to-go idea, BUILD FROM CVS on RH -STATUS
by Mike Trest - Personal 11 Mar '08

11 Mar '08

To All, Thanks for the suggestions. I have complete a working build from CVS on RH environment. I had to change my hat which said "Enterprise Tester" and put on the hat that says "Software Developer" and read thru the code in CVS tree. I was able to chase down the issues with "db_key_t" and a few other db things. I discovered that the CVS tree was NOT (as I had expected) in a state where "everything is in order". I discovered a lot of the DB material in a state of change between previous generations to the new 2.0 / 2.1 generation. I took Jiri's suggestion and built a config based on the ser-oof.cfg . I only got into issues when I added other modules not on that specific loadmodule list. I am still waiting for input from the more experienced developers as to exactly what to include in The Bundle. I am still hopeful. Regards, ..mike.. >Date: Fri, 22 Feb 2008 10:59:15 -0500 >To: serusers(a)lists.iptel.org >From: Mike Trest - Personal <Mike(a)Trest.COM> >Subject: [Serusers] SIP Express Bundle >ready-to-go idea, BUILD FROM CVS on RH ? > >Hi, >Followup on prior message: > >When doing make .... modules I found a similar >error that did not involve a system include. > >The offending line in avpops_db.h is: > >int db_store_avp( db_key_t *keys, db_val_t *vals, int n, char *table); > >Both examples below and this example are related to forward declarations >with pointers to structures. > >Regards, ..mike.. > > > > > >make[1]: Entering directory `/usr/local/src/ser-2.0.x-CVS/modules/avpops' >gcc -fPIC -DPIC -g -O9 >-funroll-loops -Wcast-align >-minline-all-stringops -falign-loops >-ftree-vectorize >-mtune=athlon64 -Wall -DNAME='"ser"' >-DVERSION='"2.1.0-dev17-tcp"' -DARCH='"i386"' >-DOS='linux_' -DOS_QUOTED='"linux"' >-DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 >-D__OS_linux -DSER_VER=2001000 >-DCFG_DIR='"/usr/local/etc/ser/"' >-DMODS_DIR='"/usr/local/lib/ser/modules/"' >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK >-DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >-DHAVE_RESOLV_RES -DUSE_DNS_CACHE >-DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST >-DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK >-DADAPTIVE_WAIT >-DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM >-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN >-DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL >-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H >-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER >-DHAVE_EPOLL -DHAVE_SIGIO_RT >-DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c avpops.c -o avpops.o >In file included from avpops_impl.h:41, > from avpops_parse.h:38, > from avpops.c:50: >avpops_db.h:66: error: expected â)â before â*â token >make[1]: *** [avpops.o] Error 1 >make[1]: Leaving directory `/usr/local/src/ser-2.0.x-CVS/modules/avpops' >make: *** [modules] Error 1 > > >======================================================================== >Hi, >I am doing a ser build from CVS check-out on a RH based linux with gcc 4.1.1. >The build commands are this. >make clean >make proper >make group_include="standard standard-dep stable mysql" >When I get to any core or module section code >which included futexlock.h header, I get the following: > >gcc -g -O9 -funroll-loops -Wcast-align >-minline-all-stringops -falign-loops >-ftree-vectorize -mtune=athlon64 >-Wall -DNAME='"ser"' >-DVERSION='"2.1.0-dev17-tcp"' -DARCH='"i386"' >-DOS='linux_' -DOS_QUOTED='"linux"' >-DCOMPILER='"gcc 4.1.1"' -D__CPU_i386 >-D__OS_linux -DSER_VER=2001000 >-DCFG_DIR='"/usr/local/etc/ser/"' >-DMODS_DIR='"/usr/local/lib/ser/modules/"' >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK >-DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >-DHAVE_RESOLV_RES -DUSE_DNS_CACHE >-DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST >-DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK >-DADAPTIVE_WAIT >-DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM >-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN >-DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL >-DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H >-DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER >-DHAVE_EPOLL -DHAVE_SIGIO_RT >-DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c core_cmd.c -o core_cmd.o >In file included from mem/../futexlock.h:48, > from mem/../lock_ops.h:85, > from mem/shm_mem.h:76, > from core_cmd.c:32: >/usr/include/linux/futex.h:96: error: expected â)â before â*â token >/usr/include/linux/futex.h:98: error: expected â)â before â*â token >make: *** [core_cmd.o] Error 1 >========================================================================= > >The offending lines came from the system >header <futex.h>. The offending lines are: > >long do_futex(u32 *uaddr, int op, u32 val, >unsigned long timeout, u32 *uaddr2, u32 val2, u32 val3); > >extern int handle_futex_death(u32 *uaddr, struct task_struct *curr, int pi); > >By pointing to an alternate version of <futex.h> >where these lines are commented out >I can compile all the way thru with non-fatal warnings. > >My questions are: >1) can anyone tell me what are the equivalent lines in the DEB version where >this code/compile was most recently developed? >2) has anyone got a fix for this error? > >..mike.. > > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers

3 5

[OpenSER-Users] Openser DB with Mysql-cluster
by Antonio Reale 10 Mar '08

10 Mar '08

Hi all, I'm trying to setup openser with mysql cluster. I see that the default db engine used by openser tables is MyISAM, I need to change it with the ndbcluster engine or I can leave it as is? Thanks. Regards, Antonio

4 6

[Serusers] How to make only one module
by Mike Trest - Personal 10 Mar '08

10 Mar '08

Hi, I am building SER a few modules from sources that need mysql.so. I understand the groupings standard and standard_dep in the makefile. What I do not find is how to build just one module (mysql.so) without going all the way to the top and rebuilding the whole kit from scratch. I have looked at the new make_and_install (which I like) and have not yet found how to build just one without resetting the whole config or tinkering with the master Makefile. Any suggestions? ..mike..

2 1

[OpenSER-Users] OpenSER Summit & Pavilion 2008
by Bogdan-Andrei Iancu 10 Mar '08

10 Mar '08

Hi everybody, I'm glad to make public the follow up of the OpenSER related events. I'm sorry I was not able to do it sooner, but making all the arrangements took more than estimated ;). The second OpenSER Summit will take place in San Jose, USA , on the 17th of March, 2008. This is the first US edition of the OpenSER Summit - following the European edition, a large segment of the OpenSER community have continuously asked for an edition to address also the American community. The summit is again hosted inside the VoN Spring 2008 Exhibition, as a pre-conference event. All participants to register via OpenSER site before Friday, March 7th, will get free access to the OpenSER event. More details about the OpenSER Summit, agenda and registration process, are to be found on the project web page: http://www.openser.org/mos/view/OpenSER-Summit-2008/ This OpenSER Summit edition is sponsored by a parallel OpenSER related event, the OpenSER Pavilion . The pavilion is a common exhibiting area - booth 1027, inside VoN expo, gathering, under the OpenSER name, six different companies working or using the project. Their direct statement to the visitor is: "Yes, we do open source in our business and OpenSER is a success recipe and we want to share this with you" More details about the OpenSER Pavilion, location and participants, are to be found on the project web page: http://www.openser.org/mos/view/OpenSER-Pavilion-2008/ The dual event, OpenSER Summit & Pavilion is new concept of a more complex event, aiming to create a larger diversity and to give more power to the understanding of the OpenSER project. Again, I'm very happy we were able to continue the path opened with the OpenSER Summit in Berlin and show the continuity in the OpenSER project development. Also, thanks to the industry support for the project, we were able again to offer a free participation to the OpenSER Summit and allow a good community convergence. For any questions about the OpenSER Summit & Pavilion do not hesitate to contact me. Best regards, Bogdan

1 1

[OpenSER-Users] Presence modules crash my server ...
by G.O. Jensen 10 Mar '08

10 Mar '08

So, once again I tried to get presence to work... the result being OpenSER crashes immediately when I try to do an INVITE. r[HANDLE_PRESENCE] Subscribe - (158.38.38.115) sip:gojensen@test-sip.uninett.no SUBSCRIBE sip:57122@test-sip.uninett.no (ID=3c2688d10849-w1mwr2kl7l8n) ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: ^Ievent= dialog INFO:core:handle_sigs: child process 22386 exited by a signal 11 INFO:core:handle_sigs: core was generated INFO:core:handle_sigs: terminating due to SIGCHLD INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received INFO:core:sig_usr: signal 15 received NOTICE:presence:destroy: destroy module ... My route script looks like this for this part; sl_send_reply("100","Trying"); if (is_method("PUBLISH")) { handle_publish(); t_release(); } else if (is_method("SUBSCRIBE")) { xlog("L_INFO","r[HANDLE_PRESENCE] Subscribe - ($si) $fu $rm $ru (ID=$ci)"); handle_subscribe(); t_release(); } else if ((is_method("OPTIONS")) && (!uri=~"sip:.*[@]+.*")) { options_reply(); }; As you can see it crashes upon trying to do a handle_subscribe() I'm currently running openser-1.3.x-svn-20080307-141101.tgz which was the latest yesterday when I tried it... I've taken care of the core file but my debugging skills are limited to c'n'p logs and doing what I'm told... The phone in question is a SNOM320 which sends subscribes to every number it dials... presumably for some presence stuff. - gojensen / UNINETT

2 1

[OpenSER-Users] Presence status?
by Zappasodi Daniele 10 Mar '08

10 Mar '08

Hello, I'm doing some tests on presence functionality in openser, but it doesn't seem to work. I'm using openser 1.3.0 (modules: presence, xcap_client, presence_xml with force_active=1) with eyebeam and xlite and with few tests I have found a lot of serious problems: 1) Query fail in mysql Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:mysql:db_mysql_query: error while submitting query Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:presence:get_db_subs_auth: while querying watchers table Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:presence:handle_subscribe: getting subscription status from watchers table Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:mysql:db_mysql_submit_query: driver error: Commands out of sync; you can't run this command now Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:mysql:db_mysql_query: error while submitting query Feb 6 17:05:00 SAM-IP ser[3865]: ERROR:presence:get_p_notify_body: querying presentity and Duplicate entry 'sip:10208@10.45.10.36-10298-10.45.10.36-presence' for key 2 2) Problem with PUBLISH (sometimes with segmentation fault): Feb 6 16:29:17 SAM-IP ./ser[3807]: route <sip:10206@10.45.8.249> <PUBLISH> <sip:10206@10.45.8.249> Feb 6 16:29:17 SAM-IP ./ser[3805]: ERROR:presence:update_presentity: No E_Tag match Feb 6 16:29:17 SAM-IP kernel: ser: unhandled page fault at pc=0x40117334, lr=0x4041204c (bad address=0x47a9e6e4, code 0) Feb 6 16:29:17 SAM-IP kernel: pc : [<40117334>] lr : [<4041204c>] Tainted: PF Feb 6 16:29:17 SAM-IP kernel: sp : bfffe3fc ip : 404367ac fp : 000d62bc Feb 6 16:29:17 SAM-IP kernel: r10: 404366b4 r9 : 407e2470 r8 : 0011fcb8 Feb 6 16:29:17 SAM-IP kernel: r7 : 00000000 r6 : 00000001 r5 : 0011fc60 r4 : 0000000c Feb 6 16:29:17 SAM-IP kernel: r3 : 000d62c0 r2 : 0011fcb8 r1 : 47a9e6e4 r0 : 47a9e6e6 Feb 6 16:29:17 SAM-IP kernel: Flags: Nzcv IRQs on FIQs on Mode USER_32 Segment user Feb 6 16:29:17 SAM-IP kernel: Control: 39FF Table: 033AC000 DAC: 00000015 3) Problem with SUBSCRIBE Feb 5 23:52:01 SAM-IP ser[1248]: ERROR:presence:handle_subscribe: Missing or unsupported event header field value Feb 5 23:52:01 SAM-IP ser[1248]: ERROR:presence:handle_subscribe: ^Ievent= dialog 4) Mar 6 17:02:16 localhost ./openser[12810]: ERROR:presence:get_stored_info: record not found in hash_table Mar 6 17:02:16 localhost ./openser[12810]: ERROR:presence:handle_subscribe: getting stored info So I would like to know the status of the presence project. ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

4 4

[OpenSER-Users] "user not found" and "user not exist"
by Amit Vijayvargiya 10 Mar '08

10 Mar '08

Hi All, How we can differentiate between "user not found" and "user not exist". what changes done in openser.cfg. currently i m getting 404 response in both the cases "user not found" and "user not exist" Thanks & Regards, Amit

3 4

[OpenSER-Users] about:openser and freeradius, please help me!
by htl_22 10 Mar '08

10 Mar '08

Dear openser.org: I have studied "Build SIP-based VOIP Service With RADIUS AAA Using OpenSER And FreeRadius" web pages.This documentation is valid for OpenSER v1.0.1 and FreeRadius v1.1.0. My OS is unbuntu 6.10.The radius and openser install are successful.MySQl work well,and the radius start also,but OpenSER don't start! The error information is: tr<0xb7f603a0> 0(5988) ); 0(5988) } else { 0(5988) }; 0(5988) exit( 0(5988) type<0> 0(5988) ); 0(5988) exit( 0(5988) type<0> 0(5988) ); 0(5988) Listening on udp: 10.1.37.22 [10.1.37.22]:5060 Aliases: udp: htl-desktop:5060 *: openser.org:* WARNING: no fork mode 0(5988) fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(5988) fm_malloc_init: F_HASH_SIZE=2067, fm_block size=16548 0(5988) fm_malloc_init(0xb5a9f000, 33554432), start=0xb5a9f000 0(5988) shm_mem_init: success 0(0) core dump limits set to 4294967295 0(0) DEBUG: init_mod: mysql 0(0) mysql: MySQL client version is 5.0.24a 0(0) DEBUG: init_mod: sl_module stateless - initializing 0(0) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(0) unixsock_register_cmd: New command (sl_stats) registered 0(0) DEBUG: MD5 calculated: 8f36450dbf9496a39daffb98efb4221a 0(0) DEBUG: init_mod: tm 0(0) TM - (size of cell=2832, sip_msg=768) initializing... 0(0) Call-ID initialization: '2ff5b7ea' 0(0) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(0) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(0) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(0) DEBUG: register_fifo_cmd: new command (t_reply) registered 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered 0(0) unixsock_register_cmd: New command (t_hash) registered 0(0) unixsock_register_cmd: New command (t_reply) registered 0(0) DEBUG: lock_initialize: lock initialization started 0(0) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(0) unixsock_register_cmd: New command (t_stats) registered 0(0) DEBUG: MD5 calculated: 741ad5dfb14662b78989a07dc5c8b59e 0(0) DEBUG: MD5 calculated: eb9313e49dd2c8bc3fd8ed1343760911 0(0) DEBUG: init_mod: rr 0(0) rr - initializing 0(0) DEBUG: init_mod: maxfwd 0(0) Maxfwd module- initializing 0(0) DEBUG: init_mod: avpops 0(0) AVPops - initializing 0(0) DEBUG: init_mod: usrloc 0(0) usrloc - initializing 0(0) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(0) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: found <db_use_table> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_init> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_close> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_query> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_raw_query> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_free_result> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_insert> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_delete> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_update> in module mysql [/usr/local/lib/openser/modules/] 0(0) find_mod_export: found <db_replace> in module mysql [/usr/local/lib/openser/modules/] 0(0) DEBUG: init_mod: registrar 0(0) registrar - initializing 0(0) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_bind_usrloc> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_register_udomain> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_insert_urecord> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_delete_urecord> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_get_urecord> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_lock_udomain> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_unlock_udomain> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_release_urecord> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_insert_ucontact> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_delete_ucontact> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_get_ucontact> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_update_ucontact> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_register_watcher> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_unregister_watcher> in module usrloc [/usr/local/lib/openser/modules/] 0(0) find_export: found <ul_register_ulcb> in module usrloc [/usr/local/lib/openser/modules/] 0(0) DEBUG: init_mod: textops textops - initializing 0(0) DEBUG: init_mod: xlog 0(0) XLOG: initializing ... 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_export: found <load_tm> in module tm [/usr/local/lib/openser/modules/] 0(0) ERROR: acc: can't get code for the Sip-Src-IP attribute 0(0) init_mod(): Error while initializing module acc ERROR: error while initializing modules 0(0) XLOG: destroy module ... 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(5988) shm_mem_destroy 0(5988) destroying the shared memory lock I wish that you can hint me and help me solving the problem! If openser's version is 1.1.0 and it need the "presence" module, Which freeradius version will be choosed? sincerely tlhe

2 1

[Serusers] SER + SKYPE
by Giovanni Chirico 10 Mar '08

10 Mar '08

Hi, There is someone that are using SER with SKYPE, for example with SKIP2PBX or SKYSTONE. I would like to use SER with SkyStone but I don't Know how to configure SER to works fine. Can you help me? Bye, G. Chirico

1 0

[Serusers] Few Questions
by Jai Rangi 10 Mar '08

10 Mar '08

Hello, I see some of these errors and warnings in my log file on SER. WARNING: warning skipped -- too big ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established ERROR: warning_builder: buffer size exceeded How can I get rid of these. Also I had a question on children=4 I know this has been asked before and has been answered. This is what I found online. *"children=4" mean that there will be 4 UDP, 4 TCP and 4 loopback **listenters on your SER, when you start it (you can check with "serctl ps**s" to verify). * Question is does this have any affect on the number of simultaneous calls or sip users etc. Say I want to make sure my system works without any problem for 20K user and 5000 simultaneous calls. What can be pros and cons if I set children to 10, 50 or 100. I will appreciate if anyone can educate me little more on this. Thank you, -Jai

1 0

[OpenSER-Users] isflagset from other peer ?
by David Loh 10 Mar '08

10 Mar '08

Hi All, I'm facing a situation for P2P call where it need to determine both caller and callee network type, except for both parties having direct (internet) connection (on Public IP), the rest of combinations will implement force_rtp_proxy. (the 4 combinations are: NAT-> NAT, NAT -> Public IP, Public IP -> NAT, Public IP -> Public IP). Wondering if I can use isflagset or fetch flag from remote peer (callee) ? or is there any function that allow this to be happen ? Thanks, David Loh

2 2

[OpenSER-Users] HELP - OpenSER 1.3 not parallel forking?
by Daryl Sanders 09 Mar '08

09 Mar '08

Hi All, I am testing parallel forking so I can setup ring groups, but it seem I cannot get OpenSER 1.3 to parallel fork contacts of the same q-value. It always seem to use the first record found in MySQL, and ignores all others. I have tried adding append_branch after lookup("location") , but that did not seem to help. I've also done ethereal captures and confirmed that it is only sending an INVITE to the first contact in MySQL (I've tried db_mode 3 and 1). I'm know I'm missing something, but not sure what. I've included relevant portions of my test config below. Any help would be greatly appreciated. Regards, Daryl route { # ----------------------------------------------------------------- # Sanity Check, NAT Check & General Routing # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; } if (nat_uac_test("19")) { setbflag(6); xlog("L_INFO", "NAT CLIENT - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); } if (method=="INVITE" || method=="ACK" || method=="BYE") { setflag(1); } if (method!="REGISTER") { record_route(); } if (loose_route()) { xlog("L_INFO", "LOOSE ROUTING - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (!t_relay()) { sl_reply_error(); } exit; } if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; } route(1); exit; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; } t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); } exit; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; } consume_credentials(); if (!save("location")) { sl_reply_error(); } xlog("L_INFO", "REGISTER - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- xlog("L_INFO", "INVITE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # Trusted IP if (src_ip==xxx.xxx.xxx.xxx) { strip(2); xlog("L_INFO", "FROM PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); } else { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } consume_credentials(); } if (lookup("location")) { xlog( "L_NOTICE", "USER FOUND - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); } else { xlog( "L_NOTICE", "USER NOT FOUND! - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); } if (isbflagset(6) || !lookup("location")) { xlog( "L_NOTICE", "RELAYING MEDIA - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); route(4); exit; } route(1); exit; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- force_send_socket(xxx.xxx.xxx.xxx:5060); xlog("L_NOTICE", "FORWARDING TO PSTN\n" ); ds_select_domain("1","0"); route(1); exit; }

2 7

[OpenSER-Users] branch specific destination uri?
by jh＠tutpro.com 09 Mar '08

09 Mar '08

a question: is it possible to specify destination uri ($du) per branch, i.e., if i set $du a value in branch route, does it apply only to that branch when t_relay is called? -- juha

1 1

[OpenSER-Users] append_branch and $T_branch_idx behavior
by Ovidiu Sas 07 Mar '08

07 Mar '08

Hello, I was playing wit append_branch and I run into this issue. Here's a snippet of the code: route[2] { $var(uri) = $ru; append_branch(); $ru = $var(uri) + ";transport=udp"; t_on_branch("2"); } branch_route[2] { xlog("L_INFO", "entering branch[$T_branch_idx] for $ru\n"); if ($T_branch_idx == 1) { xlog("L_INFO", "dropping branch[$T_branch_idx]\n"); drop(); } } In the logs, I got: entering branch[1] for sip:<user>@<IP>;transport=udp dropping branch[1] entering branch[1] for sip:<user>@<IP> dropping branch[1] ERROR:tm:t_forward_nonack: failure to add branches Is this the expected result? It seems that we do have two branches, since we enter twice in the branch_route, but the $T_branch_idx doesn't seem to be updated. Or, it is supposed to be updated? Any hints would be appreciated. BTW: this is with 1.3 code base. Regards, Ovidiu Sas

2 1

[Serusers] help for "parse_orig_uri"
by Liu Wenlong 07 Mar '08

07 Mar '08

I compiled & installed ser-0.9.7. But cannot start it. The Log is: 0(28624) loading module /usr/local/lib/ser/modules/acc.so 0(28624) ERROR: load_module: could not open module </usr/local/lib/ser/modules/acc.so>: /usr/local/lib/ser/modules/acc.so: undefined symbol: parse_orig_uri 0(28624) parse error (41,13-14): failed to load module 0(28624) set_mod_param_regex: No module matching acc found | 0(28624) parse error (43,19-20): Can't set module parameter 0(28624) set_mod_param_regex: No module matching acc found ... And in compiling log, there is a acc.c:291: warning: implicit declaration of function 'parse_orig_uri' it seems that I lost some devel packages. who can tell me in which package defined "parse_orig_uri"? My OS is CentOS 5.1 x64 Thank you very much! Liu Wenlong -- aRES iN aNYWhere

1 0

Re: [OpenSER-Users] issue - remote sip client must re-register after failover in Openser Registrar HA config
by Naija Man 07 Mar '08

07 Mar '08

Thanks Patrick, I have re-configured a common database for the servers and I am still having the same issues. Can this be an issue with PAT at the Load balancer after failover since the INVITE is coming from openser_2 and going to the same IP:port associated with openser_1. I am also open to trying other methods to achieve OpenSER registrar HA if any available. Thanks. Buki A > > > ---------- Forwarded message ---------- > > From: Patrick Miccio <pmiccio(a)upcbroadband.com> > > To: <users(a)lists.openser.org> > > Date: Thu, 6 Mar 2008 11:04:40 +0100 > > Subject: Re: [OpenSER-Users] issue - remote sip client must re-register > > after failover in Openser Registrar HA config > > Hi, > > > > you need to have a common mysql table for the registrations so that when > > OpenSER_2 taks over it has the informations > > OpenSER_1 wrote in the DB. > > > > you can achieve this with a single mysql server, or better with a > > cluster ;) > > > > > > cheers, > > > > Patrick. > > > > > > > Hi friends, > > > > > > I am having problems setting up seamless openser registrar failover. I > > am trying to setup OpenSER registrar HA using > > > 2 openser registrars v1.3.0 behind a Netscaler citrix Load balancer > > and I cant get the failover scenario to work > > > without re-registration of the remote sip client. > > > > > > I am desperately looking for tips to make this configuration work or > > any other ideas to achieve SIP registrar HA > > > using OpenSER. Thanks. > > > > > > > > > > > > The setup is configured as active-passive and the 2nd OpenSER becomes > > active only after a Openser_1 fails and the LB > > > fails-over to the 2nd. The 2 openser servers are SIP registrars for my > > voice network. > > > > > > REGISTRATION: > > > remote SIP UA ---> remote firewall ---> internet ---> my firewall ---> > > citrix NA loadbalancer (VIP) ----> openSER_1 > > > > > \---> > > > openSER_2 > > > > > > 1. Remote SIP client configures the LB VIP as the SIP proxy and > > registers > > > 2. the citrix LB gets the registrations and forwards to the active > > openser (say, openser_1) > > > 3. openser_1 stores the contact as MIP:port in the mysql database > > > 4. openser_1 replicates the registration to openser_2 (t_replicate) > > > > > > OUT-GOING CALL - before fail-over > > > remote SIP UA <--- remote firewall <--- internet <--- my firewall <--- > > citrix NA loadbalancer (VIP) <----> openSER_1 > > > > > ^ > > > openSER_2 | > > > > > Asterisk > > > > > > 1. call (INVITE) is generated from asterisk and sent to LB VIP > > > 2. the citrix LB gets the INVITE and forwards to the active openser > > (say, openser_1) > > > 3. openser_1 looks-up the URI in the MySQL database, finds the > > MIP:port that the registration was received from the > > > LB and forwards the INVITE to that destination. 4. the LB receives > > the INVITE and forwards it to the remote SIP > > > client > > > > > > > > > OUT-GOING CALL - after fail-over > > > remote SIP UA ---x remote firewall ---x internet ---x my firewall > > ---x citrix NA loadbalancer (VIP) ----x openSER_1 > > > > > ^ > > > \-----< openSER_2 | > > > > > Asterisk > > > > > > 1. call (INVITE) is generated from asterisk and sent to LB VIP > > > 2. the citrix LB gets the INVITE and forwards to the openser now > > active (openser_2) > > > 3. openser_2 looks-up the URI in the MySQL database, finds the same > > MIP:port of the LB that was replicated from > > > openser_1 and forwards the INVITE to that destination. 4. the LB > > receives the INVITE and drops the packet without > > > forwarding it out to the remote SIP client > > > > > > > > > > > > For the outgoing call to work: the remote sip client either has to > > re-register, so the registration gets passed to > > > openser_2. OR, openser_1 is made active again. > > > > > > > > > Configuration of OpenSER: > > > > > > REGISTRATION: > > > # If this is a registration from the public UA, > > > # then we store it > > > fix_nated_contact(); # use ip address of the packet > > instead of the sip message > > > force_rport(); # append rport. might not be necessary > > > > > > # save Registration in location table > > > save("location"); > > > xlog("L_DBG", "SipMsgId[$mi] Save Registration: > > To[$tu] Contact[$ct]\n"); > > > append_hf("CtlRegFwd: \r\n"); > > > # replicate the register message to the backup > > registrar > > > # if I am the backup, ignore the message > > > if(!t_replicate("sip:openser_2:5060")) { > > > xlog("L_WARN", "SipMsgId[$mi] Fail to replicate > > Contact[$ct] to fail over registrar\n"); > > > > > > OUTGOING CALL: > > > > > > # Do lookup so it gets to the correct NATed destination > > > if (!lookup("location")) > > > { > > > # this user has not REGISTERed or registration has > > expired > > > xlog("L_INFO", "SipMsgId[$mi] Send 404-Not Found (Msg > > rejected because user not found)\n"); > > > sl_send_reply("404", "Not Found"); > > > exit; > > > }; > > > > > > # keep track of reply > > > t_on_reply("11"); > > > if (!t_relay()) > > > { > > > xlog("L_DBG", "SipMsgId[$mi] Failed t_relay. Do > > sl_reply_error.\n"); > > > sl_reply_error(); > > > }; > > > > > > > > > > > > Thank you for your anticipated help. > > > > > > Regards, > > > > > > Buki A > > > > > > >

1 0

[OpenSER-Users] ERROR:core:main: bad user name/uid number: -u openser
by bsb093 07 Mar '08

07 Mar '08

Hi all, I am still trying to run OpenSER. Now I see the following error: ERROR:core:main: bad user name/uid number: -u openser I am trying to install version 1.3 with MySQL. I have succeeded to create the db but when trying to run OpenSER I still have this error. I have found in the forum that maybe I don't have openser user in /etc/passwd and that is true but I can't figure how to create that user. Any suggestions? Thanks in advance, Shine -- View this message in context: http://www.nabble.com/ERROR%3Acore%3Amain%3A-bad-user-name-uid-number%3A--u… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

4 9

Re: [OpenSER-Users] Users Digest, Vol 34, Issue 17
by Naija Man 07 Mar '08

07 Mar '08

Thanks Patrick, I have re-configured a common database for the servers and I am still having the same issues. Can this be an issue with PAT at the Load balancer after failover since the INVITE is coming from openser_2 and going to the same IP:port associated with openser_1. I am also open to trying other methods to achieve OpenSER registrar HA if any available. Thanks. Buki A > ---------- Forwarded message ---------- > From: Patrick Miccio <pmiccio(a)upcbroadband.com> > To: <users(a)lists.openser.org> > Date: Thu, 6 Mar 2008 11:04:40 +0100 > Subject: Re: [OpenSER-Users] issue - remote sip client must re-register > after failover in Openser Registrar HA config > Hi, > > you need to have a common mysql table for the registrations so that when > OpenSER_2 taks over it has the informations > OpenSER_1 wrote in the DB. > > you can achieve this with a single mysql server, or better with a cluster > ;) > > > cheers, > > Patrick. > > > > Hi friends, > > > > I am having problems setting up seamless openser registrar failover. I > am trying to setup OpenSER registrar HA using > > 2 openser registrars v1.3.0 behind a Netscaler citrix Load balancer and > I cant get the failover scenario to work > > without re-registration of the remote sip client. > > > > I am desperately looking for tips to make this configuration work or any > other ideas to achieve SIP registrar HA > > using OpenSER. Thanks. > > > > > > > > The setup is configured as active-passive and the 2nd OpenSER becomes > active only after a Openser_1 fails and the LB > > fails-over to the 2nd. The 2 openser servers are SIP registrars for my > voice network. > > > > REGISTRATION: > > remote SIP UA ---> remote firewall ---> internet ---> my firewall ---> > citrix NA loadbalancer (VIP) ----> openSER_1 > > > \---> > > openSER_2 > > > > 1. Remote SIP client configures the LB VIP as the SIP proxy and > registers > > 2. the citrix LB gets the registrations and forwards to the active > openser (say, openser_1) > > 3. openser_1 stores the contact as MIP:port in the mysql database > > 4. openser_1 replicates the registration to openser_2 (t_replicate) > > > > OUT-GOING CALL - before fail-over > > remote SIP UA <--- remote firewall <--- internet <--- my firewall <--- > citrix NA loadbalancer (VIP) <----> openSER_1 > > > ^ > > openSER_2 | > > > Asterisk > > > > 1. call (INVITE) is generated from asterisk and sent to LB VIP > > 2. the citrix LB gets the INVITE and forwards to the active openser > (say, openser_1) > > 3. openser_1 looks-up the URI in the MySQL database, finds the > MIP:port that the registration was received from the > > LB and forwards the INVITE to that destination. 4. the LB receives the > INVITE and forwards it to the remote SIP > > client > > > > > > OUT-GOING CALL - after fail-over > > remote SIP UA ---x remote firewall ---x internet ---x my firewall ---x > citrix NA loadbalancer (VIP) ----x openSER_1 > > > ^ > > \-----< openSER_2 | > > > Asterisk > > > > 1. call (INVITE) is generated from asterisk and sent to LB VIP > > 2. the citrix LB gets the INVITE and forwards to the openser now > active (openser_2) > > 3. openser_2 looks-up the URI in the MySQL database, finds the same > MIP:port of the LB that was replicated from > > openser_1 and forwards the INVITE to that destination. 4. the LB > receives the INVITE and drops the packet without > > forwarding it out to the remote SIP client > > > > > > > > For the outgoing call to work: the remote sip client either has to > re-register, so the registration gets passed to > > openser_2. OR, openser_1 is made active again. > > > > > > Configuration of OpenSER: > > > > REGISTRATION: > > # If this is a registration from the public UA, > > # then we store it > > fix_nated_contact(); # use ip address of the packet > instead of the sip message > > force_rport(); # append rport. might not be necessary > > > > # save Registration in location table > > save("location"); > > xlog("L_DBG", "SipMsgId[$mi] Save Registration: To[$tu] > Contact[$ct]\n"); > > append_hf("CtlRegFwd: \r\n"); > > # replicate the register message to the backup registrar > > # if I am the backup, ignore the message > > if(!t_replicate("sip:openser_2:5060")) { > > xlog("L_WARN", "SipMsgId[$mi] Fail to replicate > Contact[$ct] to fail over registrar\n"); > > > > OUTGOING CALL: > > > > # Do lookup so it gets to the correct NATed destination > > if (!lookup("location")) > > { > > # this user has not REGISTERed or registration has > expired > > xlog("L_INFO", "SipMsgId[$mi] Send 404-Not Found (Msg > rejected because user not found)\n"); > > sl_send_reply("404", "Not Found"); > > exit; > > }; > > > > # keep track of reply > > t_on_reply("11"); > > if (!t_relay()) > > { > > xlog("L_DBG", "SipMsgId[$mi] Failed t_relay. Do > sl_reply_error.\n"); > > sl_reply_error(); > > }; > > > > > > > > Thank you for your anticipated help. > > > > Regards, > > > > Buki A > > >

1 0

Re: [Serusers] General Question on BYE
by Mike Trest - Personal 07 Mar '08

07 Mar '08

Thanks Greger. I set debug to 9 and found the BYE inside the log file. It was seen by SER. Now I have a trail on the culprit. ..mike.. At 11:24 AM 3/7/2008, Greger Viken Teigre wrote: >Easy answer: none. >If there is a packet on a port SER is listening to, it will try to >parse it. If you don't see it with xlog, there should be an error >log message or it is dropped in a firewall. >g-) > >Mike Trest - Personal wrote: >>Hi, >> >>I have see recent discussions on this topic among developers >>but I am trying to ask a general question without getting into >>ser.cfg and 0.9x vs 2.x comparisons. >> >>My Question: >> >>What are the practical reasons why SER 2.0 would simply ignore "BYE" >>messages sent to it from a gateway when the user terminates a call? >> >> user => gateway => ser => gateway >> >>Good connection and conversation. When the user hangs up >>Gateway sends BYE message. SER does not reply or forward >>to destination gateway. SER never even sees it. >> >>I see the BYE message on tshark but it does not appear in ser.cfg >>with xlog() statements right at the top or route{}. >> >>Anyone with an opinion or suggestion to offer? >> >>..mike.. >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> > >

1 0

Re: [Serusers] [Sems] Poll on dropping unix socket interface ("unixsockctrl")
by samuel 07 Mar '08

07 Mar '08

In the past I experienced "bad performance" using unixsoctrl with medium load so I think it's worth making users moving to new interfaces. vote for removing it samuel. 2008/3/7, Juha Heinanen <jh(a)tutpro.com>: > > Raphael Coeffic writes: > > > - reduces configuration and run-time complexity > > > this proved to be true. my openser configuration became much simpler > when i migrated it from unixsockets to sip. > > > -- juha > > _______________________________________________ > Sems mailing list > Sems(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/sems >

3 2

[OpenSER-Users] How to access pseudovariables in exported functions?
by Peter Lemenkov 07 Mar '08

07 Mar '08

Hello All! I need to put more than 2 parameters in my module, and there are pseudovariables among them. If I find a way to access them within my function then two parameters will be enough. Namely I need to access $fU, $tU and $rU. Using parse_headers function I can only get $fu, $tu and $ru. -- With best regards!

2 1

[OpenSER-Users] ERROR: presence_xml:get_final_notify_body?
by KevinKinnan 07 Mar '08

07 Mar '08

My testbed is combination of openser and openxcap. openser.cfg file is similar with the example on the website of OpenXCAP. After updating published information by sending PUBLISH, no NOTIFY message is sent to client-side. I got the following errors(debug=3) in syslog file. Any ideas anyone? --------------------------------------------------------------------------------------- ERROR:presence_xml:get_final_notify_body: while extracting the transformation node ERROR:presence_xml:pres_apply_auth: in function get_final_notify_body ERROR:presence:send_notify_request: in function apply_auth_nbody ERROR:presence:notify: sending Notify not successful ERROR:presence:publ_notify: Could not send notify for presence --------------------------------------------------------------------------------------- Regards, Kevin _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

2 1

Re: [OpenSER-Users] OpenSER + OpenXcap for pres-rules: no NOTIFY
by Mircea Amarascu 07 Mar '08

07 Mar '08

2 1

[OpenSER-Users] CANCEL with Route headers
by Andreas Granig 07 Mar '08

07 Mar '08

Hi all, I've a Cisco PGW interconnecting with an OpenSER server. The INVITE is sent to OpenSER, which does some record-routing via some different hops and then sends back the 180 from the called party. If the calling party now cancels the call, the PGW tries to loose-route the CANCEL by adding the Route header to the CANCEL and altering the R-URI, which is rejected by my OpenSER since I don't allow initial loose-routing. So the question is: is it valid for a CANCEL to carry a Route header which wasn't available in the INVITE? §9.1 of RFC3261 says it MUST contain the Route header if it was present in the request being cancelled. What about the other way around? I heavily suspect it's violating the RFC since CANCEL should be more or less a 1:1 copy of the INVITE, but can anyone confirm this? If it violates the RFC, anyone with some Cisco knowledge who knows how to disable this behavior, or is it a known bug? Cheers, Andreas

4 3

[OpenSER-Users] "noisy_ctimer" parameter in TM module
by Bogdan-Andrei Iancu 07 Mar '08

07 Mar '08

Hi, I want to get some feedback from the users regarding one of the TM parameters: "noisy_ctimer" . This parameter, is disabled, would try to let a call to ring for ever (openser will never give internal timeout). But, the parameter is automatically turned on (disregarding the script setting) in certain conditions: - parallel forking is done - a failure route is set - a failure callback is set by other module (like acc, cpl, dialog, etc) - a fr_invite timeout was configured via AVP - some reply was already received - no other module explicitly asked for this (like siptrace, acc,osp) Following some discussion on the tracker, there is large support for removing this parameter as: 1) it is difficult to anticipate the final behaviour due complicated logic 2) due all dependencies, in 99% of the case, the param will be automatically turned on 3)it breaks the RFC3261, which make mandatory to have a final reply form a stateful proxy My question is: Is anybody having a good point in not removing this parameter (and having noisy_ctimer behaviour all the time)? I'm pushing this question only on users list, as between the developers, there is an consent in removing it ;) Regards, Bogdan

9 20

[OpenSER-Users] serializing parallel forking
by jh＠tutpro.com 07 Mar '08

07 Mar '08

a few days ago there was a question on if lcr module is able to serialize parallel forking of branches with the same q value. i replied that there is no such capability in lcr module. now that i have thought about the matter a bit, in my opinion the best solution is to write a separate serialize function that rewrites the q values rather than trying adding yet another feature to already too complex lcr module. -- juha

2 2

[Serusers] Cool New Website!
by Goran Donev 06 Mar '08

06 Mar '08

Cool New Website. www.dialaway4free.com I think they are using a combination of Ser and Asterisk to achieve this service. Everyone should look into it. Thanks

3 2

[OpenSER-Users] OpenSER & NAT Helper
by Ross Beer 06 Mar '08

06 Mar '08

Hi, I am looking to implement an OpenSER NAT helper. I need OpenSER to correct mistakes in SIP packets and pass them on to Asterisk. I do not want OpenSER to handle registrations. Does anyone have any good examples for this type of setup? Kind regards, Ross _________________________________________________________________ Who's friends with who and co-starred in what? http://www.searchgamesbox.com/celebrityseparation.shtml

3 3

Re: [OpenSER-Users] OpenSER & NAT Help
by Peter Lemenkov 06 Mar '08

06 Mar '08

2008/3/6, Ross Beer <ross_beer(a)hotmail.com>: > Just general sip message problems, i.e. if a user is behind a firewall. Take a look at the following commands: force_rport fix_nated_contact fix_nated_sdp force_rtp_proxy Don't forget also to mark (with setflag()) registered users behind NAT and fix replies in onreply_route. > I think I need to use the NAT Helper module, however I can not seam to > pass registrations on to Asterisk. BTW don't forget to CC to users(a)openser.org (this maillist badly configured) -- With best regards!

2 2

[Serusers] How to force IPtel.org SIP proxy not to include RTP proxy in Media path (achive peer-to-peer flow)
by girish kumar 06 Mar '08

06 Mar '08

Hi, I have 2 SIP based softphones (Xlite) registered on IPtel.org SIP service and testing the calls among them. These 2 softphones are behind different NAT and IPtel.org SIP proxy is replacing the SDP connection details in Call setup message (Invite and 200 OK) with RTPproxy IP/ports. Hence, RTPproxy is relaying the Voice media stream among them during the call. Although, both the softphones are behind different NAT address but these are not Symmetric NATs and hence RTP can directly flow between these softphone using STUN discovery. Is it possible to disable the RTP proxy behaviour of IPtel.org SER server for these 2 softphones through some configuration changes or any special header parameter in call setup message? In User Management web interface of IPtel.org, there is a option to specify the connectivity realm for a user account under My Account -> Other tab. The description of this field is "connectivity realm - behind the same NAT or possibility to communicate directly". I believe this field is playing some part in NAT transversal mechanism of Iptel.org SER implementation. Can anyone please describe the significance of this user account parameter? Can it be used to disable NAT transversal of media stream for some individual clients? I tried putting in same value of this field for 2 user accounts but Iptel.org SIP server still proxied media stream between these 2 user accounts through its RTP proxy. Thanks in advance. Girish ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

3 4

[OpenSER-Users] OpenSER Randomly crashes.
by Sergio Gutierrez 06 Mar '08

06 Mar '08

Hello everybody. I have written to list these days, I am still having the same symptom, so I apologize for writting again. My OpenSER 1.3 installation running on Solaris Sparc is facing random and unexpected crashes, in appearance related to timer process. The last core presents the following backtrace #0 0xfe977a04 in get_expired_dlgs (time=4233810208) at dlg_timer.c:194 #1 0xfe977540 in dlg_timer_routine (ticks=7980, attr=0x0) at dlg_timer.c:210 #2 0x000a839c in timer_ticker (timer_list=0x15ec00) at timer.c:275 #3 0x000a80ec in run_timer_process (tpl=0x1b8088, do_jiffies=1) at timer.c :357 #4 0x000a8668 in start_timer_processes () at timer.c:386 #5 0x00035ea8 in main_loop () at main.c:873 #6 0x000397c4 in main (argc=-4195024, argv=0x150e9c) at main.c:1372 Thanks in advance for any hint you can give me. Best regards. Sergio Gutierrez

3 5

Re: [Serusers] SER + Fax service
by roayu 06 Mar '08

06 Mar '08

Hi Ali, When I do the testing, it seems that in the SER, there are something configured for requiring an authentification for the re-invite. Does SER send an authentificatin required back? Thanks. Best regards, RoaYu _____ From: Ali Zaidi [mailto:aligzaidi@gmail.com] Sent: Thursday, March 06, 2008 2:06 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Fax service Roa, It will work, as long as you use g711, make sure both end use same codec. The reason i said use g711 because fax does not like any compression during transmission, so keep it simple. Ali... On Wed, Mar 5, 2008 at 9:30 PM, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net> > wrote: Good day! I just brought an ATA that support Fax and I would like to know does SER able to handle the fax service as well? My environment is as the following: Fax machine A --> ATA --> SER --> ATA --> Fax machine B Is there any issue to perform the above transaction? Or is there any setting that I need to configure on SER ? Thanks. Best regards, Roa Yu _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> http://lists.iptel.org/mailman/listinfo/serusers <http://lists.iptel.org/mailman/listinfo/serusers>

2 1

[Serusers] separately calls limitation
by tatarskiy＠severen.net 06 Mar '08

06 Mar '08

Нello ! I need of help (it is important for me) Canaanite ! Я использую SER Sip EXpress Router 0.9.6), пока в режиме теста. Он у меня работает как: - Location Server (+ Auth); - Proxy Server; Используется MySQL. Вопрос (question) ! Как мне сделать так, чтобы SER ограничивал колличество вызовов по номеру и/или по имени !? Пример: number(E164) 8121111111 - 5 lines number(E164) 8122222222 - 3 lines number(E164) 8123333333 - 1 line username "Sergey" ------ 2 lines username "Alexey" ------ 9 lines Если можно, то приведите пример ser.cfg Спасибо.

4 7

[OpenSER-Users] OpenSER + OpenXcap for pres-rules: no NOTIFY
by greg.co 06 Mar '08

06 Mar '08

Hi all, I am stuck with a presence authorization problem using openSer 1.3 and openXcap 0.9.9. I have read some similar posts in the mailing list but I couldn't find a solution, so I'll try to explain my scenario to see if it is my mistake or it is a bug. I explain the scenario: User test1 and test2 successfully register and sunscribe to presence.winfo. Both receive a NOTIFY informing that their winfo subscriptions are active. Test2 adds test1 as a contact so he subscribes to test1's presence, then test1 receives a winfo NOTIFY requesting authorization for test2. Test1 does it sending a XCAP document to openXcap the document is correctly stored in openSer xcap table. User test2 receives a presence NOTIFY in pending state, even if user test1 publishes other presence state, the subscription state of test2 remains pending forever. If we look the data base in that moment: +------+--------------------------+------------------+----------------+---------+----------------+----------------+----------+----------------------+---------------+--------------------------------------+------------+-------------+-------------------+--------------+------------+--------+--------+---------+------------------------+------------------------+ | id | presentity_uri | watcher_username | watcher_domain | to_user | to_domain | event | event_id | to_tag | from_tag | callid | local_cseq | remote_cseq | contact | record_route | expires | status | reason | version | socket_info | local_contact | +------+--------------------------+------------------+----------------+---------+----------------+----------------+----------+----------------------+---------------+--------------------------------------+------------+-------------+-------------------+--------------+------------+--------+--------+---------+------------------------+------------------------+ | 1177 | sip:test2@Mydomain | test2 | Mydomain | test2 | Mydomain | presence.winfo | NULL | 10.2670.1209516359.7 | cvvUDerBXK1Qc | b4c2997e-64ac-122b-2780-39a48cb53b8d | 1 | 96206117 | sip:192.168.1.222 | NULL | 1209519959 | 1 | | 1 | udp:192.168.1.210:5060 | sip:192.168.1.210:5060 | +------+--------------------------+------------------+----------------+---------+----------------+----------------+----------+----------------------+---------------+--------------------------------------+------------+-------------+-------------------+--------------+------------+--------+--------+---------+------------------------+------------------------+ WATCHERS +----+--------------------------+------------------+----------------+----------+--------+--------+---------------+ | id | presentity_uri | watcher_username | watcher_domain | event | status | reason | inserted_time | +----+--------------------------+------------------+----------------+----------+--------+--------+---------------+ | 80 | sip:test1@Mydomain | test2 | Mydomain | presence | 2 | NULL | 1209516407 | +----+--------------------------+------------------+----------------+----------+--------+--------+---------------+ XCAP +-----+----------+----------------+-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+----------------------------------+--------+-------------+------+ | id | username | domain | doc | doc_type | etag | source | doc_uri | port | +-----+----------+----------------+-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+----------------------------------+--------+-------------+------+ | 190 | test2 | Mydomain | <?xml version="1.0" encoding="UTF-8"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"><list name="Users"><entry uri="sip:test1@Mydomain"><display-name>test1</display-name></entry></list></resource-lists> | 4 | 8ea82eeb40e00fcf626dbe17d1961ad5 | 0 | buddies.xml | 0 | | 191 | test2 | Mydomain | <?xml version='1.0' encoding='UTF-8'?><ruleset xmlns='urn:ietf:params:xml:ns:common-policy' xmlns:pr='urn:ietf:params:xml:ns:pres-rules'><rule id='whitelist'><conditions><identity></identity></conditions><actions><pr:sub-handling>allow</pr:sub-handling></actions></rule><rule id='politeblocklist'><conditions><identity></identity></conditions><actions><pr:sub-handling>polite-block</pr:sub-handling></actions></rule><rule id='blacklist'><conditions><identity></identity></conditions><actions><pr:sub-handling>block</pr:sub-handling></actions></rule></ruleset> | 2 | 46353b3b55d23d24fd4d9c87a5ff7ba4 | 0 | index.xml | 0 | | 192 | test1 | Mydomain | <?xml version='1.0' encoding='UTF-8'?><resource-lists xmlns='urn:ietf:params:xml:ns:resource-lists'><list name='Users'></list></resource-lists> | 4 | f854971d48ed1a8913e20b3409a6b723 | 0 | buddies.xml | 0 | | 193 | test1 | Mydomain | <?xml version="1.0" encoding="UTF-8"?> <ruleset xmlns="urn:ietf:params:xml:ns:common-policy" xmlns:pr="urn:ietf:params:xml:ns:pres-rules"><rule id="whitelist"><conditions><identity><one id="sip:test2@Mydomain"/></identity></conditions><actions><pr:sub-handling>allow</pr:sub-handling></actions></rule><rule id="politeblocklist"><conditions/><actions><pr:sub-handling>polite-block</pr:sub-handling></actions></rule><rule id="blacklist"><conditions/><actions><pr:sub-handling>block</pr:sub-handling></actions></rule></ruleset> | 2 | 8afbf7f0464de1273303bbea554404ff | 0 | index.xml | 0 | +-----+----------+----------------+-----------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------+----------+----------------------------------+--------+-------------+------+ We can see in 'WATCHERS' table that test2 has a pending presence subscription but as can be seen in 'XCAP' table user test1 has authorized test2 to see his presence. My thinking is that once test1 changes his authorization rule, the presence subscription should chage to active, am I right? My conclusion is that something between the presence rules authorization and the subscriptions is not working fine, or I did something wrong and I haven't noticed it. Please, I really need help with that as I have been stuck here for weeks. Thanks in advance!! Gregorio -- View this message in context: http://www.nabble.com/OpenSER-%2B-OpenXcap-for-pres-rules%3A-no-NOTIFY-tp15… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[OpenSER-Users] issue - remote sip client must re-register after failover in Openser Registrar HA config
by Naija Man 06 Mar '08

06 Mar '08

Hi friends, I am having problems setting up seamless openser registrar failover. I am trying to setup OpenSER registrar HA using 2 openser registrars v1.3.0behind a Netscaler citrix Load balancer and I cant get the failover scenario to work without re-registration of the remote sip client. I am desperately looking for tips to make this configuration work or any other ideas to achieve SIP registrar HA using OpenSER. Thanks. The setup is configured as active-passive and the 2nd OpenSER becomes active only after a Openser_1 fails and the LB fails-over to the 2nd. The 2 openser servers are SIP registrars for my voice network. REGISTRATION: remote SIP UA ---> remote firewall ---> internet ---> my firewall ---> citrix NA loadbalancer (VIP) ----> openSER_1 \---> openSER_2 1. Remote SIP client configures the LB VIP as the SIP proxy and registers 2. the citrix LB gets the registrations and forwards to the active openser (say, openser_1) 3. openser_1 stores the contact as MIP:port in the mysql database 4. openser_1 replicates the registration to openser_2 (t_replicate) OUT-GOING CALL - before fail-over remote SIP UA <--- remote firewall <--- internet <--- my firewall <--- citrix NA loadbalancer (VIP) <----> openSER_1 ^ openSER_2 | Asterisk 1. call (INVITE) is generated from asterisk and sent to LB VIP 2. the citrix LB gets the INVITE and forwards to the active openser (say, openser_1) 3. openser_1 looks-up the URI in the MySQL database, finds the MIP:port that the registration was received from the LB and forwards the INVITE to that destination. 4. the LB receives the INVITE and forwards it to the remote SIP client OUT-GOING CALL - after fail-over remote SIP UA ---x remote firewall ---x internet ---x my firewall ---x citrix NA loadbalancer (VIP) ----x openSER_1 ^ \-----< openSER_2 | Asterisk 1. call (INVITE) is generated from asterisk and sent to LB VIP 2. the citrix LB gets the INVITE and forwards to the openser now active (openser_2) 3. openser_2 looks-up the URI in the MySQL database, finds the same MIP:port of the LB that was replicated from openser_1 and forwards the INVITE to that destination. 4. the LB receives the INVITE and drops the packet without forwarding it out to the remote SIP client For the outgoing call to work: the remote sip client either has to re-register, so the registration gets passed to openser_2. OR, openser_1 is made active again. Configuration of OpenSER: REGISTRATION: # If this is a registration from the public UA, # then we store it fix_nated_contact(); # use ip address of the packet instead of the sip message force_rport(); # append rport. might not be necessary # save Registration in location table save("location"); xlog("L_DBG", "SipMsgId[$mi] Save Registration: To[$tu] Contact[$ct]\n"); append_hf("CtlRegFwd: \r\n"); # replicate the register message to the backup registrar # if I am the backup, ignore the message if(!t_replicate("sip:openser_2:5060")) { xlog("L_WARN", "SipMsgId[$mi] Fail to replicate Contact[$ct] to fail over registrar\n"); OUTGOING CALL: # Do lookup so it gets to the correct NATed destination if (!lookup("location")) { # this user has not REGISTERed or registration has expired xlog("L_INFO", "SipMsgId[$mi] Send 404-Not Found (Msg rejected because user not found)\n"); sl_send_reply("404", "Not Found"); exit; }; # keep track of reply t_on_reply("11"); if (!t_relay()) { xlog("L_DBG", "SipMsgId[$mi] Failed t_relay. Do sl_reply_error.\n"); sl_reply_error(); }; Thank you for your anticipated help. Regards, Buki A

2 1

[Serusers] e2e cancel - ghost calls
by Nuno Ribeiro 06 Mar '08

06 Mar '08

Hi all, I'm having a problem related with the "early" CANCEL's. The scenario is the following one : A PSTN call to a SIP phone. When the PSTN decides to cancel the call only a right after initiating, the PSTN will send the CANCEL message to the SER but this is discarded and not forwarded to the correct path to the SIP Phone. So what happens is that we have a ghost call.... The PSTN has already cancelled the call but the SIP phone continues to ring. The code that I have in the SER script is really simple: if(subst_uri('/^sip:(\+[0-9]+)@ 192.168.20.69.*user=phone$/sip:\1@ptinovacao.pt/i')){ record_route(); loose_route(); t_relay_to_udp("192.168.20.5", "5060"); break; } In the log file I see that: RFC3261 transaction matching failed t_lookup_request: no transaction found e2e_cancel: e2e cancel proceeding Could you help me? Thanks in advance. Best Regards, -- Nuno Ribeiro

2 2

[OpenSER-Users] WeSIP ans Openser configurations problems
by bsb093 06 Mar '08

06 Mar '08

Hi, I am new in this area and as expected I am having real problem with setting up the Openser with WeSIP . I have installed OpenSER 1.2.0 on debian platform and download WeSIP from WeSIP website. When trying to run Openser I get an error that fifo="tmp/openser_fifo" line has parse error (in the openser.cfg). Therefore I remove this line and instead I added "modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")". Why do I get the error message if that is the same in the examples? Second, "modparam("seas", "listen_sockets", "192.168.2.100:5080");" has parse error. I can't see why the IP is correct and if I change it to localhost/127.0.0.1 I get the same error. What can be the problem in this line? Everywere I checked it is just the same. Third, when trying to run WeSIP with WeSIP/bin/startup.sh I get the following Java exceptions: -Dlog4j.configuration=file:/usr/local/src/wesip/conf/log4j.properties org.apache.catalina.startup.Bootstrap -config /usr/local/src/wesip/conf/server.xml start Catalina.start using /usr/local/src/wesip/conf/server.xml: java.lang.IllegalArgumentException: This container is not a StandardHost java.lang.IllegalArgumentException: This container is not a StandardHost at org.apache.commons.digester.Digester.createSAXException(Digester.java:2792) at org.apache.commons.digester.Digester.createSAXException(Digester.java:2818) at org.apache.commons.digester.Digester.endElement(Digester.java:1070) at com.sun.org.apache.xerces.internal.parsers.AbstractSAXParser.endElement(AbstractSAXParser.java:633) at com.sun.org.apache.xerces.internal.parsers.AbstractXMLDocumentParser.emptyElement(AbstractXMLDocumentParser.java:221) at com.sun.org.apache.xerces.internal.impl.XMLDocumentFragmentScannerImpl.scanStartElement(XMLDocumentFragmentScannerImpl.java:872) at com.sun.org.apache.xerces.internal.impl.XMLDocumentFragmentScannerImpl$FragmentContentDispatcher.dispatch(XMLDocumentFragmentScannerImpl.java:1693) at com.sun.org.apache.xerces.internal.impl.XMLDocumentFragmentScannerImpl.scanDocument(XMLDocumentFragmentScannerImpl.java:368) at com.sun.org.apache.xerces.internal.parsers.XML11Configuration.parse(XML11Configuration.java:834) at com.sun.org.apache.xerces.internal.parsers.XML11Configuration.parse(XML11Configuration.java:764) at com.sun.org.apache.xerces.internal.parsers.XMLParser.parse(XMLParser.java:148) at com.sun.org.apache.xerces.internal.parsers.AbstractSAXParser.parse(AbstractSAXParser.java:1242) at org.apache.commons.digester.Digester.parse(Digester.java:1572) at org.apache.catalina.startup.Catalina.start(Catalina.java:428) at org.apache.catalina.startup.Catalina.execute(Catalina.java:358) at org.apache.catalina.startup.Catalina.process(Catalina.java:129) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:585) at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:156) Please advise. Thanks in advance Shine -- View this message in context: http://www.nabble.com/WeSIP-ans-Openser-configurations-problems-tp15826527p… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Problem in Installing OpenSER-1.3.x
by srinivas.banda＠wipro.com 06 Mar '08

06 Mar '08

Hi all, I am new to OpenSER, I want to install OpenSER on my Redhat Linux machine. I had downloaded OpenSER latest1.3.x. I have some doubts regarding installation of openser 1. is mysql is to be installed before installing openser?? 2. I had installed openser1.3.x with installation procedure given in the openser.org, ( I did make all) 3. if it is mandatory to have mysql running before installing openser1.3 please tell where can I get mysql compatible for redhat linux?? 4. if any body have complete step by step installation procedure and what are all pre required software is to be installed plz give me suggestion. Please give suggestions to my queries as soon as possible because it is very urgent.. Thanks in advance Regards Srinivas The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Serusers] SER + Fax service
by roayu 06 Mar '08

06 Mar '08

Good day! I just brought an ATA that support Fax and I would like to know does SER able to handle the fax service as well? My environment is as the following: Fax machine A --> ATA --> SER --> ATA --> Fax machine B Is there any issue to perform the above transaction? Or is there any setting that I need to configure on SER ? Thanks. Best regards, Roa Yu

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Re: [Serusers] Variables in the Log command
by Jiri Kuthan 05 Mar '08

05 Mar '08

use xlog module for formattable logging. -jiri At 21:44 05/03/2008, Davis, Al wrote: >Content-Language: en-US >Content-Type: multipart/alternative; > boundary="_000_8F86FF097D265D4EB4D9D2F29DB00AA83F9797253DEXVMBX0152exc_" > > >Is there a way to output variables such as SIP_SRCIP and SIP_RURI to /var/log/messages using the log command? >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] Variables in the Log command
by Davis, Al 05 Mar '08

05 Mar '08

Is there a way to output variables such as SIP_SRCIP and SIP_RURI to /var/log/messages using the log command?

1 0

Re: [Serusers] [Sems] Poll on dropping unix socket interface ("unixsockctrl")
by Stefan Sayer 05 Mar '08

05 Mar '08

Hello, Raphael Coeffic wrote: > Hi Srinivas, > > this is a very good question you are bringing! I forgot to describe these > aspects. So i will try to describe the experience we gathered in the last > weeks. > > binrpcctrl: > - increases signaling performance (100% according to Stefan's measurements) this is not correct, its much more: http://lists.iptel.org/pipermail/semsdev/2008-January/002015.html The relation is about 750 cps vs 80 cps, which is ~ 9 times performance increase. > - increases flexibility (all the informations passed between SER and SEMS are > configurable). > > sipctrl: > - increases signaling performance (80% against binrpcctrl according to > Stefan's measurements) I don't know where this figure comes from, it is as well much more (I said this: http://lists.iptel.org/pipermail/semsdev/2008-February/002221.html , 1000 cps on lo on a laptop, versus a quad xeon over lan). > - reduces configuration and run-time complexity > > Concerning the stability, it is clear that the next weeks as well as the > community will show us what those new control plug-in are worth. From my > knowledge i would say that we are close to completion for both new plug-ins. you can help by trying out your application with the new plugins and report any possible issues. > > Dropping the support does not mean that we will remove it from SVN, that just > means that future version should not use it. Existing installations are not > concerned by those measures. i.e. i would say it should be part of 1.0 release, but marked as deprecated. After that release I would move for removing it, in favor of more flexibility that new control plugins provide. Stefan > > Cheers > Raphael. > > On Wednesday 05 March 2008 15:08, Srinivas Kotamarti wrote: >> Hi Raphael, >> What are the advantages of binrpcctrl and sipctrl over the unixsockctrl? >> >> Has enough testing done on them? We are currently using SEMS with SER0.9.6 >> (with unixsockctrl) and it is working OK for us. IF we move to the >> binrpcctrl or sipctrl would we get a better performance? >> >> Thanks >> >> --Srinivas >> >> -----Original Message----- >> From: sems-bounces(a)lists.iptel.org [mailto:sems-bounces@lists.iptel.org] On >> Behalf Of Raphael Coeffic Sent: Wednesday, March 05, 2008 7:00 AM >> To: sems(a)lists.iptel.org >> Cc: serusers(a)lists.iptel.org >> Subject: [Sems] Poll on dropping unix socket interface ("unixsockctrl") >> >> >> Hi *! >> >> with the emergence of the binary RPC interface for SER2 ("binrcpctrl" >> plug-in), and SEMS' own sip stack ("sipctrl" plug-in), Stefan and I believe >> that the unix socket control interface is defeated. >> >> That's why we would like to know who is in favor of dropping support for >> the unix socket interface. >> >> Cheers and happy polling! >> >> Stefan & Raphael. >> _______________________________________________ >> Sems mailing list >> Sems(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/sems > -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

2 1

[OpenSER-Users] Openser HA
by Cosimo Fadda 05 Mar '08

05 Mar '08

Hi, I want to setup two openser in a HA enviroment; So, correct me if I'm wrong, I need three IP addresses, one for each box and a shared one, My question is, I have to set the advertised_address= <shared-IP> in both openser servers? Thanks in advance, Cosimo Fadda *Ing. Cosimo Fadda System Engineer Klarya s.r.l.* ____

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Re: [Serusers] [Sems] Poll on dropping unix socket interface ("unixsockctrl")
by Srinivas Kotamarti 05 Mar '08

05 Mar '08

Hi Raphael, What are the advantages of binrpcctrl and sipctrl over the unixsockctrl? Has enough testing done on them? We are currently using SEMS with SER0.9.6 (with unixsockctrl) and it is working OK for us. IF we move to the binrpcctrl or sipctrl would we get a better performance? Thanks --Srinivas -----Original Message----- From: sems-bounces(a)lists.iptel.org [mailto:sems-bounces@lists.iptel.org] On Behalf Of Raphael Coeffic Sent: Wednesday, March 05, 2008 7:00 AM To: sems(a)lists.iptel.org Cc: serusers(a)lists.iptel.org Subject: [Sems] Poll on dropping unix socket interface ("unixsockctrl") Hi *! with the emergence of the binary RPC interface for SER2 ("binrcpctrl" plug-in), and SEMS' own sip stack ("sipctrl" plug-in), Stefan and I believe that the unix socket control interface is defeated. That's why we would like to know who is in favor of dropping support for the unix socket interface. Cheers and happy polling! Stefan & Raphael. _______________________________________________ Sems mailing list Sems(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/sems

1 0

[OpenSER-Users] Why every body return "404 Not Found" instead of "480 Not Available"?
by Iñaki Baz Castillo 05 Mar '08

05 Mar '08

Hi, after a large discussion in sip-implementros maillist I realize that a proxy/location server shouldn't reply "404" if a callee DOES exist but it's not registered in that moment. In fact: 480 Temporarily Unavailable [...] This status is also returned by a redirect or proxy server that recognizes the user identified by the Request-URI, but does not currently have a valid forwarding location for that user. 404 Not Found The server has definitive information that the user does not exist at the domain specified in the Request-URI. [...] Why then every body suggest to reply "404 Not Found" when "lookup(location)" returns -1? For example, the "registrar" module doc suggests it: Example 1-22. lookup usage lookup("location"); switch ($?) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; Best regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] natural_event
by Pezhman Lali 05 Mar '08

05 Mar '08

Hi Dear I have a problem, with my ser 0.9.6 server, as a sip server. the following results are with grandstream 487 or 287. about 30% of customers, can install the sip phone very easy with the first configuration. about 25%, can register after changing adsl router, or sip phone's configuration, about 15%, can register with another sip phones, like linksys, or. ... and about 30% can not using sip services. is it a natural event???? do u have any idea ???? best Pezhman ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[Serusers] Installing serweb
by Adeyeye O. Michael 05 Mar '08

05 Mar '08

Hi, With the recent development in PEAR Network, where XML_RPC isn't available any longer, has anyone been able to install serweb recently? I can't get XML_RPC package which is one of the requirements to get SERWEB running. Where I was able to get the raw files from PEAR but not ABLE to run "pear install XML_RPC", i can't make install and the likes. Hence, i doubt if XML_RPC will be successfully installed into my PEAR. Any help!! Mike.

2 1

[Serusers] meet at IETF and/or VoN
by Jiri Kuthan 05 Mar '08

05 Mar '08

FYI: I'm at the IETF meeting in the week of March/10th and at VoN in the week of March/17th. Drop me a note if you would like to meet whether for some SER/SIP related discussion or just a beer chat :-) -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] AVP/SELECT type / length in ser.cfg?
by Mike Trest - Personal 04 Mar '08

04 Mar '08

Hi, How can I reference the type and length attributes of a variable ( @set or $avp) ? ? ? ..mike..

4 5

[OpenSER-Users] Is possible to convert 183 in 180 in "on_reply_route"?
by Iñaki Baz Castillo 04 Mar '08

04 Mar '08

Hi, in order to avoid replying with early_media I'm trying to transform a "183" with SDP into a "180". Ok, my conclusion is that I cannot do it. Textops functions seem to work just with SIP headers and body but never can modify the SIP reply code, is true? Also, I would need to delete the entire body in the 180|183 reply but functions supposed to do it don't work as expected. For example: * subst_body('/^.*//g') does NOTHING, the only way it can do somehting is by: subst_body('/^.*=.*//g') any reason for this? Thanks for any suggestion. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

4 4

[Serusers] RTCP generation
by Keyur Parikh 04 Mar '08

04 Mar '08

Hi, I am working on project involving SER, SEMS. Things works OK as configured. I am curious to know how I can generate RTCP message towards caller. Greatly appreciate the response. Thanks, Keyur Parikh

1 0

[Serusers] How to enable RTCP reports
by Srinivas Kotamarti 04 Mar '08

04 Mar '08

Hi, I am working on a project that uses SER, SEMS and Media Proxy. I was wondering how and where to enable the RTCP reports. I could not find any useful information either in the product documentation or in the user groups. Does any of these products support generating RTCP reports? IF not how do we add the feature and where would be best place to add it? Thanks for the help. --Srinivas Kotamarti

1 0

[OpenSER-Users] OpenSER and Security - how?!
by Max Bowsher 04 Mar '08

04 Mar '08

I've been looking at the possibility of using OpenSER as an ingress/egress gateway, mediating access between the internet at large, and a private network containing amongst other things SIP servers through which a call may be routed to provide services such as IVR and call archiving, but which should otherwise be hidden from the outside world. I'm finding two interlinked problems: (1) The internal layout of the network is revealed in Via headers - OK, so this is somewhat intrinsic in SIP, and not really OpenSER's fault, but.... (2) ... If an inbound SIP request has Route headers, loose_route() pretty much sends it whereever the requester asks. There are admonitions in the OpenSER docs about the need to secure loose_route(), but there's no information I can find on how you should do this. In particular, a simple authorization scheme is not good enough - just because someone should be allowed to place calls through the gateway, doesn't mean it should be allowed absolute control over the routing of the request, or they could use information gleaned from Via headers of previous transactions to add or bypass routing steps within the private network at will. It is possible to securely use OpenSER on a security boundary? If so, how? Max.

2 1

[OpenSER-Users] Last occurrence of some thing...
by Daniel Cardoso Alves 04 Mar '08

04 Mar '08

Hi people... With the Textops Module is possible search for the first match of re in the body of the message and appends txt after it, ex: search_append, search_append_body... But I want append a TXT after the last occurrence of some thing... Example: after the last occurrence of .*a=rtpmap:.* Is there some way for my doubt? --------------------------------- Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!

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[OpenSER-Users] fail to authenticate INVITE messages
by antalsia＠free.fr 04 Mar '08

04 Mar '08

Hi all, I'm trying to authenticate INVITE messages with OpenSER 1.3.0 (SIP Proxy + RTPPROXY) and LDAP module. Although REGISTER authentication works well, I cannot check user authorization for INVITE messages. I'm using an openser.cfg configuration template from sipwise (see an extract below). pv_proxy_authorize() routine always returns false, even if my user is registered and authenticated through our OpenLDAP. Can you tell me what I'm doing wrong ? Many thks, --------------------------- openser.cfg --------------------------- route { ... if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } ... } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(is_present_hf("Authorization")||is_present_hf("Proxy-Authorization")) { xlog("L_INFO", "is_present_hf Authorization or Proxy-Authorization\n"); if (!ldap_search("ldap://blabla/blabla?uid,userPassword?sub?(uid=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found xlog("L_INFO", "Ldap user not found\n"); sl_send_reply("404", "User Not Found"); exit; case -2: # internal error xlog("L_INFO", "Internal server error during authentication\n"); sl_send_reply("500", "Internal server error"); exit; default: exit; } } ldap_result("uid/$avp(s:username)"); ldap_result("userPassword/$avp(s:password)"); } if(!pv_www_authorize("")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("mydomain", "1"); 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"); 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(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (!pv_proxy_authorize("")) { xlog("L_INFO", "TESTING : NOT AUTHENTICATED !!!!\n"); } else { xlog("L_INFO", "TESTING : AUTHENTICATED !!!!\n"); } route(3); exit; } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } --------------------- DEBUG LOGS --------------------- eb 29 15:33:14 proxy_sip /usr/sbin/openser[5444]: DBG:core:parse_headers: flags=80 Feb 29 15:33:14 proxy_sip /usr/sbin/openser[5444]: DBG:core:parse_headers: flags=80 Feb 29 15:33:14 proxy_sip /usr/sbin/openser[5444]: Call to foreign domain - M=INVITE RURI=sip:beacon@columbia.edu F=sip:test@xxxxxxxx T=sip:beacon@co lumbia.edu IP=xxxxxxx ID=282a363f-5be5-dc11-8272-0015c56ccfaa@xxxxxx Feb 29 15:33:14 proxy_sip /usr/sbin/openser[5444]: DBG:core:parse_headers: flags=10000 Feb 29 15:33:14 proxy_sip /usr/sbin/openser[5444]: DBG:auth:pre_auth: credentials with given realm not found Feb 29 15:33:14 proxy_sip /usr/sbin/openser[5444]: TESTING : NOT AUTHENTICATED

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[OpenSER-Users] Openser sending BYE to itself.
by Daniel Cardoso Alves 04 Mar '08

04 Mar '08

Hi people... I am using the module dialog to finish a call. http://www.openser.org/docs/modules/1.3 ... tml#AEN461 It is working... I need that the openser send a BYE to CALLE, to CALLER and to itself... The BYE to endpoints(CALLE and CALLER) is ok, but it not sending BYE to itself... Somebody can help? Thanks in advance... --------------------------------- Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!

2 4

[Serusers] SIP Express Bundle. Step 1.
by SIP 03 Mar '08

03 Mar '08

Okay, It's time to get things going on the SIP Express Bundle. I'm stepping in to take the reins as project manager on this one so we can get things rolling. First please take a look at the project page: http://www.iptel.org/bundle/project Step 1 here, I'd like to verify a headcount of people working on this project as of now. This doesn't mean people can't join up later, and it certainly doesn't mean that I'm going to expect you to put more time into this than you can commit, but we need to know where we stand. Currently, on the project list, we have: * Jai: testing and installation work * ram: testing * SIP: testing (and now project management) * Jiri: debian, oob * Mike: testing and documentation * Stefan: SEMS Are the people listed here still available to do these particular tasks? Has anyone else volunteered time/resources on this project that hasn't been listed here? If so, shoot me an email, and I'll add you to the list. We need a people inventory so we know what roles we're missing. For instance, I've heard a few people mention that they'd be willing to donate a server to the cause so we have somewhere to test things upon. I don't have those people listed, but that's certainly an area we're going to need. Also, we've had varying reports of which direction we're going -- ISO or Virtual Machine. For now, I think we should concentrate on an ISO as that's going to be the most attractive to people for building a ready-to-go hardware appliance, and it will probably be a little less of a headache for creating a drop-in installation. However, I also think a virtual machine should be on the list in the future, and once we get an ISO ready, creating a virtual machine from that configuration should be relatively straightforward. However, on the list of people, I don't see anyone down for scripting an install -- a kickstart disc or what have you. That's another area we're lacking. If anyone has experience in that area, shoot me an email and I'll add you to the list. We need to focus. For this reason, we'll start small and work from there. Do NOT expect this first SER bundle to have every feature you could imagine in it, and don't get discouraged if your favourite feature is missing. As the project grows, we'll most certainly be adding capabilities and allowing a tailored install. But for now, the goal should be to get things running and working first in a simple format and then add on from there. Having a bundle like this is perhaps the biggest step we can take toward bringing SER to the masses and finally breaking down some of those barriers that have kept people out of the project. I'm looking forward to working with all of you on this project! Neil

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[OpenSER-Users] "append_branch()" in "failure_route" after modify SDP
by Iñaki Baz Castillo 03 Mar '08

03 Mar '08

Hi, imagine OpenSer changes the SDP ("use_media_proxy()") and the call is rejected by the callee. Then a "faiure_route" is runned in which we want to send the INVITE to a voicemail server. When we do the "append_branch()" the SDP is not the original but the previously modified. Is it the expected behaviour? what about if I want to use the original SDP for the second branch? (imagine my voicemail server has comedia support for example). -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 2

[OpenSER-Users] TLS prob
by Ali Jawad 03 Mar '08

03 Mar '08

Hi all I got tls working fine..I can login without problems..I have udp working fine too...however when I enable TLS and I try to make a call through my PSTN gw I get a timout error..however this does not happen with udp.. Any hints please ? Thx

2 2

[OpenSER-Users] SMS module
by Jean Boite 03 Mar '08

03 Mar '08

Hello, I am using openser-1.2.2-notls. I want to send a sms thanks to a IM (I am using Eyebeam, but i don't know if it's possible to make that with this software). Thats why, i need your help to explain me how send a sms thanks to the sms module in openser. I put in my openser.cfg, following "if(!method==MESSAGE)" the function which is on the documentation : if (sms_send_msg_to_net("Orange")) { if (!t_reply("202", "yes sir, SMS sent over")) { # if replying failed, retry statelessly sl_reply_error(); }; } else { if (!t_reply("502", "Bad gateway - SMS error")) { # if replying failed, retry statelessly sl_reply_error(); }; exit; }; Openser is running good, my problem is thats I don't know how to formulate my message in Eyebeam to send correctly my sms. Thanks.

3 6

[OpenSER-Users] Why $rp (RURI port) is no writeable???
by Iñaki Baz Castillo 03 Mar '08

03 Mar '08

If I can do: $rU = ... $rd = ... why can't I do: $rp = ... ? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[OpenSER-Users] No RTCP generation
by Keyur Parikh 03 Mar '08

03 Mar '08

Hi, I am working on project involving SER, SEMS and MediaProxy. Things works OK as configured. I am curious to know how I can generate RTCP message towards caller if the called UA is not generating one. I appreciate your response. Thanks, Keyur Parikh

2 1

[OpenSER-Users] OpenSER prepaid CDRTool 6.2.3 problem with replacing '+' with '00'
by Marc LEURENT 03 Mar '08

03 Mar '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I have trouble with number dialed starting with + CDRTool doesn' understand it. So I have to strip it from the request and replace it by 00! The problem is that even using this at the begenning of OpenSER script, it's not working (see CDRTool log below) Any idea how to solve it? Thanks # normalization to e164 if($ruri.user =~ "^\+[1-9][0-9]+") { xlog("STRIPING + For e164 NORMALIZATION \r\n"); strip(1); prefix("00"); subst('/^To:(.*)sip:\+(.*)$/To:\1sip:00\2/ig'); }; Mar 3 12:24:49 krusty CDRTool[15025]: MaxSessionTime From=mleurent(a)sip.wifirst.net To=sip:+33677832975@sip.wifirst.net Dura tion=10800 Lock=1 Mar 3 12:24:49 krusty CDRTool[15025]: Load last minute: 1 calls @ 0.00 cps, memory: 16.77 MB, uptime: 0.6 hours Mar 3 12:24:49 krusty CDRTool[15025]: Call to +33677832975(a)sip.wifirst.net, no limit imposed Mar 3 12:24:49 krusty CDRTool[15025]: GetBalance From=mleurent(a)sip.wifirst.net Mar 3 12:24:51 krusty /usr/sbin/openser[18637]: ACC: call missed: timestamp=1204543491;method=INVITE;from_tag=xirvq;to_tag= ;call_id=cpvrqboancqnuii(a)192.168.95.139;code=487;reason=Request Terminated Mar 3 12:24:51 krusty CDRTool[15025]: DebitBalance From=mleurent(a)sip.wifirst.net To=sip:0033677832975@sip.wifirst.net Durat ion=0 OpenSER config extract: # ATTENTION: DO NOT PUT ; at the end of the radius_extra attribute modparam("acc", "radius_extra", " Called-Station-Id=$tu; Calling-Station-Id=$fu; Canonical-URI=$avp(to_destination); User-Name=$avp(user_name); Billing-Party=$avp(billing_party); Sip-User-Realm=$ar; SIP-Proxy-IP=$Ri; Source-IP=$si; Source-Port=$sp; Acct-Authentic=$avp(s:authentic); From-Header=$hdr(from); User-Agent=$hdr(user-agent); Contact=$hdr(contact); Event=$hdr(event)") # Set the acc flags xlog("JUSTE BEFORE SETTING FLAGS FOR RADIUS\r\n"); if(is_method("INVITE") && !has_totag()) { xlog("L_INFO", "I AM SETTING THE FLAGS FOR RADIUS \r\n"); # $avp(s:authentic) = 1; # $avp(can_uri) = $ru; # SIP Request's URI $avp(user_name) = $au + "@" + $ar; $avp(billing_party) = $au + "@" + $ar; $avp(to_destination) = $tu; xlog("SETTING FLAGS 1 FOR RADIUS \r\n"); setflag(1); # radius_flag xlog("SETTING FLAGS 2 FOR RADIUS \r\n"); setflag(2); # radius_missed_flag }; -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHy+T8rxOjjFYWQtoRAjUzAJ9kUYZpNUEEcKO7aqRuD9Uob0myuQCdEUyU 67e6YOc9jVgzTPQk+A6ccec= =ZGdn -----END PGP SIGNATURE-----

3 7

[OpenSER-Users] Error starting Openser 1.3 with postgres
by Mauricio Viveros 03 Mar '08

03 Mar '08

Hello My name is Mauricio Viveros, I'm a Colombian programmer. I'm implementing VoIP system in a little company, so I'm installed OPENSER with POSTGRES module but when I try to run openser I get dis error message: Feb 27 01:05:53 [7788] DBG:core:find_cmd_export_t: found <sl_reply_error>(0) in module sl [/usr/local/lib/openser/modules/] Feb 27 01:05:53 [7788] CRITICAL:core:yyerror: parse error in config file, line 376, column 2-6: syntax error Feb 27 01:05:53 [7788] CRITICAL:core:yyerror: parse error in config file, line 376, column 2-6: Feb 27 01:05:53 [7788] ERROR:core:main: bad config file (2 errors) to see all trace, see the attached file openser_error_message.txt I attached too the config files: openser.cfg y openserctlrc I hope that you can help me with this problem. Thank you very much. Mauricio Viveros Programmer Cali-Colombia

4 6

[OpenSER-Users] Configuring the openSER with the DISC gives problem
by dilip.m＠renovau.com 03 Mar '08

03 Mar '08

hello friends, I start openSER there are no errors (i.e. the auth_diameter module loads), but the DISC client complains that the peer is unknown. Both SER and the DISC client are running on the same machine, so I added aaa://localhost as a peer, but that does not seem to help. Openser can not communicate with diameter client. Regards, Dilip

3 3

[OpenSER-Users] openser+opendiameter issue---->kindly do reply
by bhargav.b＠renovau.com 03 Mar '08

03 Mar '08

hi all, if anybody have got the information regarding authenticating a user with the help of openser and opendiameter ,pleaase help me in doing that.it seems that i am really spoiling a lot of days in doing the same by my own. i have configured openser and opendiameter correctly and even they are communicating with each other.i have used the auth_diameter module of openser.but it seems there is some problem.on the client side of the opendiameter,i am getting a messege like"invalid session id avp" when i do a call through x-lite softphone to openser. seeking ur responces regards bhargav.b

2 1

Re: [Serusers] SIP Express Bundle. Step 1.
by Mike Trest - Personal 02 Mar '08

02 Mar '08

Hi, This is to summarize my opinions about FC* distro use. IMHO, I think FC* is best selection as it contains many more fixes than does the older CENTOS (based on 5). I have deployed several hundred FC* boxes in VoIP applications. This is over 10,000 active ports without "Enterprise" stability issues. IMHO this project needs the quickest path to the Enterprise community regardless of the OS/distro used. I suppose the ultimate question is who is our target? Ourselves, naturally. However, I suggest our target is not the bankers or major corporations with lots of rules and procedures. That group will never adopt SER until they have a commercial-grade support system to advise their IT folks what to do for every question they may have. IMHO our initial target is those early adopters who are trying to create new businesses in telecomm or consulting-on-telecom. We want them to have a solid core that they can leverage into their new appliances and specialized applications. The early adopters are risk-takers (This means us as well!) They demand an open box in which they can face the SIP world with some assurance of standards compliance while at the same time they can face their clients with something better, faster, cheaper, and innovative enough to get paid well for their efforts. Making a technology "buy - in" decision at any point in time is only a check point - not a final resting place. IMHO, we are better off selecting an OS/distro effort that has a large share of both early adopters and long term commercial support - - - so long as it meets our current and future technical **AND** target market requirements. Research confirms that the RH/FC community is the largest community with name recognition and respect among both the "geek-innovator" community as well as the Enterprise community. ..mike..

7 14

[OpenSER-Users] Radius accounting
by Henri Keski-Sikkila 02 Mar '08

02 Mar '08

Hello, When radius accounting is in use and radius_extra is used like this: modparam("acc","radius_extra","Sip-Source-IP-Address=$si;Sip-Source-Port=$sp") and when freeradius gets an accounting-request packet, it includes: Sip-Source-IP-Address = 46.50.57.49 Sip-Source-Port = 808857653 Is this a known issue? BR, Henri Keski-Sikkila

3 5

[OpenSER-Users] Routing calls from PSTN
by Andy Smith 02 Mar '08

02 Mar '08

Hi list, I'd like to know, from others experience, what solutions might be best for configuring OpenSER routing from a PSTN gateway to multiple IP PBXs. I guess there may be various solutions ranging from quick and dirty to the other extreme but Id be interested to get some ideas what are typical solutions for others. As OpenSER is so configurable and there are so many optional modules its quite hard, coming from zero exposure to any production SIP envrionments, to know how best to go about this... thanks Andy.

2 1

[OpenSER-Users] Does t_realy chooseTCP/UDP protocol if it's necessary?
by Iñaki Baz Castillo 02 Mar '08

02 Mar '08

Hi, very strange: - I call to OpenSer using a TCP client. - The INVITE arrives via TCP to OpenSer. - I do: $rd=ASTERISK_IP; t_relay(); exit; - Asterisk only speaks UDP of course. - And OpenSer sends the INVITE to Asterisk by UDP. WHY does it work??? I hoped it would fail trying to forward the INVITE by TCP to Asterisk since I didn't force the outgoing socket, just the two lines above. Thanks for any explanation. -- Iñaki Baz Castillo

2 3

[OpenSER-Users] Setting smaller register "expires" value for TCP
by Iñaki Baz Castillo 01 Mar '08

01 Mar '08

Hi, in order to mantain TCP conecction with UACs behind NAT the only working solution (AFAIK) is setting register expires value smaller than the time UAC mantains open the TCP connection (value "x"). Certainly, I don't know which is the typical "x" value (i.e. Twinkle 1.2 that suports TCP mantains TCP connection for 32 seconds). Is 30 a good value? Of course, in UDP I use "OPTIONS" keepalive from OpenSer, so I don't need (and don't want) a very small register "expires" value. I just want to set "expires" value to 30 seconds in case of TCP. But "registrar" module seems not to allow me it, I just see: default_expires (integer) min_expires (integer) max_expires (integer) It would be nice if I could set an AVP to set the max expires time in the script, and set itto 30 if "proto==TCP". Is it possible in some way? Thanks a lot for any suggestion. Regards. -- Iñaki Baz Castillo

2 2

[OpenSER-Users] LCR and Load Balancing, Routing
by Knud Müller 01 Mar '08

01 Mar '08

Hi, I'm pretty familiar with the asterisk, but have only little understanding of SER/OpenSER. I read through some cookbooks and found some things that might help, but I just want to ensure that I'm on the right track before investing a lot of time into it. I have an asterisk running that terminates currently against one (SIP) carrier. We are noticing that the carrier has an increasing rate of short outtages and want a more redundant solution. Currently we have two more carriers and have to switch manually which was fine for the testing period. Now I'd like to route automatically. The routing decision should be based on a least cost basis, for each number the cheapest route shall be used. But if a route is not working properly, mostly these are 5XX Messages, the second cheapest route shall be used. Can I do this failover aspect with OpenSER/LCR too? Is the LCR Module the right point to start, as I have not seen failover strategies? There are some questions left which are more detailed - if a carrier shows to fail often, I just want a few calls routed to that carrier, to determine if its working again. Is there any chance to include network/packet information into the decision too, like roundtrip times, jitter etc? Hopefully someone can help me find the best solution... regards, Knud

2 1

[OpenSER-Users] X-ten failed to make connection
by Michael Chan 29 Feb '08

29 Feb '08

Derek Please see the attached mailing address and send them the infor for help. MC

1 0

[OpenSER-Users] Why $rP is not writeable?
by Iñaki Baz Castillo 29 Feb '08

29 Feb '08

Hi, sometimes I need to forward a message to a SIP UAS that just speaks UDP (for example a PSTN gateway), so if I recevie an INVITE from a TCP client I must use: t_relay("udp:gateway:5060") since the following would fail: $rd = "gateway"; t_relay(); The problem is that this second approach is more flexible and it can be used in block routes, being them independent fo the final destination (set prevously with asignements like "$rd = gateway"). A solution for this could be allow $rP: $rP - reference to transport protocol of R-URI so I could do: $rP = "UDP"; $rd = "gateway"; t_relay(); But unfortunatelly $rP is no writeable: http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#assignment Another solution could be if "force_send_socket" function would allow setting **just** the protocol type, but it doesn't allow it. So there is no solution excepting making dirty the script and setting all the destination (protocol:ip:port) using "t_relay(protocol:ip:port)". Why $rP is no writeable? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 4

Re: [OpenSER-Users] mediaproxy server NEW FEATURE!!!
by Gonzalo J. Sambucaro 29 Feb '08

29 Feb '08

Raju, Thanks for testing it, now I am working in making it more secure. Regards, Gonzalo. > Gonzalo, > > Thanks!! It works very well esp. we have few clients > behind DSL which is unstable and the connectivity is > remarkably improved. > > Good contribution to mediaproxy. > > regards, > > Raju > --- "Gonzalo J. Sambucaro" > <gonzalo.sambucaro(a)mslc.com.ar> wrote: > >> Raju, >> please replase the file rtphandler.py. >> >> Regards, >> Gonzalo. >> >> > Hi Gonzalo, >> > >> > Can you please let me know how I can try out this >> > addition. I am currently using Mediaproxy 1.9.1 >> along >> > with openser-tls-1.1.1. >> > >> > >> > Please do let me know where I can get this patch. >> > >> > regards, >> > >> > Raju >> > --- "Gonzalo J. Sambucaro" >> > <gonzalo.sambucaro(a)mslc.com.ar> wrote: >> > >> >> Andreas, >> >> our local ISP´s are disconnecting DSL >> lines >> >> more ofthen but it >> >> renews the IP almost instantaneously. >> >> >> >> Regards, >> >> Gonzalo. >> >> >> >> > Hello, >> >> > nice feature, i know the problem, because our >> >> local ISP´s are >> >> > disconnecting DSL lines every 8 hours. >> >> > But the DSL line needs nearly 8 seconds, to be >> up >> >> and running again(PPP >> >> > dialup), if i can´t hear my >> >> > partner for a few seconds i will disconnect the >> >> line and redial, my >> >> > opinion. >> >> > >> >> > Same in mobile environments, when the handover >> >> doesn´t work, saying >> >> > "hello? - hello?" -> hangup -> >> >> > redial. >> >> > >> >> > But i´m willing to try, leave me a note, where >> i >> >> can download your >> >> > version. >> >> > >> >> > regards, >> >> > Andreas M. >> >> > >> >> > >> >> > >> >> > Gonzalo J. Sambucaro schrieb: >> >> >> Folks, >> >> >> I developed a new feature in the Media >> >> Proxy project of >> >> >> AG-Proyect. >> >> >> With this new feature the Media Proxy Server >> is >> >> now able to manage the >> >> >> problem of the NAT IP change in an established >> >> call. It is >> >> >> currently in production and working without >> any >> >> problem. >> >> >> >> >> >> Somebody wants to test it? >> >> >> >> >> >> I would like to know what to do to make this >> >> feature available to >> >> >> everybody. Does anybody know? >> >> >> >> >> >> Regards, >> >> >> >> >> >> ------------------------------ >> >> >> >> >> >> Details: >> >> >> ------- >> >> >> Supose that there is an established rtp >> >> connection between two >> >> >> endpoints and the media proxy is in the middle >> >> doing the relay of >> >> >> the rtp streams, let´s say >> >> >> >> >> >> (MP = mediaproxy) >> >> >> >> >> >> EP A <--------->[NAT with IP1]<-----------> MP >> >> <--------------------> EP >> >> >> B >> >> >> >> >> >> EP A sends rtp to MP_IP:MP_PORT passing >> through >> >> the NAT box. >> >> >> EP B sends rtp to MP_IP:MP_PORT without >> passing >> >> through a NAT box. >> >> >> >> >> >> The MP know that the caller = >> NAT_IP1:NAT_PORT1, >> >> and the called = >> >> >> EP_B_IP:EP_B_PORT >> >> >> >> >> >> Now, supose that the NAT box change their >> PUBLIC >> >> IP from IP1 to IP2, so >> >> >> this escenary >> >> >> >> >> >> EP A <------->[NAT with IP1]<---------> MP >> >> <-------------> EP B >> >> >> >> >> >> will change to this >> >> >> >> >> >> EP A <------->[NAT with IP2]<---------> MP >> >> <--------------> EP B >> >> >> >> >> >> so the MP should detect that change of IPs and >> >> continue relaying the rtp >> >> >> streams but now to IP2:PORT2 instead of >> >> IP1:PORT1. >> >> >> >> >> >> Well, that was the situation y have >> experienced. >> >> >> >> >> >> To fix this, I developed this solution, >> changing >> >> rtphandler.py file: >> >> >> >> >> >> 1) When the first rtp packet of a source >> arrives, >> >> save the SSRC field in >> >> >> the MP. >> >> >> - Save the SSRC of the caller. >> >> >> - Save the SSRC of the called. >> >> >> >> >> >> 2) If arrives a rtp packet with unknown source >> IP >> >> but with the same SSRC >> >> >> field of some of the two streams, updates the >> >> binding (with the new IP >> >> >> detected) between the caller and the MP or >> >> between the called and the MP >> >> >> according to the field SSRC previously saved. >> >> >> >> >> >> Note: SSRC (RFC 3550 RTP), (from the rfc: >> "The >> >> SSRC identifier carried >> >> >> in >> >> >> the RTP header and in various fields of RTCP >> >> packets is a random 32-bit >> >> >> number that is required to be globally unique >> >> within an RTP session ") >> >> >> >> >> >> -------------------------------- >> >> >> >> >> > >> >> > -- >> >> > g, >> >> > Andreas M. >> >> > >> >> > _______________________________________________ >> >> > Users mailing list >> >> > Users(a)lists.openser.org >> >> > >> >> >> > >> > http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > >> >> >> >> >> >> -- >> >> Gonzalo J. Sambucaro >> >> Ingeniería de Software >> >> Tel: +54-341-4230504 >> >> MSLC >> >> gonzalo.sambucaro(a)mslc.com.ar >> >> www.mslc.com.ar >> >> Ocampo y Esmeralda - Vivero de Empresas de Base >> >> Tecnológica >> >> Ciudad Universitaria Rosario UNR, CCT CONICET >> >> Rosario - Santa Fé - Argentina >> >> >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users(a)lists.openser.org >> >> >> > >> > http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> >> > >> > >> >> >> -- >> Gonzalo J. Sambucaro >> > === message truncated ===> >> ''' >> Handle media stream relaying and provide control >> ports to receive and >> process requests from SER (SIP Express Router) or >> the proxydispatcher. >> >> Copyright 2003-2005 Dan Pascu >> ''' >> >> import asyncore >> import socket >> import sys, os, grp, errno, traceback, atexit >> import time >> import struct >> import re >> import weakref >> >> from utilities import * >> from datatypes import * >> from version import release >> from request import Request, InvalidRequestError >> from configuration import readSettings, >> ConfigSection, Boolean, dumpSection >> >> >> class ProxyConfig(ConfigSection): >> _dataTypes = {'socket': ControlSocket, >> 'proxyIP': IPAddress, >> 'portRange': PortRange, >> 'TOS': HexNumber, >> 'listen': NetworkAddress, >> 'allow': NetworkRangeList} >> socket = '/var/run/mediaproxy.sock' >> group = 'openser' >> proxyIP = thisHostIP ## defined in utilities >> portRange = (60000, 65000) >> TOS = 0xb8 >> listen = None >> allow = None >> idleTimeout = 60 >> holdTimeout = 60*60 >> forceClose = 0 >> >> >> ## We use this to overwrite some of the settings >> above on a local basis if needed >> readSettings('MediaProxy', ProxyConfig) >> #dumpSection(ProxyConfig) >> >> if ProxyConfig.allow and NetworkRange('Any') not in >> ProxyConfig.allow: >> ## Always add ourselves to the accept list >> myself = NetworkRange('127.0.0.1') >> if myself not in ProxyConfig.allow: >> ProxyConfig.allow.append(myself) >> >> maxDataSize = 8*1024 >> >> try: >> from accounting import accounting, StopRecord, >> StopRecordSerializer, QueuedItemProcessingThread >> except ImportError: >> warning("accounting is enabled but the >> accounting module is missing. accounting is not >> available!") >> accounting = Null() >> dispatcher = Null() >> class StopRecord(object): >> def __new__(typ, *args, **kwargs): >> return None >> else: >> class >> DispatcherNotifyThread(QueuedItemProcessingThread): >> ''' >> Notify the proxydispatcher of calls that did >> timeout and provide >> the dispatcher with the accounting >> information for them. >> ''' >> def __init__(self): >> >> QueuedItemProcessingThread.__init__(self, >> name='DispatcherNotify', >> file='.dispatcher-notify.dat') >> >> def process(self, record): >> address = record['dispatcher'] >> expires = record['expires'] >> del record['dispatcher'], >> record['expires'] >> >> msg = 'timeout %s %s\n' % (record['id'], >> StopRecordSerializer().dump(record)) >> >> s = socket.socket(socket.AF_INET, >> socket.SOCK_STREAM) >> try: >> s.settimeout(5) >> except AttributeError: >> pass >> try: >> start = time.time() >> try: >> s.connect(address) >> s.send(msg) >> result = >> s.recv(maxDataSize).strip() >> except (socket.error, >> socket.gaierror, socket.herror, socket.timeout), >> why: >> try: motive = >> why[1] >> except IndexError: motive = why >> now = time.time() >> if now < expires: >> warning("couldn't notify >> dispatcher at %s (%s). will retry later." % >> (address[0], motive)) >> record['dispatcher'] = >> address >> record['expires'] = expires >> # eventually use a different >> queue only for failures >> delay = 1 >> duration = now - start >> if duration < delay: >> >> time.sleep(delay-duration) >> self.put(record) ## put back >> in the queue and process later >> else: >> error("couldn't notify >> dispatcher at %s (%s). giving up." % (address[0], >> motive)) >> # eventually log the record >> to syslog or a file >> return >> if result.lower() == 'ok': >> info("notified dispatcher at %s >> about expired session %s" % (address[0], >> record['id'])) >> elif result.lower() == 'not found': >> info("session %s already closed >> by the dispatcher at %s" % (record['id'], >> address[0])) >> else: >> error("failed to notify >> dispatcher at %s (returned `%s')." % (address[0], >> result)) >> # eventually log the record to >> syslog or a file >> finally: >> s.close() >> >> def notify(self, session): >> if not session.dispatcher: >> return ## no dispatcher to notify. >> record = StopRecord(session) >> record['dispatcher'] = >> session.dispatcher >> record['expires'] = time.time() + 3600 >> ## try for an hour >> self.put(record) >> >> dispatcher = DispatcherNotifyThread() >> dispatcher.start() >> >> ## Internal state of mediaproxy (DON'T TOUCH!!!) >> >> ## A few shortcuts, for faster access >> proxyIP = ProxyConfig.proxyIP >> minPort = ProxyConfig.portRange[0] >> maxPort = ProxyConfig.portRange[1] >> crtPort = minPort >> >> Sessions = {} >> >> ## For computing traffic through mediaproxy >> byteBucket = [0, 0, 0] >> Traffic = [0, 0, 0] >> >> ## Non public networks. This includes RFC1918 >> networks, localnet (127.0.0.0/8), >> ## broadcast networks (224.0.0.0/4) and 0.0.0.0/8 >> nonPublicNetworks = [ >> {'name': '0.0.0.0', 'value': 0x00000000L, >> 'mask': 0xff000000L}, >> {'name': '10.0.0.0', 'value': 0x0a000000L, >> 'mask': 0xff000000L}, >> {'name': '127.0.0.0', 'value': 0x7f000000L, >> 'mask': 0xff000000L}, >> {'name': '172.16.0.0', 'value': 0xac100000L, >> 'mask': 0xfff00000L}, >> {'name': '192.168.0.0', 'value': 0xc0a80000L, >> 'mask': 0xffff0000L}, >> {'name': '224.0.0.0', 'value': 0xe0000000L, >> 'mask': 0xf0000000L} >> ] >> >> rtpPayloads = { >> > === message truncated === > > -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4230504 MSLC gonzalo.sambucaro(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

1 0

Re: [Serusers] SIP Express Bundle. Step 1.
by Mike Trest - Personal 29 Feb '08

29 Feb '08

At 03:42 AM 2/29/2008, samuel wrote: >I can't recall right now whehter we have discussed the option of >creating in parallel a vmware/xen image so people can start using it >in their favorite virtual solution...more things for the future (?) We discussed. Conclusion was future options , not first out of the gate.

1 0

Re: [OpenSER-Users] [OpenSER-Devel] New module - userblacklist
by Henning Westerholt 29 Feb '08

29 Feb '08

On Tuesday 26 February 2008, Ovidiu Sas wrote: > Can you update the README file with the structure of the table? > Also a "Database setup" section would help a lot. Hi Ovidiu, the README was updated with some informations about the database setup. Please let me know if you have some other suggestions. Cheers, Henning

1 0

Re: [OpenSER-Users] [OpenSER-Devel] Blacklist
by Bogdan-Andrei Iancu 29 Feb '08

29 Feb '08

Yes, it is . Thiago Maluf wrote: > Is it there available this parameter in Openser 1.2.X? > > On 2/29/08, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > >> Hi Thiago, >> >> You just have to disable the dns_blacklist support in openser if your >> Asterisk GW is not compliant from reply codes point of view. >> >> Set : >> disable_dns_blacklist = yes >> in your script >> >> Regards >> Bogdan >> >> Thiago Maluf wrote: >> >>> Hi List, >>> In my environment I have one OpenSER and one gateway Asterisk. >>> The Gateway work send the call to PSTN or H.323 World. >>> Then when I send one call to Asterisk and the my E1 link is out, it >>> send to OpenSER one 503 Response! >>> After it, every calll during 4 minutes Openser don't send the call to >>> Asterisk. Then I don't send call to H.323 world for example. >>> I read this page http://www.mail-archive.com/users@openser.org/msg11839.html >>> And I believe that I would possible to change this time to 0 minutes. >>> But I am reading this code section and I don't find where I make this change. >>> Would someone help me? >>> thanks in advanced. >>> Thiago >>> >>> >>> >> > > >

1 0

[OpenSER-Users] New module - userblacklist
by Henning Westerholt 29 Feb '08

29 Feb '08

Hi, i just imported a new module, called 'userblacklist', into the svn trunk. To quote from the README: The userblacklist module allows OpenSER to handle blacklists on a per user basis. This information is stored in a database table, which is queried to decide if the number is blacklisted or not. An additional functionality that this module provides is the ability to handle global blacklists. This lists are loaded on startup into memory, thus providing a better performance then in the userblacklist case. This global lists could also used to prevent the blacklisting of important numbers, as whitelisting is also supported. This is useful for example to prevent the customer from blocking emergency call number or service hotlines. The module exports two functions, check_blacklist and check_user_blacklist for usage in the config file. Furthermore its provide a FIFO function to reload the global blacklist cache. The documentation for this module can be found at: http://www.openser.org/docs/modules/devel/userblacklist.html Best regards, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe

5 13

[OpenSER-Users] OpenSER 1.2.2 and 302 Moved, funny log issue?
by Tobias Lindgren 28 Feb '08

28 Feb '08

Hi there, I was trying to do this in OpenSER 1.2.2: failure_route[1] { if (t_check_status("302")){ xlog("L_ERR", "302 source ip address is $si"); if (src_ip == 192.168.5.1) { # Do something }; }; }; But the proxy sending the 302 (192.168.5.1) is not the one who ends up in my logs, instead my sip-server (192.168.1.1) who initiated the call ends up in the log. "302 source ip address is 192.168.1.1". Have I understood the "$si" psuedo variable wrong? Can I find the correct source address of the 302-message somehow? Br, /Tobias

4 10

[OpenSER-Users] Building openser-1.3.0 on Debian stable
by Sajith T S 28 Feb '08

28 Feb '08

When trying to build openser-1.3.0-tls (tarball from openser.org) on Debian stable, "make all" throws this error: if [ "no" = "yes" ]; then \ cd utils/db_berkeley; make all ; \ fi ; \ /bin/sh: -c: line 4: syntax error: unexpected end of file make: *** [utils] Error 2 The last backslash (after "fi ;") is the problem (line #521 in the top-level Makefile). This is in SVN trunk and 1.3 branch too as far as I can tell. Thanks, Sajith. -- Don't you wish you had more energy... or less ambition? 9DB8FF06 : CB80 0BA6 7D13 B10A 6FBB D43E B4D2 28AD 9DB8 FF06 echo tbkjui(a)opozfo.jo | tr [b-y] [a-x]

2 1

Re: [OpenSER-Users] OpenSER 1.2.2 and 302 Moved, funny log issue?
by Zahid Mehmood 28 Feb '08

28 Feb '08

Thank you for your replies. The sipwise code did have an append branch and a branch route that dropped local branches. -- zahid -----Original Message----- From: "Andreas Granig" <agranig(a)sipwise.com> To: "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro> Cc: "Zahid Mehmood" <zm23(a)columbia.edu>; "users(a)lists.openser.org" <users(a)lists.openser.org> Sent: 2/28/08 6:32 AM Subject: Re: [OpenSER-Users] OpenSER 1.2.2 and 302 Moved, funny log issue? Hi, Puh, it's been a while since I wrote that, so let me think :) As far as I remember, the idea was to extract the first branch and push it into R-URI, then jump somewhere into the route sections where quite some R-URI manipulation and checking is done (and most of the functions can only operate on r-uri, so...)... after that, the ruri is again appended as branch before sending it out. Only problem is that you can only use one destination. Bogdan, what exactly do you mean by "if it does append_branch somewhere later, it will just duplicate a branch"? Cheers, Andreas Bogdan-Andrei Iancu wrote: > Hi Zahid, > > To be honest, I do not see the purpose of this: > > # get last URI from destination-set and set it as R-URI > avp_delete("$avp(s:tmp)/g"); > $avp(s:tmp) = $ds; > avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); > avp_pushto("$ru", "$avp(s:tmp)"); > > This code fetches the first branch (by using the $ds pv and the regexp) > and push it as RURI; but if there is no following append_branch(), it > has 0 effect. And if it does append_branch somewhere later, it will just > duplicate a branch.... > > Maybe Andreas (here CCed) can give you some details on this..... > > Regards, > Bogdan > > > Zahid Mehmood wrote: >> Hi, >> If get_redirect() is already pushing the contacts as branches, >> then what is the purpose / benefit / drawback of doing something >> like this: >> >> taken from sipwise online config: >> >> if(!get_redirects("1:1")) >> { >> >> xlog("L_ERROR", "Failed to fetch contact '$ct' from >> 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >> acc_db_request("480", "acc"); >> t_reply("480", "Temporarily Unavailable"); >> exit; >> } >> # get last URI from destination-set and set it as R-URI >> avp_delete("$avp(s:tmp)/g"); >> $avp(s:tmp) = $ds; >> avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); >> avp_pushto("$ru", "$avp(s:tmp)"); >> >> >> I'd greatly appreciate your answers as they will help me better >> understand the use of branches/destination set. >> >> Thanks. >> >> >

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Re: [OpenSER-Users] asterisk load balance dispatcher
by Daniel-Constantin Mierla 28 Feb '08

28 Feb '08

Hello, I believe your problem is related nat traversal. Do you do port forwarding on the firewall? As a tip, you should do record routing on openser. To troubleshoot, watch the sip traffic on your network and on the firewall box -- see whether messages are going caller -- asterisk all the time or they are blocked by firewall. Cheers, Daniel On 02/28/08 12:59, Antonio Basti wrote: > > Hi , i’ve a problem with the dispatcher module, for the load balancing > for asterisk > > The schema of the network is this : > > Firewall (public ip: 199.199.199.199:5060) > > | > > | > > ------------------- > > Pc1 pc2 > > Pc1: 192.168.0.1:5060 Asterisk Pc2:192.168.0.3:5060 Asterisk > > Pc1:192.168.0.2:5060 Openser > > The ip public is natted with Openser > > The dispatcher.list is: > > 1 sip:192.168.0.1:5060 > > 1 sip:192.168.0.3:5060 > > Openser.cfg: > > route{ > > if ((method=="INVITE") ){ > > ds_select_dst("1","4"); # 4 = round-robin > > t_relay(); > > exit(); > > } > > } > > In asterisk for test there is an ivr ,and when I make a call from an > other site to this public ip, ast answer but after few seconds I see > from the log of the pc01 the the call hang but from the remote > asterisk (the one where I make the call) the call remain up. > > The configuration of openser is correct ? > > thanks > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 2898 (20080223) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] asterisk load balance dispatcher
by Antonio Basti 28 Feb '08

28 Feb '08

Hi , i've a problem with the dispatcher module, for the load balancing for asterisk The schema of the network is this : Firewall (public ip: 199.199.199.199:5060) | | ------------------- Pc1 pc2 Pc1: 192.168.0.1:5060 Asterisk Pc2:192.168.0.3:5060 Asterisk Pc1:192.168.0.2:5060 Openser The ip public is natted with Openser The dispatcher.list is: 1 sip:192.168.0.1:5060 1 sip:192.168.0.3:5060 Openser.cfg: route{ if ((method=="INVITE") ){ ds_select_dst("1","4"); # 4 = round-robin t_relay(); exit(); } } In asterisk for test there is an ivr ,and when I make a call from an other site to this public ip, ast answer but after few seconds I see from the log of the pc01 the the call hang but from the remote asterisk (the one where I make the call) the call remain up. The configuration of openser is correct ? thanks

1 0

[OpenSER-Users] What is "timezone" field in "subscriber" table for?
by Iñaki Baz Castillo 28 Feb '08

28 Feb '08

Hi, "subscriber" table has a column "timezone". While it could be useful for CDR purposes and so, I find no relationship with OpenSer itself of its modules. Why does this column exist by default in subscriber table? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] What is "timezone" field in "subscriber" table for?
by Bogdan-Andrei Iancu 28 Feb '08

28 Feb '08

Hi Jonathan, More or less this is the whole idea - many openser admins configures a different additional set of fields in the subscriber table, based on the needs of some provisioning tools. Of course, we cannot find a common set for this extra (non-openser) fields, so the most clear way to do it is to keep in the openser default DB struct only the fields that are used by openser. Then, anybody may "customize" the tables, based on his particular needs, without conflicting with the "particular" needs of somebody else ;) Regards, Bogdan Jonathan K. Creasy wrote: > I have applications that use this to display information to the customer (CDR data for example). You may still want to take it out but I just wanted to speak up that there could still be valid use for it. > > -Jonathan > > >> -----Original Message----- >> From: users-bounces(a)lists.openser.org [mailto:users- >> bounces(a)lists.openser.org] On Behalf Of Bogdan-Andrei Iancu >> Sent: Thursday, February 21, 2008 3:50 AM >> To: Henning Westerholt >> Cc: users(a)lists.openser.org >> Subject: Re: [OpenSER-Users] What is "timezone" field in "subscriber" >> table for? >> >> Hi Henning, >> >> I would definitely say so. >> >> Thanks and regards, >> Bogdan >> >> Henning Westerholt wrote: >> >>> On Thursday 21 February 2008, Bogdan-Andrei Iancu wrote: >>> >>> >>>> Hi Inaki, >>>> >>>> Right - this field is more or less used only by serweb (cdr >>>> >> purposes) - >> >>>> openser is not using it at all. >>>> >>>> >>> Hi Bogdan, >>> >>> so i guess its safe if i remove this from trunk? >>> >>> Henning >>> >>> >>> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > >

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[Serusers] off-topic: nokia e90 sip provisioning via sms ota
by Mark Anthony C. Delfin 28 Feb '08

28 Feb '08

Hi List, Do you have some sample xml config file for provisioning sip settings on nokia E90? I'm using kannel with an smpp connection to send the config. Thanks. Mark D.

1 0

[Serusers] UDP to TCP using SER (as conversion purpose), need ser.cfg example
by Goh Sheng Mai 28 Feb '08

28 Feb '08

hi i have large voip udp packets over internet problems and i need the transfer medium to be tcp. below is what i had in mind. SIPp_client A (TCP out) -> (listen TCP) SER 1 (UDP out) --> Soft-switch (UDP in /out only) --> (to) ring Client B (listen UDP) Client A and Client B registers to above Soft-switch. All the above are individual servers (Linux CentOS based) i need help in a (conversion) ser.cfg. Can anyone post a sample ?? below is my ser.cfg (tried using t_relay_to_udp(), rewritehost(), rewritehostport(), forward functions BUT none seems to work the way i want it to) SER 1 == 192.168.10.203 Softswitch == 192.168.10.251 Client A == 192.168.10.221 or .236 (SIPp client-registered to soft-switch successfully) Client B == 192.168.x.x (registered to soft-switch successfully) Thanks shengmai shengmai_goh(a)innomedia.com.sg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* 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 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- 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 -- 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 >= 2048 ) { 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(); #if (method == "INVITE") { # if (uri =~ "sip:*@*"){ # log(1, "Forwarding to Vsp12K\n"); # rewritehostport("192.168.10.204:5060"); # t_relay(); # break; # } #}; if (uri =~ "sip:*@*"){ #log(1, "Forwarding to Vsp12K\n"); #rewritehostport("192.168.10.204:5060"); if (uri =~ "sip:*@192.168.10.203"){ rewritehostport("192.168.10.251:5060"); } t_relay_to_udp("192.168.10.251", "5060"); #t_relay(); break; }; # 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(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ----------------------------------------------------------------- # Pass on stuff going to PSTN to Asterisk # ----------------------------------------------------------------- #if (uri=~"^sip:*@.*") { # rewritehostport ("*192.168.10.204:5060"); # if (!t_relay()) { # # sl_send_reply ("403", "prout"); # sl_reply_error(); # }; # break; #}; rewritehostport("192.168.10.204:5060"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; break; }

2 3

[OpenSER-Users] No answers whatssoever??
by Taisto Qvist (WM) 27 Feb '08

27 Feb '08

Hi, and sorry if I am getting repetitious but since I am not getting any replies on what I thought was an important question(rfc-complicance) I need to ask the question again...(at least indicate why you disagree with me?) I was hoping someone could comment on what I thought was some fairly clear rfc-references, that indicate that an ACK for 2xx should NOT match the INVITE transaction, since it should be terminated at reception/sending of 2xx. I hope I dont offend in any way, I would just like to get some clarification on this. I hope you can clarify your standpoint, especially if you disagree with me, which I get the feeling you do. Kind Regards Taisto Qvist Taisto Qvist wrote: > I seems like we dont agree on the interpretation of the txn handling > for invite. Sure, there is a timer for handling retransmitted responses, > but thats in earlier states than Terminated. > Sure, there is nothing that says that the inner workings of your > transactions > must exactly behave as the statemachine diagrams in the rfc as long as the > external behavior stays the same, but still, I think that the ACK for 2xx > should not match the INVITE txn. > (It probably boils down to how rfc-isch you want to be.) > > >The end-to-end ACK establish a separate transaction (RFC 3261) and these > >ACKs are not match as part of the INVITE transactions, but correlated > >with them. > > [TQ] > Where does it say that ACKs establish a separate transaction? > There is no defined ACK transaction, only INVITE/nonInvite, server and > client. > The ACK is either a part of the INVITE txn, or its a separate request, > but I would > never call it a standalone transaction. > The real purpose of txn's (in my view), is to enable/simplify forking, > and to > handle retransmissions. Retransmissions of ACK and 2xx, are done by > the UAC/UAS, > so there is no need for a ack-txn. > > >but even describe the wait timer. So, there is no contradiction. > [TQ] > I'd say there is. Where does it describe that this wait-timer should be > used for all responses, or for matching ALL acks? > > My two main reasons for saying that ACKs for 2xx should not be matched, > come from the following two texts. (17.1.1.2, and 17.2.1) > > 17.1.1.2 (client invite txn) > The client transaction MUST be destroyed the instant it enters the > > "Terminated" state. This is actually necessary to guarantee correct > operation. The reason is that 2xx responses to an INVITE are treated > differently; each one is forwarded by proxies, and the ACK handling > in a UAC is different. Thus, each 2xx needs to be passed to a proxy > core (so that it can be forwarded) and to a UAC core (so it can be > acknowledged). No transaction layer processing takes place. > Whenever a response is received by the transport, if the transport > layer finds no matching client transaction (using the rules of > Section 17.1.3), the response is passed directly to the core. Since > the matching client transaction is destroyed by the first 2xx, > > subsequent 2xx will find no match and therefore be passed to the > core. > > [TQ] > It even says "actually nessesary to guarantee...." > Since the txn's are there to (among other things) absorb retransmissions, > the receiving the 2xx MUST destroy the txn, so that when the next > retransmission > of 2xx is received, it is not consumed by the txn layer. > This is what your txn-layer does for 3++ right? Any additional 3++ > received on > a txn while waiting for Timer D to expire, will just be consumed, and > the proxy > core will never have to process it. > > > 17.2.1. (server invite txn) > The purpose of the "Confirmed" state is to absorb any additional ACK > messages that arrive, triggered from retransmissions of the final > response. > ..snip... > Once the transaction is in the "Terminated" state, it MUST be destroyed > immediately. > > [TQ] > The Confirmed state is there to absorb retransmissions, not the > terminated. > When receiving 2xx you go directly to terminated bypassing confirmed. > > Also, in 18.2.1 the text explicitly says: > > .. Note that when a UAS core sends a 2xx response to INVITE, > the server transaction is destroyed. This means that when the ACK > arrives, > there will be no matching server transaction, and based on this rule, > the ACK is passed to the UAS core, where it is processed. > > Thats faily clear right? :-) > The think that puzzles my a little bit though, is that you must have added > "extra" code for this matching...? > I mean, you're using the ;branch=z9xxxx for transaction matching right, if > its there? And the ACK for 2xx doesnt have the same branch, indicating > a "new txn". > > So matching ALL ACKs would imply that for any ACK that doesnt match a txn, > your really checking up Call-Id's, tags and stuff, just to be able to > match > this ACK to the 2xx? > > Or are you checking callid's, cseq's, tags, and stuff, for ALL > requests in your > txn matching? > > Now, I been working on a sip-stack the last few years myself, and I > naturally > know that there is often a need to match the ACK, and the stack I've > build has > similar functions, it just doesnt do it on the txn layer, where we try > to be as > RFC-isch as possible. > > What I would like, is simply have some function-parameters to the > t_check_trans(<DEFANGED_param>), > or even better in my little mind, a "modparam("tm", > "ack_2xx_matching", 1)". > > I am looking forward to hearing your reply, and in the mean time, > thanks to all of you developers for an extraordinary software :-) > > Regards > Taisto Qvist > > > Hi guys, > > > > Just to bring some clarifications on the TM module. > > > > once a transaction is completed (negative or 2xx reply), it is put on > > wait (wait timer - see RFC3261) for catching any potential > > retransmissions of replies. > > So, the transaction is completed, but not removed from memory - RFC does > > not say that you need to trash immediately all the transaction > > information, but even describe the wait timer. So, there is no > > contradiction. > > > > The ACK (for 2xx)catching is done based on the completed INVITE > > transaction (from wait timer) - nothing else. > > > > > > Inaki, could you please detail what you mean by: > > > > <quote> > > In my opinion OpenSer does a special treatment for ACK in tm mode, > even if > > they are for failed transaction (hop-by-hop ACK's) or succesfull INVITE > > (end-to-end ACK's). > > > > </quote> > > > > Maybe I can explain you more if I understand you question.... > > > > Regards, > > Bogdan > ---- Taisto Qvist, IP-Solutions Mobile: +46-708-88 02 63 "We are Pentium of Borg. Division is futile, You will be approximated"

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[OpenSER-Users] no-answer timeout
by Mark Sayer 27 Feb '08

27 Feb '08

Hi - I am using 1.2.1-notls. When using t_relay() to INVITE a call, I assume that the timeout before returning a 408 would be governed by fr_inv_timer and fr_inv_timer_avp. Although the 1.2 documentation states that the default for fr_inv_timer is 120 seconds I seem to get a timeout at 60 seconds and neither fr_inv_timer or fr_inv_timer_avp seems to have any effect. Have a made an error in interpreting the docs? How should I be varying this timeout? Thanks, Mark

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[OpenSER-Users] Out of order notifications
by Leonard Broman 27 Feb '08

27 Feb '08

Hi! I have integrated OpenSER as a dedicated presence server in an ims network and I'm somewhat suffering from the delays the traversal of the ims core causes. Here's whats happening: Upon subscription the presence module sends a 202 Accepted (actually named OK, cosmetic bug?). At almost the same instant it also sends the first, in-dialog notify. What happens later is that sometimes the notify arrives before the ok. When this happens the client (on top of jain-sip) cannot identify the subscription dialog for the notification. It does not treat it correctly and does not respond OK to it. The absence of the response results in a timeout and the presence server invalidates the subscription. Now, to solve this issue, would it be wise to introduce a parameter in the presence module which delays to transmission of the first notification to ease up the situation for the client. Another solution is of course that the client simply inspects the local tag to prematurely identify the upcoming sip dialog for the subscription and then be able to treat the request properly. However, how does this stand against the standards? I would be grateful for your input on the subject. Leonard Broman

1 0

[OpenSER-Users] XMLRPC
by Helmut Kuper 27 Feb '08

27 Feb '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to get some Informations from openser via mi_xmlrpc module. Unfortunately I allways get an "404 Not found" Reply back from openser. I checked the communication between openser and php xml client. My client sends this to openser: POST / HTTP/1.0 Host: 85.16.246.9 Content-Type: text/xml User-Agent: me Content-length: 157 <?xml version="1.0"?> <methodCall> <methodName>get_statistics</methodName> <params> <param><value><string>tm:</string></value></param> </params></methodCall> That looks like as documentated in mi_xmlrpc module doc. I send different MI commands to openser: real and non real. But I always get that 404 error. In mi_xmlrpc log is nothing except those 404 answeres again. Debug level 9 says nothing. Sending MI commands to openser via mi_fifo works. Is there anything else to configure for xmlrpc? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHxEEz4tZeNddg3dwRAmpwAJoDKWW6bWU3zRyaaisa6HOOX6JdKwCcDAqM 2pmp4F96L85s8SmEI5L76ak= =23Fr -----END PGP SIGNATURE-----

2 2

Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA
by Anca Vamanu 27 Feb '08

27 Feb '08

Hi, I made the fix for both 1.3 and 1.4 versions. There is no directory with patches. The newest version is found in the on svn for branch 1.3 and trunk respectively. If you want to make use of this patch you have to update to download the latest svn version. See: http://www.openser.org/mos/view/Download/. <http://www.openser.org/mos/view/Download/> regards, Anca Vamanu Jeremy George wrote: > Anca, > > Is this a patch for 1.3? Or, a fix that went into the 1.4 development > tree? If it's for 1.3, how do I get it? I looked through openser.org but > can't find a directory of patches. Thanks much. > > - Jeremy > > > >> -----Original Message----- >> From: users-bounces(a)lists.openser.org [mailto:users- >> bounces(a)lists.openser.org] On Behalf Of Anca Vamanu >> Sent: Thursday, February 21, 2008 11:14 AM >> To: Zahid Mehmood >> Cc: users(a)lists.openser.org >> Subject: Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA >> >> Hi, >> >> Thank you for the print. It helped finding the bug. >> I have just committed a fix. It would be great if you would update with >> the svn version, test and confirm if it works. >> >> Thanks, >> Anca Vamanu >> >> Zahid Mehmood wrote: >> >>> Hi Anca, >>> Thank you for your quick response. Attached below is the >>> information you requested. >>> >>> Thanks. >>> >>> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > > > >

2 1

Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA
by Jeremy George 27 Feb '08

27 Feb '08

Hey Anca, That's great. Thanks. So, I have now openser-1.3.x-svn-latest.tgz . If it doesn't wear your patience thin with too-simple questions, I'd like to ask just a couple more. Is this version still considered a stable release? And, I can't find a file within the tarball that distinguishes this version from others. For example, the ChangeLog doesn't show it. Where would I find such information? Again, thanks for your expertise, and patience. - Jeremy >-----Original Message----- >From: Anca Vamanu [mailto:anca@voice-system.ro] >Sent: Wednesday, February 27, 2008 10:24 AM >To: 'Jeremy George' >Cc: users(a)lists.openser.org >Subject: Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA > >Hi, > >I made the fix for both 1.3 and 1.4 versions. There is no directory with >patches. The newest version is found in the on svn for branch 1.3 and >trunk respectively. If you want to make use of this patch you have to >update to download the latest svn version. See: >http://www.openser.org/mos/view/Download/. ><http://www.openser.org/mos/view/Download/> > >regards, >Anca Vamanu > >Jeremy George wrote: >> Anca, >> >> Is this a patch for 1.3? Or, a fix that went into the 1.4 development >> tree? If it's for 1.3, how do I get it? I looked through openser.org but >> can't find a directory of patches. Thanks much. >> >> - Jeremy >> >> >> >>> -----Original Message----- >>> From: users-bounces(a)lists.openser.org [mailto:users- >>> bounces(a)lists.openser.org] On Behalf Of Anca Vamanu >>> Sent: Thursday, February 21, 2008 11:14 AM >>> To: Zahid Mehmood >>> Cc: users(a)lists.openser.org >>> Subject: Re: [OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA >>> >>> Hi, >>> >>> Thank you for the print. It helped finding the bug. >>> I have just committed a fix. It would be great if you would update with >>> the svn version, test and confirm if it works. >>> >>> Thanks, >>> Anca Vamanu >>> >>> Zahid Mehmood wrote: >>> >>>> Hi Anca, >>>> Thank you for your quick response. Attached below is the >>>> information you requested. >>>> >>>> Thanks. >>>> >>>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.openser.org >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >> >> >> >> > >

2 2

Re: [OpenSER-Users] Call timeout
by Bogdan-Andrei Iancu 27 Feb '08

27 Feb '08

Hi David, The setup time is actually controlled by TM (during the INVITE transaction). If you want to set terminate a call, see the dialog module, in openser 1.3 version: http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN461 regards, Bogdan David Villasmil wrote: > Hello, > > Timeout for a setup I understand is done with SST/Dialog, I mean > disconnect an established call. > > > thanks > > On Wed, Feb 27, 2008 at 10:16 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hi David, > > What do you mean by "call timeout" - do disconnect an established call > after some time? or timeout during call setup? > > Regards, > Bogdan > > David Villasmil wrote: > > Hello all, > > > > I have a network already operational for voip-voip/voip-pstn > > servicing about 20k customer and all is "well". At the moment I'm > > using Asterisk as a B2BUA mainly for call timeout capabilities. Is > > there absolutely NO WAY of setting a call timeout (disconnect) > after X > > seconds? Do I have to use a separate B2BUA? > > > > > > Thanks everyone. > > > > David > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

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[Serusers] OpenSER summit at VONX Spring 2008
by Greger Viken Teigre 27 Feb '08

27 Feb '08

I know many SER users also use OpenSER in combination with SER. Indeed, I see also quite a lot of confusion with people asking OpenSER questions on the SER mailing list... So, not to add to the confusion: OpenSER is a fork of the original SIP Express Router project, where OpenSER and SER code bases started to diverge at that point. It was only the SIP proxy of iptel.org that was forked, thus: SEMS, SERWeb, and sipsak are still iptel.org only projects with no forks (rtpproxy.org, formerly loosely coupled with iptel.org project has now its own home: http://rtpproxy.org). So, with that clarification, some seruser subscribers may find the below announcement of interest. I forward it on the request of Bogdan, one of the maintainers of OpenSER. g-) The second edition of OpenSER Summit will take place in San Jose, USA , on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of the event, see http://www.openser.org/mos/view/OpenSER-Summit-2008. All participants to register via OpenSER site before Friday, March 7th, will get free access to the OpenSER event. This OpenSER Summit edition is sponsored by a parallel OpenSER related event, the OpenSER Pavilion . The pavilion is a common exhibiting area - booth 1027, inside VoN expo, gathering, under the OpenSER name, six different companies working or using the project. You can find more about the OpenSER Pavilion and its participants here - http://www.openser.org/mos/view/OpenSER-Pavilion-2008. The dual event, OpenSER Summit & Pavilion is new concept of a more complex event, aiming to create a larger diversity and to give more power to the understanding of the OpenSER project.

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[OpenSER-Users] User Documentation - request for participation
by Henning Westerholt 27 Feb '08

27 Feb '08

Hello all, its a well know fact that there exists some issues with the OpenSER documentation, for some parts there is almost no documentation available at all. Existing documentation is sometimes a little bit unstructured or incomplete. Its hard for new users to understand the configuration of the server, let alone how they should setup some services. And after he or she has gone through the basics, its again the same for configuring complex modules.. During the last IRC meeting we decided to start a new effort to tackle this problems more efficiently than in the past. So a new mailling list 'openser-docs' has been created, targeted in improving and extending the existing documentation, and also create new content, for example a 'OpenSER Guide'. As its difficult for many of the existing developers to write user related documentation, because they're already to deep involved into the project, i would like to ask here for participation from you. We're seeking people that are also dissatisfied with the current situation, and want invest something to improve this. No coding experiences or knowledge about certain tools or languages are necessary. So please join us and subscribe at http://lists.openser.org/cgi-bin/mailman/listinfo/openser-docs ! Thank you, Henning

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[OpenSER-Users] Call timeout
by David Villasmil 27 Feb '08

27 Feb '08

Hello all, I have a network already operational for voip-voip/voip-pstn servicing about 20k customer and all is "well". At the moment I'm using Asterisk as a B2BUA mainly for call timeout capabilities. Is there absolutely NO WAY of setting a call timeout (disconnect) after X seconds? Do I have to use a separate B2BUA? Thanks everyone. David

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[OpenSER-Users] Fwd: [Serdev] RTPproxy has gotten its own mailing lists
by Ovidiu Sas 27 Feb '08

27 Feb '08

FYI Regards Ovidiu Sas ---------- Forwarded message ---------- From: Maxim Sobolev <sobomax(a)sippysoft.com> Date: Tue, Feb 26, 2008 at 7:17 PM Subject: [Serdev] RTPproxy has gotten its own mailing lists To: serdev(a)lists.iptel.org Hi, Just a quick note that today we've finally launched mailing lists for the RTPproxy. Anybody is welcome to join. http://lists.rtpproxy.org/mailman/listinfo Regards, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com _______________________________________________ Serdev mailing list Serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

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[OpenSER-Users] help.. openser 1.3.0 crash -- presence/ BLA
by Zahid Mehmood 26 Feb '08

26 Feb '08

Hi All, I need help with troubleshooting openser 1.3.0 crash. I am running openser 1.3.0 (not from svn) to provide only BLA presence service. The config file is based on the sample provided on openser website. This is the first time openser actually crashed. Twice before it stopped working but the process never died. Originally openser was running with debug=3. I have changed it to log more information now. What else can i do to generate more useful information to help resolve this issue? Thanks in advance for your help! -- Zahid From openser log: Feb 15 16:56:52 torte /usr/sbin/openser[25542]: 7bfbf097-25539(a)10.0.0.129 : handle_publish Feb 15 16:56:52 torte /usr/sbin/openser[25539]: 15b7f2f2-11174(a)10.0.0.129 : bla_handle_notify OK Feb 15 16:57:13 torte /usr/sbin/openser[25570]: CRITICAL:core:receive_fd: EOF on 17 Feb 15 16:57:13 torte /usr/sbin/openser[25522]: INFO:core:handle_sigs: child process 25551 exited by a signal 11 Feb 15 16:57:13 torte /usr/sbin/openser[25522]: INFO:core:handle_sigs: core was generated Feb 15 16:57:13 torte /usr/sbin/openser[25522]: INFO:core:handle_sigs: terminating due to SIGCHLD Feb 15 16:57:13 torte /usr/sbin/openser[25534]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25560]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25568]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25565]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25533]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25531]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25553]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25570]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25538]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25558]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25542]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25539]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25548]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25540]: INFO:core:sig_usr: signal 15 received Feb 15 16:57:13 torte /usr/sbin/openser[25522]: NOTICE:presence:destroy: destroy module ... bash-3.00# openser -V version: openser 1.3.0-notls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, 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 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 3304 2007-12-11 12:55:11Z bogdan_iancu $ main.c compiled on 21:52:01 Jan 6 2008 with gcc 3.4.6 gdb info: Core was generated by `/usr/sbin/openser'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libdl.so.2...done. (gdb) backtrace #0 0x00b3bba3 in strncmp () from /lib/tls/libc.so.6 #1 0x00140bcb in get_dialog (dialog=0xb60fe518, hash_code=22) at hash.c:289 #2 0x0014ae10 in subs_cback_func (t=0xb61210b0, cb_type=256, ps=0xdb6314) at send_subscribe.c:248 #3 0x00d945b2 in run_trans_callbacks (type=256, trans=0xb61210b0, req=0x0, rpl=0xffffffff, code=408) at t_hooks.c:205 #4 0x00da14ca in local_reply (t=0xb61210b0, p_msg=0xffffffff, branch=0, msg_status=408, cancel_bitmap=0xbfe95834) at t_reply.c:1255 #5 0x00da5f4c in timer_routine (ticks=89623, attr=0x0) at timer.c:254 #6 0x080b403f in start_timer_processes () at timer.c:275 #7 0x08066d60 in main (argc=1, argv=0xbfe95a94) at main.c:873

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[OpenSER-Users] MI commands
by David Villasmil 26 Feb '08

26 Feb '08

Hello all, Is there a way to generate a BYE message, disconnect message via Manager Interface? thanks all David

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[OpenSER-Users] openser+opendiameter integration--->kindly pay attention
by bhargav.b＠renovau.com 26 Feb '08

26 Feb '08

hi all, i am trying to authenticate a user using openser and opendiameter.currently i have configured both of them on the same pc.my problem is that i cannot use anything other than opendiameter due to some architechture restrictions of our company plan. when i run xlite[softphone],openser,opendiameter i get error in opendiameter client saying that"missing session id avp".i am not able to identify where exactly the problem lies,either in openser's auth_diameter module which is developed for diameter or in the opendiameter. if somebody have done anything like what i am trying to do or if anybody have some idea please help me out.Because i think its really very hard to do this without somebodies help

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Re: [OpenSER-Users] OpenSER failure when MySQL redundant database fails
by Henning Westerholt 26 Feb '08

26 Feb '08

On Monday 25 February 2008, Arturo Díaz Almagro wrote: > Please, see my comments below. > > [..] > > For recent mysql driver versions this is not needed anymore, i've > > committed in revision 3606 a improvement to the trunk version of openser > > that uses the recently introduced extensions in the mysql lib. This could > > be easily backported to your version, take a look into the mysql > > documentation if this different timeouts are supported on your driver > > version. > > Does that mean that I could compile the mysql module in the trunk version > and replace the old mysql.so module by the new generated one from the > trunk? Is it fully backward compatible? > > Kind regards. Hi Arturo Díaz Almagro, (lets continue this discussion on the user list) No, but this will not work at all. Even replacing the code and recompile will not work, the changes were to extensive. You need to backport the changes from the mentioned revision to the driver you use internally in your company. You could get the changes with 'svn diff' or search on the devel list for the commit message. Cheers, Henning

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[Serusers] OPTION using sipsak
by YAHYAOUI Mohammed 26 Feb '08

26 Feb '08

Hello, Having 1 boxes with SIP service Box A: ip a.b.c.d with sip at udp 5060 and a NMS machine with ip x.y.z.t I am looking to monitor the SIP service via OPTION sent from NMS (x.y.z.t) to the box A (no user defined on the box) Any one to help how to write the OPTION request using sipsak I tried this way but no success sipsak -f option -s sip:a.b.c.d (I don't know if we have to set the udp port with sipsak) On file option I have: OPTIONS sip:a.b.c.d:5060 SIP/2.0 Max-Forwards: 70 To: <sip:ping@a.b.c.d:5080> From: NMS <sip:x.y.z.t> Call-ID: 83185420572 CSeq: 8764 OPTIONS Allow: ACK,CANCEL Accept: application/sdp Content-Length: 0 BR

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[OpenSER-Users] rtpproxy on separate server
by Ron 26 Feb '08

26 Feb '08

Hi, What is the advantage of having rtpproxy on a separate server against having it on the same server as the sip registrar/proxy? TIA Regards, Ron

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Re: [Serusers] Intercept SIP call
by roayu 26 Feb '08

26 Feb '08

Thanks for the explanation! I will try to work it out. Roa :D -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, February 25, 2008 8:11 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Intercept SIP call Ringing multiple clients is something the server does -- not the client. Eg: User A logs into SER with IP:22.33.44.55 and leaves his client running. He then goes to work and logs into SER with IP: 88.99.66.77 In SER, there's now a table that has that information: User A: 22.33.44.55, 88.99.66.77 When user B calls, SER rings every place User A is logged in from. The first phone to answer gets the call. Call pickup is something completely different. And yes, for that, you'd need Asterisk running. N. roayu wrote: > Thanks. > > So, if the client aren't able to perform this function. Can SER do that? Or > I need to setup the Asterisk server to help me to perform the 'call pickup' > functions. Correct me if I'm wrong. > > Thanks. > > Roa Yu > > -----Original Message----- > From: SIP [mailto:sip@arcdiv.com] > Sent: Monday, February 25, 2008 11:17 AM > To: roayu > Cc: samuel; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Intercept SIP call > > When your client registers to SER as a particular user, that > registration is stored in SER. > > If you register multiple times with the same user, ALL of those contacts > are called when a call is sent to that user. > > N. > > > roayu wrote: > >> Thanks for your reply and explanation, samuel. >> >> But what do you mean by "If you want to send the initial call to both >> phone B and C, either in parallel or sequentally, then SER can easily >> do the job"? Could you please further explain on this or you can >> provide me so links to read bout this. >> >> Thanks! >> >> Roa Yu. >> >> ------------------------------------------------------------------------ >> >> *From:* samuel [mailto:samu60@gmail.com] >> *Sent:* Friday, February 22, 2008 4:39 PM >> *To:* roayu >> *Cc:* serusers(a)lists.iptel.org >> *Subject:* Re: [Serusers] Intercept SIP call >> >> The way you have explained the use case would require the phones to do >> complex SIP operations. >> Phone C need to know the dialog identifiers (dialog info package >> probably would help)of the call between A and B and then use replaces >> header to "intercept the call". I presonally haven't seen phones doing >> this. >> >> If you want to send the initial call to both phone B and C, either in >> parallel or sequentally, then SER can easily do the job. Just read a >> few mails or the getting started document and you'll see how to do this. >> >> Hope it helps, >> Samuel. >> >> 2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net>>: >> >> Hi there! >> >> Recently, I had setup a SIP environment with few sip phones, SIPphone >> A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to >> SIPphone B, but B is not able to answer the call. Can SIPphone C >> answer A's call? >> >> I would like to know whether is SER able to have the intercept sip >> calls function? If yes, what module or configuration that I need to set? >> >> Thanks in advance! >> >> Best regards, >> >> RoaYu >> >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >>

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[Serusers] gw table lookup 2.0
by Mike Trest - Personal 26 Feb '08

26 Feb '08

Hi, can someone point me to a 2.0 config snip that will lookup gw_name and return data element[s] from "gw" table? As an alternative, a simple found/not found return value will do. ..mike..

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[OpenSER-Users] Add record route in reply route?
by Peter P GMX 26 Feb '08

26 Feb '08

I have the following scenario: UA - OpenSER - SIPX - MicrosoftUM2007 As OpenSER has the public IP address and SIPX has no public IP, Openser has to be in the loop of all SIP messages. Therefore I set a record-route header in OpenSER. SipX is needed for the UDP/TCP gateway to MS UM2007 so therefore SipX needs to add another record-route header. UM2007 then receives a record with 2 record-routes (the one of OpenSER and the one of SipX) Unfortunately our UM2007 replaces the OpenSER record-route in the OK messages by the IP of SipX IP only, so Openser is out of the loop. The UA then tries to reply with ACK to the (private) IP of SipX and fails. My question: Ist here a chance to add another record-route with OpenSer's IP in OpenSER 1.3 in the reply route? According to the documentation in rr it doesn't work. Any hint is appreciated. Best regards Peter

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[OpenSER-Users] email address update
by Daniel-Constantin Mierla 25 Feb '08

25 Feb '08

Hello, my new email address for any matter is miconda(a)gmail.com -- as I was not able to access my inbox over last week, please resend to the new address any message you sent to me (not the ones sent to the lists, I will pick them up from there). Thank you, Daniel

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[OpenSER-Users] #remove
by Silvio Netto 25 Feb '08

25 Feb '08

_________________________________________________________________ Conheça o Windows Live Spaces, a rede de relacionamentos do Messenger! http://www.amigosdomessenger.com.br/

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[OpenSER-Users] Status code 408 returned by OpenSER when fr_timer expires; should it be 504?
by Anatoly Pidruchny 25 Feb '08

25 Feb '08

Hello everybody, I noticed that OpenSER replies with status code "408 Request Timeout" when the fr_timer (tm module) expires. I am wondering if the response code "504 Server Timeout" would be more appropriate in this case. The status code 408 means that "the server could not produce a response, e.g., a user location, within the time indicated in the Expires request-header field. The client MAY repeat the request without modifications at any later time."; and the status code 504 means "the server did not receive a timely response from an external server it accessed in attempting to process the request." It sounds to me that the description of the status code 504 better describes what happens in OpenSER when the fr_timer expires. Any comments? I am using OpenSER 1.2.x. Best regards, Anatoly Pidruchny.

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Re: [OpenSER-Users] OpenSER failure when MySQL redundant database fails
by Henning Westerholt 25 Feb '08

25 Feb '08

On Friday 22 February 2008, Arturo Díaz Almagro wrote: (forward to user list for the archives) > This is my case. We have a SIP telephony system based on OpenSER with a > redundant MySQL database ring, so the OpenSER queries always to the primary > DB unless it fails. When the "master" database dies, a monit process > detects the failure and modify properly the database entry in the DNS zone > file. We have tested this redundancy in two way I expose you here: > > - mysql process is killed in the database host machine. In this case > OpenSER realizes the failure and asks the DNS inmediately for the > database IP address and the service keeps working as expected. No problem > here. > > - shutdown the interface of the machine where mysql is hosted. In this > case OpenSER does not ask the DNS for the new address. It keeps trying > over and over the connection with the master and until the interface is not > turn on the system keeps hung. > > The difference between the two cases is that in the first case, the host > machine responses with an ICMP (port destination unreachable) when OpenSER > queries the database. In the second case we have no answer from the MySQL > host machine. > > So, if the mysql's host interface is shutdown the system hangs but if the > mysql process dies the system works properly. In the first case both > machines (OpenSER and MySQL) has IP visibility, not in the second, so the > problem should be related to that. Do you know why this is happening? Might > you, please, be so kind to tell me how to fix this problem or at list how > to take a workaround to get it working properly? Hi Arturo, Well, this problem is probably caused because the mysql_query function works synchronly, i.e. it blocks the process if no progress on the query could be made. You could easily reproduce this with simple IP filtering with iptables. If you don't get a negative reply to the query, it will simply hang for the TCP timeout from the kernel. We've written in the past a custom extension to the mysql module, called mysql-mt. This elevates this problem by uses a multi-threading approach. But this worked not that great. For recent mysql driver versions this is not needed anymore, i've committed in revision 3606 a improvement to the trunk version of openser that uses the recently introduced extensions in the mysql lib. This could be easily backported to your version. Take a look into the mysql documentation if this different timeouts are supported on your driver version. Cheers, Henning

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[OpenSER-Users] UA Gets Disconnected
by Ali Jawad 25 Feb '08

25 Feb '08

Hi All I am using 1.3 with tls enabled, I am using the default certs that come with the source code and I did disable all the tls_verify options, I can logon to my UA and make calls however there is one problem, every couple minutes the sip client disconnects and re-connects "it does have a re-login option". Any ideas why the UA is getting disconnected every couple of minutes ? Note: I have also tried with UDP without tls and the same thing happened upon monitoring SIP packets I noticed the following, as long as the client is connected it regularly sends REGISTER packets, when the disconnect happens the following happens SIP/2.0 401 Unauthorized Thx With Regards Ali Jawad

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[OpenSER-Users] ERROR:core:_tls_read: something wrong in SSL:
by John Barry 25 Feb '08

25 Feb '08

Hello, I am trying to do a performance test of OpenSER (1.3.0) with TLS using SIPP as the UAS/UAC. The TLS authentication seems to work and the calls are stablished and then terminated (SIPP default scenario). However, I am receiving the following error message from OpenSER: Feb 19 12:58:40 [12709] ERROR:core:_tls_read: something wrong in SSL: Feb 19 12:58:40 [12709] ERROR:core:tcp_read_req: failed to read What do these messages mean? Can I ignored them? The SIP command I am using is: sipp -s 100 -sn uac 192.168.1.10:5061 -t ln -tls_cert 101-cert.pem -tls_key 101-privkey.pem The TLS configuration I am using in my openser.cfg file is: disable_tls = no listen = tls:192.168.1.10:5061 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 1 tls_method = SSLv23 tls_certificate = "/openser/etc/openser/tls/proxy/proxy-cert.pem" tls_private_key = "/openser/etc/openser/tls/proxy/proxy-privkey.pem" tls_ca_list = "/openser/etc/openser/tls/proxy/proxy-calist.pem" I will appreciate any help about this matter. Thanks, JB _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

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Re: [Serusers] Intercept SIP call
by roayu 25 Feb '08

25 Feb '08

Thanks. So, if the client aren't able to perform this function. Can SER do that? Or I need to setup the Asterisk server to help me to perform the 'call pickup' functions. Correct me if I'm wrong. Thanks. Roa Yu -----Original Message----- From: SIP [mailto:sip@arcdiv.com] Sent: Monday, February 25, 2008 11:17 AM To: roayu Cc: samuel; serusers(a)lists.iptel.org Subject: Re: [Serusers] Intercept SIP call When your client registers to SER as a particular user, that registration is stored in SER. If you register multiple times with the same user, ALL of those contacts are called when a call is sent to that user. N. roayu wrote: > > Thanks for your reply and explanation, samuel. > > But what do you mean by "If you want to send the initial call to both > phone B and C, either in parallel or sequentally, then SER can easily > do the job"? Could you please further explain on this or you can > provide me so links to read bout this. > > Thanks! > > Roa Yu. > > ------------------------------------------------------------------------ > > *From:* samuel [mailto:samu60@gmail.com] > *Sent:* Friday, February 22, 2008 4:39 PM > *To:* roayu > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] Intercept SIP call > > The way you have explained the use case would require the phones to do > complex SIP operations. > Phone C need to know the dialog identifiers (dialog info package > probably would help)of the call between A and B and then use replaces > header to "intercept the call". I presonally haven't seen phones doing > this. > > If you want to send the initial call to both phone B and C, either in > parallel or sequentally, then SER can easily do the job. Just read a > few mails or the getting started document and you'll see how to do this. > > Hope it helps, > Samuel. > > 2008/2/20, roayu <roayu(a)ctisys.net <mailto:roayu@ctisys.net>>: > > Hi there! > > Recently, I had setup a SIP environment with few sip phones, SIPphone > A, SIPphone B, and SIPphone C. For instance, when SIPphone A called to > SIPphone B, but B is not able to answer the call. Can SIPphone C > answer A's call? > > I would like to know whether is SER able to have the intercept sip > calls function? If yes, what module or configuration that I need to set? > > Thanks in advance! > > Best regards, > > RoaYu > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Fwd: SIP Express Bundle ready-to-go idea, request for input
by Mike Trest - Personal 25 Feb '08

25 Feb '08

Greger, et. al., I like your idea enough to volunteer as a tester + documentation writer. I do a lot of tech and user writing. Active in VoIP since the beginning. I am active VoIP engineer and developer. Also PSTN & VoIP network designer-builder. I have my own private machines, network, and VoIP gateway resources. Prefer CentOS or FC* ..mike.. >Hi guys, >I have been playing with the following idea: >Create a ready-to-run OS image with everything that is needed for >iptel.org apps pre-installed + a complete installation of: >* SER 2.0 (release) >* rtpproxy >* SEMS >* SERweb >* maybe sipsak, some monitoring tools, etc > > >The idea is to create a small script 'config_iptelorg' for >configuring the installation to your needs. You should then be able >to download the ready image, boot it, go through the script and have >an up and running iptel.org proxy and app server just like the >iptel.org free SIP service in maybe 10-15 minutes. This way you >could host a SIP service for your own domain with close to no setup at all. > >Some questions to you: >* Is there any interest for this at all? >* I was thinking about using Ubuntu 7.10 server as the OS. Any >thoughts/preferences? >* Should the image be an Amazon EC3 image (you could use >http://www.rightscale.com and get it running in no time with 10 >run-hours free) or should it be a VMware appliance to be run with >free VMware Player? >* Other suggestions/comments? > >I could need some help with this, anyone interested in lending me a >hand? (could be anything, documentation, testing, installation, etc) >g-)

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[Serusers] Openser - Ejabberd Integration
by Frank Gonzalez 25 Feb '08

25 Feb '08

Hi, im trying to stablish the real time connection between Openser and Ejabberd both with mysql support, any suggestions ? -- Thanks Frank Gonzalez 414-6260492

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