- sr-users - kamailio.org

[OpenSER-Users] [wiki] Adding page about SIP devices supporting presence PUBLISH
by Iñaki Baz Castillo 09 Oct '07

09 Oct '07

Hi, pua_usrloc has an issue with SIP device that publish their state, since the PUBLISH is generated twice: 1) With the REGISTER by pua_usrloc module. 2) With the PUBLISH sent by the device. If for example the user sets "Online" - "Away" status and after that the device refreshes the REGISTER, then its state will just be "Online" (the state pua_usrloc generates). In order to avoid this issue it could be nice to query a BD table looking for the "User-Agent" and just exec "pua_set_publish()" if the device doesn't support presence PUBLISH. The query could be done by regular expressions in order to allow different versions of same device, all of them supporting presence. A doubt I have is: Would be easier store user-agents of devices supporting presence? or not supporting presence? which one list will be shorter? Because it, I'd like to start a page in the wiki containing all the devices "User-Agent" (with regular expression) supporting (or not) presence PUBLISH, and a small script to generate the table after copy&paste the whole list from the web. What do you think about it? Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[Serusers] Get RealTime Online Users status
by roayu 09 Oct '07

09 Oct '07

Hi there! Can anyone tell me how to get the RealTime Online user status other than using command 'serctl ul show'? Or how can I store the realtime online user status to the MySQL db? Thanks. Cheers, Roa Yu :-)

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Re: Re: R: [OpenSER-Users] Problem with 2 PC behind the same NAT
by Daniel Grotti 09 Oct '07

09 Oct '07

I've just tried and in this way (UA1-------> NAT--------> PROXY---------> UA2), with UA2 in the same net of SIP proxy, Mediaproxy works. El Tuesday 09 October 2007 13:26:19 Daniel Grotti escribió: > I have not configured it explicitely. > This is my opense.cfg file: Please, first try what I said: A call from UA1 (behind your NAT) to other device out of your LAN. MediaProxy (if it well configured) should work in that case. Does it? -- Iñaki Baz Castillo ibc(a)in.ilimit.es _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users <https://mail.unibo.it/exchweb/bin/redir.asp?URL=http://openser.org/cgi-bin/…>

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[OpenSER-Users] CDRTool --Suggestion Required
by srinivas Antarvedi 09 Oct '07

09 Oct '07

Hello users, Actually i want to implement prepaid billing solution to my customers using radius accounting I have gone through the documentation provided with the sources of CDRTool and got confused with all dsn details,connections to mediaproxy ,asterisk,radius,openser. I want to first understand the call flow if using this CDRTool so that i can design mysetup. The present setup i have is 1)openser for authentication 2)Call control using asterisk+asterisk-b2bua in combination with radius Then u may ask why i am after this CDRTool because i was unable to deduct the balance and implement different tariffs using that setup in realtime so i wish to know from you what are different setups i can made using the jumbo CDRTool project so that i can select one and working . First give me suggestion so that i will start with minimal setup -- Srinivas Antarvedi

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[OpenSER-Users] SRV load-balancing not balancing evenly?
by Tim Madorma 09 Oct '07

09 Oct '07

Hi, The OpenSER is performing an SRV lookup which returns the following record info: ;; QUESTION SECTION: ;_sip._udp.aimcidfilter.com. IN SRV ;; ANSWER SECTION: _sip._udp.aimcidfilter.com. 180 IN SRV 1 50 5061 spinner.eng.rr.com. _sip._udp.aimcidfilter.com. 180 IN SRV 1 50 5061 bart.eng.rr.com. _sip._udp.aimcidfilter.com. 180 IN SRV 2 50 5061 homer.eng.rr.com. _sip._udp.aimcidfilter.com. 180 IN SRV 2 50 5061 crunch.eng.rr.com. As you can see, the weighting for the two priority "1" endpoints is 50/50 which indicates that the load should be distributed evenly between them. But when I run my test, it consistently sends 75% to spinner.eng.rr.com and 25% to bart.eng.rr.com. Any ideas? ----------- some config info. debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) maxbuffer=1048576 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #reply_to_via=1 children=4 log_facility=LOG_LOCAL4 dns_use_search_list=no dns_servers_no=2 # user="sipproxy" group="sipproxy" #disable_dns_blacklist=yes # FOR TESTING ONLY #dns_try_ipv6=no # FOR TESTING ONLY listen=udp:65.185.232.62:5060 alias=65.185.233.104:5060 # LOAD OpenSER MODULES mpath="/sw/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) modparam("acc","log_extra", "req_uri=$rU") modparam("tm", "fr_timer", 5) modparam("tm","fr_inv_timer",5) modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")

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[OpenSER-Users] problem with 407 return code
by tloginbr-openser＠yahoo.com.br 09 Oct '07

09 Oct '07

Hi everyone, I'm trying to build one asterisk cluster using openser to route calls to each machine. I followed a few (many) guides and examples on internet but they simple didn't work very well... I'm using the dispatcher module with openser version 1.1.0-9etch1 (running on a debian 4.0). I have two asterisk (trixbox) machines to do load balancing and failover. I can register in my openser machine with my softphone (twinkle). When I place a call to one extension that exists in both asterisks the call goes always to the second ip in the list. Looking in the debug messages I found the following part: 7(7905) DEBUG:tm:t_should_relay_response: T_code=100, new_code=407 7(7905) DEBUG:tm:t_pick_branch: picked branch 0, code 407 7(7905) ERROR:tm:t_forward_nonack: no branch for forwarding 7(7905) ERROR:tm:w_t_relay: t_forward_nonack failed 7(7905) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 7(7905) old size: 555, new size: 474 7(7905) build_res_from_sip_res: copied size: orig:124, new: 43, rest: 431 msg= SIP/2.0 407 Proxy Authentication Required in my understanding that would mean that I need a password of some kind to relay this message to the asterisk box. The funny part is that the call gets completed and goes always to the same box, the second one in the list. If I change the order of the IPs inside my dispatcher.cfg file it starts to go to the other asterisk, that is the 2nd in the list now. After a lot reading in the debug messages and writing some log messages I found out that the invite request is always going down to the failure route. I think that this is because of the call to t_on_failure("1"); inside my if ( method=="INVITE" ) in the main route, that triggers when the code 407 comes from the password reply... thats just one idea. If there is better way to do this (load balance and failover), please tell me, because I'm pretty new to openser, just trying to make it work for the first time here. my openser.cfg has the following lines on it: loadmodule "/usr/lib/openser/modules/dispatcher.so" modparam("tm", "fr_timer", 5) modparam("dispatcher", "flags", 2) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.cfg") if ( method=="INVITE" ) { #dst_select( "GROUP", "HASH METHOD") ds_select_domain("1","4"); t_on_failure("1"); sl_send_reply("100","Trying"); t_relay(); exit(); } failure_route[1] { if(is_method("INVITE")) { ds_next_domain(); t_on_failure("1"); #if the next domain fails too t_relay(); }; } the rest of the file I didn't mess around with it, so its just the default settings. Thanks for any help, Thiago Flickr agora em português. Você clica, todo mundo vê. http://www.flickr.com.br/

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Re:R: [OpenSER-Users] Problem with 2 PC behind the same NAT
by Daniel Grotti 09 Oct '07

09 Oct '07

I have not configured it explicitely. This is my opense.cfg file: --------------------- debug=4 fork=yes log_stderror=yes sip_warning=yes listen=xxx.xxx.xxx port=5060 children=4 dns=no rev_dns=no disable_dns_blacklist=yes mpath="/usr/local/lib/openser/modules/" #------------------------------------------------------ # Load Module #------------------------------------------------------ loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "cpl-c.so" loadmodule "avpops.so" #---------------------------------------------------------- # Parameters #---------------------------------------------------------- modparam("auth_db|uri_db|usrloc|domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("rr", "enable_full_lr", 1) # CPL MODULE modparam("cpl-c", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("cpl-c", "cpl_table", "cpl") modparam("cpl-c", "cpl_dtd_file", "/usr/src/openser-1.2.2-notls/modules/cpl-c/cpl-06.dtd") modparam("cpl-c", "lookup_domain", "location") #NATHELPER MODULE modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("nathelper", "received_avp", "$avp(i:42)") #MEDIAPROXY MODULE modparam("mediaproxy", "natping_interval", 20) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/openser/rtp-asymmetric-clients") #REGISTRAR MODULE modparam("registrar", "received_avp", "$avp(i:42)") modparam("registrar", "default_expires", 1800) #------------------------------------------------------------ #MAIN ROUTE #----------------------------------------------------------- route { #------------------------------------ # Sanity checks #------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too much Hops"); exit; }; if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; }; #------------------------------------ # RECORD-ROUTE #------------------------------------ if(method=="INVITE" && client_nat_test("7")) { record_route_preset ("137.204.107.138:5060;nat=yes"); } else if (!method=="REGISTER") { record_route(); }; #---------------------------------------- # CALL TEAR DOWN #---------------------------------------- if (method=="BYE") { end_media_session(); }; if (method=="CANCEL") { end_media_session(); }; #-------------------------------------------- # LOOSE ROUTE #-------------------------------------------- if (loose_route()) { if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403","Unauthorized"); exit; }; consume_credentials(); if (client_nat_test("7") || search("^Route:.*;nat=yes")) { setbflag(6); # Set the NAT Flag use_media_proxy(); }; }; route(1); exit; }; #------------------------------------------------- #CALL TYPE PROCESSING #------------------------------------------------- if (!uri==myself) { route(4); route(1); exit; }; if (uri==myself) { if (method=="REGISTER") { xlog("L_DBG", "INFO: REGISTER Message received"); route(2); exit; }; if (method=="ACK") { route(1); exit; }; if (method=="INVITE") { route(3); exit; }; lookup("aliases"); if (!uri==myself) { route(4); route(1); exit; }; #Find the Destination - Check the R-URI if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; }; route(1); } #---------------------------------------------------- # DEFAULT MESSAGE HANDLER #---------------------------------------------------- route[1] { t_on_reply("1"); #Passa il controlla alla route on reply 1 if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } #-------------------------------------------------------- # REGISTER PROCESSING #-------------------------------------------------------- route[2] { if(!www_authorize("","subscriber")) { www_challenge("","0"); exit; } else if(!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); #Set NAT Flag in USRLOC. xlog("L_DBG","NOTICE: Bflag 6, related to BRANCH=$T_branch_idx, is setted as $bf"); fix_nated_register(); force_rport(); xlog("L_DBG","INFO: Registration behind nat"); }; sl_send_reply("100", "Trying"); if(!save("location")) { sl_reply_error(); xlog("L_DBG", "Registration of [$fU] Failed"); }; xlog("L_DBG", "INFO: Registration of [$fU] Complete."); xlog("L_DBG", "INFO: 200 OK has been sended"); } #---------------------------------------------------------- # INVITE PROCESSING #---------------------------------------------------------- route[3] { if(client_nat_test("7")) { setbflag(7); xlog("L_DBG","NOTICE: Bflag 7, related to BRANCH=$T_branch_idx of this INVITE, is setted as $bf"); if(force_rport()) { xlog("L_DBG","NOTICE: rport FORCED"); }; #Rewrites Contact HF to contain nated request's source address:port. if(fix_nated_contact()) { xlog("L_DBG","NOTICE: Contact HF is now [$ct]"); }; }; if(!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Unauthorized"); exit; }; consume_credentials(); lookup("aliases"); if(uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; if(!cpl_run_script("incoming", "force_stateful")) { t_reply("500", "CPL script execution failed"); }; route(4); route(1); } #-------------------------------------------------------------------- # NAT Traversal #-------------------------------------------------------------------- route[4] { # Se il chiamato (flag 6) o il chiamante (flag 7) Ã¨ sotto nat if (isbflagset(6) || isbflagset(7)) { # if(!isbflagset(8)) # { setbflag(8); xlog("L_DBG", "NOTICE: flag 6 or 7 setted"); if(!use_media_proxy()) { xlog("L_DBG","NOTICE: mediaproxy wasn't called!!"); }; # }; }; } #--------------------------------------------------------------------- # On Reply Route #--------------------------------------------------------------------- onreply_route[1] { if ((isbflagset(6) || isbflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Lenght:[ ]*0")) { if (use_media_proxy()) { xlog("L_DBG","NOTICE: mediaproxy in ON REPLY"); }; }; }; if (client_nat_test("7")) { fix_nated_contact(); }; } ---------------------------------------------------------------------- Inviato: mar 10/9/2007 12:28 A: users(a)openser.org Oggetto: Re: [OpenSER-Users] Problem with 2 PC behind the same NAT El Tuesday 09 October 2007 11:51:34 Daniel Grotti escribió: > I have 2 SJphone behind the same NAT (I know that it's a restricted con > NAT). > > > UA1 ----> (NAT 1) --------> PROXY > > > UA2 <-----(NAT 1)<------------- > > I've configured my openser.cfg to use Mediaproxy (installed in the same > pc) if 1 or 2 pc are behind NAT (in this case both of my pc). > But when I call from UA1 to the other and I accept the call with UA2, > the RTP stream doesn't flow through the Mediaproxy but it's direct from > UA1 to UA2. Have you tryied to call from UA1 (behind your NAT) to other device out of your LAN? MediaProxy (if it well configured) should work in that case. In case caller and called behind the same call, you explicitely **need** to allow direct RP in openser.cfg by matching $si with $dd (if pulbic source address and destination address is the same don't apply SDP fix). But if you haven't configured it explicitely then maybe your conf is worng. -- Iñaki Baz Castillo ibc(a)in.ilimit.es _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] subst_body missing /sm flags
by Aki Rehn 09 Oct '07

09 Oct '07

Hi, I need some help. Here's my input in onreply_route: --- 8< --- U 2007/10/05 12:26:58.832161 xxx.18.234.14:5060 -> xxx.18.238.2:5060 SIP/2.0 180 Ringing. From: "aki"<sip:358942419xxx@devel.foo.fi>;tag=as508b1a29. To: <sip:3584577509xxx@vox.bar.fi>;tag=eea1259-13c4-47060394-b296ee66-575d. Call-ID: 7b18b07b06335629699f1d5818e07548(a)devel.foo.fi. CSeq: 102 INVITE. Via: SIP/2.0/UDP xxx.18.238.2;branch=z9hG4bK570f.7ca86953.0. Via: SIP/2.0/UDP xxx.18.238.15:5060;branch=z9hG4bK68f201e6. Contact: <sip:3584577509xxx@xxx.18.234.14>. Record-Route: <sip:xxx.18.238.2;lr;ftag=as508b1a29>. Content-Type: multipart/mixed ;boundary="Boundary47060397_b296f9ef_3ecc_bc780ec". Content-Length: 397. . --Boundary47060397_b296f9ef_3ecc_bc780ec. Content-Type: application/SDP. . v=0. o=xxx.18.234.23 1191575887 1191575887 IN IP4 xxx.18.234.23. s=Session SDP. c=IN IP4 xxx.18.234.23. t=0 0. m=audio 11130 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=ptime:20. . --Boundary47060397_b296f9ef_3ecc_bc780ec. Content-Type: application/ISUP ;version=itu-t. . ,.....4)..9.7.7.... --Boundary47060397_b296f9ef_3ecc_bc780ec-- --- >8 --- These extra Conten-Type (application/ISUP and multipart/mixed) are messing up some older PBX's of ours, so I need to strip them out. With perl I could do it like this: s/^Content-Type: application\/ISUP ;version=itu-t.*--Boundary/--Boundary/sm; s/--Boundary.*$//mg; However the subst_body() doesnt seem to support /sm flags. Discussing this on #openser @ freenode we tried solutions with transformations (thanks NormB!) like: $(hdr(Content-Type)) = $(hdr(Content-Type){s.substr,0,0}) .. or like the same with replace* functions. However I didnt find a suitable solutions. Anyone maybe got any ideas? Br, Aki

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[OpenSER-Users] DNS blacklisting problem
by Tim Madorma 09 Oct '07

09 Oct '07

Hi, I'm interested in using DNS blacklisting to stop transmitting SIP requests towards UAS endpoints that are non-responsive. I am on version 1.2, but I can't get it to work properly. Here is what I am trying to do: 1) I get a message from a UAC that needs to be sent towards 1 of 4 destination UAS endpoints. 2) I setup the OpenSER to perform an SRV lookup which returns a record that has four A records associated with it. It looks as follows: ;; QUESTION SECTION: ;_sip._udp.aimcidfilter.com. IN SRV ;; ANSWER SECTION: _sip._udp.aimcidfilter.com. 60 IN SRV 1 100 5061 spinner.eng.rr.com. _sip._udp.aimcidfilter.com. 60 IN SRV 2 100 5061 bart.eng.rr.com. _sip._udp.aimcidfilter.com. 60 IN SRV 3 50 5061 homer.eng.rr.com. _sip._udp.aimcidfilter.com. 60 IN SRV 3 50 5061 crunch.eng.rr.com. 3) The OpenSER does a gethostbyname which performs a DNS A record lookup and gets a response for spinner.eng.rr.com. 4) When the OpenSER sends towards spinner.eng.rr.com, it does not get a SIP response from it. *It is at this point I was hoping that OpenSER would blacklist this UAS endpoint*, but for each incoming request that is received by the OpenSER, it continues to resolve the domain in the same manner and sends towards spinner.eng.rr.com even though there is no SIP response. Is what I am trying to do in accordance with how OpenSER blacklisting is supposed to work? I am trying to have OpenSER send towards bart.eng.rr.com when spinner.eng.rr.com is not responding or is not reachable. My configuration is as below. Note that I have even tried to disable sending towards spinner.eng.rr.com by manually adding a dst_blacklist entry - that doesn't even seem to work for me.. Am I doing something wrong? I have also attached a log file. Here is my output from openserctl # openserctl fifo list_blacklists 200 OK List:: net_filter owner=13 flags=1 Rule:: flags=0 IP:: 65.185.233.55 Mask:: 255.255.255.255 Proto:: 0 Port:: 5061 List:: dns owner=17 flags=6 --------------- # # Openser.cfg # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) maxbuffer=1048576 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #reply_to_via=1 children=4 log_facility=LOG_LOCAL4 dns_use_search_list=no dns_servers_no=2 user="sipproxy" group="sipproxy" disable_dns_blacklist=no disable_dns_failover=no dns_try_ipv6=no dns_retr_time=2 dns_retr_no=2 dst_blacklist = net_filter:{ ( any , 65.185.233.55, 5061 , "" )} # block towards spinner listen=udp:65.185.232.62:5060 alias=65.185.233.104:5060 # LOAD OpenSER MODULES mpath="/sw/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) modparam("acc","log_extra", "req_uri=$rU") modparam("tm", "fr_timer", 5) modparam("tm","fr_inv_timer",5) modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { route(1); }; if (method=="OPTIONS") { sl_send_reply("200", "OK"); exit; } if (uri =~ "^sip:feature_fs@" || method=="NOTIFY") { xlog("L_DBG", "TWC: received incoming message:\n <$mb>\n"); seturi("sip:feature_fs@aimcidfilter.com"); setflag(1); route(1); } else { sl_send_reply("404", "Not Found"); }; } ##################################################### # Default Message Handler ##################################################### route[1] { # Send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

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[OpenSER-Users] Problem with 2 PC behind the same NAT
by Daniel Grotti 09 Oct '07

09 Oct '07

I have 2 SJphone behind the same NAT (I know that it's a restricted con NAT). UA1 ----> (NAT 1) --------> PROXY | | UA2 <-----(NAT 1)<------------- I've configured my openser.cfg to use Mediaproxy (installed in the same pc) if 1 or 2 pc are behind NAT (in this case both of my pc). But when I call from UA1 to the other and I accept the call with UA2, the RTP stream doesn't flow through the Mediaproxy but it's direct from UA1 to UA2. Any clue? thanks daniel

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Re: [Serusers] Get RealTime Online Users status
by Jiri Kuthan 09 Oct '07

09 Oct '07

If you mean location database, check the "location" table then. -jiri At 10:05 09/10/2007, roayu wrote: >Hi there! > >Can anyone tell me how to get the RealTime Online user status other than using command �serctl ul show�? Or how can I store the realtime online user status to the MySQL db? > > >Thanks. > >Cheers, >Roa Yu J >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[OpenSER-Users] The role of "rinstance"
by Daniel Grotti 09 Oct '07

09 Oct '07

Hi everybody, Somebody Could tell me which role the Rinstance tag play in the SIP message, when I use X-lite 3 ? Thank you, Daniel

2 1

[OpenSER-Users] Can't Register SJPhone with IMS Core
by A Chaturvedi 09 Oct '07

09 Oct '07

Hi, I tried to register SJPhone with IMS Core but it failed (even after using sip2ims gatway). A part from this when I used XLite sip phone it gets registered... Please guide me if pcscf need some modification to accept the Register request from it... Also find ate ethereal dump for it.. regards: Ashish Chaturvedi Mailto: a.chaturvedi(a)tcs.com =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[OpenSER-Users] MI_XMLPRC supports new versions of libxmlrpc-c3
by Bogdan-Andrei Iancu 09 Oct '07

09 Oct '07

Hi, I know this was from the beginning a hot issue, but now, thanks to Dragos Oancea <dragos(a)voice-system.ro>, the mi_xmlrpc can: 1) be compiled against the newest versions of libxmlrpc-c3. The Makefile will detect the library version an trigger the appropriate compile options for the module; So the module supports both the old version (0.9.10) and the newest versions (1.03.10 - 1.06.18). If none of these version is detected, a warning will be issued and the module will be compiled as for the new versions. If you detect the older versions (than 1.03.10 are supported, please let us know). 2) properly auto-detects the path of the header files and the libraries that the modules needs to compile against, so this should not be a problem anymore ;) any feedback will be useful :) regards, bogdan

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[OpenSER-Users] Catching 200 OK as part of INVITE CSeq.
by Alex Balashov 09 Oct '07

09 Oct '07

Greetings, I am proxying INVITEs and need to be able to know when the 200 OK that is part of that particular transaction dialogue (same CSeq as the INVITE) appears so that I stamp voice-bearing media start in my logs. However, I cannot seem to detect it. I see from the packet capture that the 200 OK from the far end of the call does go back through my OpenSER proxy (I do record_route() on all methods but REGISTER & OPTIONS as per the spec), but I can't seem to programmatically access it in any way. Are there special steps required to get at it? This is to say, my main route contains: if(is_method("INVITE")) { ... } But I never seem to catch anything with it but actual INVITE messages. What gives? I tried doing if(t_check_status("200")) within that block, but without anything that I would describe as success. I never even get another message whose overall transaction method is INVITE even when I dump them all for debugging purposes in the main route. Insights appreciated! Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[OpenSER-Users] Direct (Postgres) SQL queries.
by Alex Balashov 09 Oct '07

09 Oct '07

Is there a better way to talk to a database directly than to use avp_db_query()? I need to use my own schemas; I have a platform that absolutely does not lend itself to storing data in the avpops' schema. Luckily, avp_db_query() saved the day. Is there anything even easier, or is this the definitive and methodologically correct way to go? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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[OpenSER-Users] Urgent!! Facing Problem while Registering SJPhone
by A Chaturvedi 09 Oct '07

09 Oct '07

Hi I was trying to register SJPhone with IMS Core and at Client end it shows Registered but at HSS it shows AUTH-PENDING. The Ethereal dump shows proper call flow with 200 OK. Have a look on HSS console debug statements... at de.fhg.fokus.diameter.DiameterPeer.transport.Communicator.run(Communicator.java:187) 2007-10-09 12:43:26,340 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:26,389 DEBUG de.fhg.fokus.hss.main.Task - execute Processing MAR! 2007-10-09 12:43:26,410 DEBUG de.fhg.fokus.hss.cx.op.MAR - generateAuthVector Auth-Scheme is Digest-MD5 2007-10-09 12:43:26,418 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:26,462 DEBUG de.fhg.fokus.hss.main.Task - execute Processing SAR! 2007-10-09 12:43:36,460 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:36,508 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:36,555 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! 2007-10-09 12:43:36,620 DEBUG de.fhg.fokus.hss.main.Task - execute Processing MAR! 2007-10-09 12:43:36,658 DEBUG de.fhg.fokus.hss.cx.op.MAR - generateAuthVector Auth-Scheme is Digest-MD5 2007-10-09 12:43:36,703 DEBUG de.fhg.fokus.hss.main.Task - execute Processing UAR! de.fhg.fokus.hss.cx.CxExperimentalResultException: IMS_Diameter_Error_User_Unknown at de.fhg.fokus.hss.cx.op.UAR.processRequest(UAR.java:112) at de.fhg.fokus.hss.main.Task.execute(Task.java:169) at de.fhg.fokus.hss.main.Worker.run(Worker.java:66) Please help me out in this.... If any body has made a successful call with SJPhone and IMS , plzz send me the configuration setting for that... Thanks in adv.... Ashish =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[OpenSER-Users] Snom number-guessing and OpenSER
by Edoardo Serra 09 Oct '07

09 Oct '07

Hi all, I'm trying to develop an extension to support SNOM's number guessing feature (I don't know if it's somehow a standard but description on how it works is here: http://kb.snom.com/kb/index.php?View=entry&CategoryID=14&EntryID=14) As I have to do some 'complicated' db queries I was thinking to use PERL module to handle number guessing request from the phone. I see there's no way (please correct me if I'm wrong) from OpenSER, neither from OpenSER::Message, to send a stateless reply with a customized body (SNOM says that possible number completion shoud be included in the body of a 200 OK reply). For this reason I have to use Net::SIP to generate the reply packet but I'm doing something wrong as OpenSER is not forwarding it back to the client. Test client is a Snom 360 and is behind a NAT, OpenSER has a public IP Here is my routing logic in openser.cfg ------------------- BEGIN openser.cfg route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (nat_uac_test("19")) { fix_nated_contact(); force_rport(); setbflag(6); } if (method=="PUBLISH") { if ($hdr(Event)=="number-guessing") { log("- Initial PUBLISH\n"); t_newtran(); t_on_reply("1"); perl_exec("numberguessing"); exit; } sl_send_reply("501", "Not implemented"); exit; } if (method=="REGISTER") { if (nat_uac_test("19")) { fix_nated_contact(); force_rport(); setbflag(6); } save("location"); exit; }; exit; } route[1] { log("--------------------- Stateful Reply\n"); if (!t_relay()) { sl_reply_error(); }; exit; } ------------------- END openser.cfg Here is my Simple Perl ------------------- BEGIN net-sip.pl use OpenSER qw ( log ); use OpenSER::Constants; use Net::SIP; # Predefined body, not querying db yet my $body = '"Franky Chang" sip:101@192.168.0.1;user=phone "Steven Jones" sip:102@192.168.0.1;user=phone "Marie Sun" sip:103@192.168.0.1;user=phone '; sub numberguessing { my $m = shift; my $pkt = Net::SIP::Request->new( $m->getMessage() ); my $resp = $pkt->create_response(200, "OK"); $resp->set_body($body); $resp->add_header('Content-Type','application/number-guessing'); my $leg = Net::SIP::Leg->new( addr => '111.222.333.444', port => '5061'); $leg->deliver( $resp, '111.222.333.444:5060' ); # 111.222.333.444 is the IP of OpenSER # I'm sending from port 5061 to port 5060 return 1; } ------------------- END net-sip.pl OpenSER in verbose debugging module says that it cannot find a matching transaction for the 200 OK reply (probably because is originated from port 5061 and it has never forwarded it to that port ? ) Here is OpenSER DEBUG log: 0(24828) SIP Reply (status): 0(24828) version: <SIP/2.0> 0(24828) status: <200> 0(24828) reason: <OK> 0(24828) parse_headers: flags=2 0(24828) get_hdr_field: cseq <cseq>: <1> <PUBLISH> 0(24828) DEBUG:parse_to:end of header reached, state=10 0(24828) DBUG:parse_to: display={"Edoardo"}, ruri={sip:eserra@111.222.333.444} 0(24828) DEBUG: get_hdr_field: <to> [43]; uri=[sip:eserra@111.222.333.444] 0(24828) DEBUG: to body ["Edoardo" <sip:eserra@111.222.333.444> ] 0(24828) Found param type 232, <branch> = <z9hG4bK-m3j054b6d8q6>; state=6 0(24828) Found param type 235, <rport> = <n/a>; state=17 0(24828) end of header reached, state=5 0(24828) parse_headers: Via found, flags=2 0(24828) parse_headers: this is the first via 0(24828) After parse_msg... 0(24828) DEBUG:forward_reply: found module tm, passing reply to it 0(24828) DEBUG: t_check: start=0xffffffff 0(24828) parse_headers: flags=22 0(24828) DEBUG: t_reply_matching: failure to match a transaction 0(24828) DEBUG: t_check: end=(nil) 0(24828) parse_headers: flags=4 0(24828) DEBUG: get_hdr_body : content_length=135 0(24828) found end of header 0(24828) ERROR:forward_reply: no 2nd via found in reply 0(24828) DEBUG:destroy_avp_list: destroying list (nil) 0(24828) receive_msg: cleaning up Here is also the tshark dump of the 2 packets (PUBLISH and its reply) Request-Line: PUBLISH sip:eserra@213.92.23.108 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.254.151:2051;branch=z9hG4bK-ma9r5n0t4jfq;rport From: "Edoardo" <sip:eserra@111.222.333.444>;tag=39wlni6ex1 To: "Edoardo" <sip:eserra@111.222.333.444> Call-ID: 3c27c039249b-jml9w7t9sywx CSeq: 1 PUBLISH Max-Forwards: 70 Event: number-guessing Content-Type: application/text Content-Length: 25 Message body Number: 10\r\n Max-Hits: 3\r\n Status-Line: SIP/2.0 200 OK Message Header call-id: 3c27c0daa600-emnlo1xcejtn cseq: 1 PUBLISH from: "Edoardo" <sip:eserra@111.222.333.444>;tag=6c1y8pp3fb to: "Edoardo" <sip:eserra@111.222.333.444> via: SIP/2.0/UDP 192.168.254.151:2051;branch=z9hG4bK-623oddiyzg56;rport content-type: application/number-guessing Content-Length: 135 Message body "Franky Chang" sip:101@192.168.0.1;user=phone\n "Steven Jones" sip:102@192.168.0.1;user=phone\n "Marie Sun" sip:103@192.168.0.1;user=phone\n Hope this info is enough for a debug :) Tnx in advance Regards Edoardo Serra EXSORSA LLC

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[OpenSER-Users] Mailing List Archives
by Norman Brandinger 08 Oct '07

08 Oct '07

http://dir.gmane.org/gmane.comp.voip.openser.devel http://www.mail-archive.com/devel@openser.org http://www.nabble.com/OpenSER-Developers-Mailing-List-f26016.html http://dir.gmane.org/gmane.comp.voip.openser.user http://www.mail-archive.com/users@openser.org http://www.nabble.com/OpenSER-Users-Mailing-List-f26015.html http://www.nabble.com/OpenSER-Users---ES-f26090.html http://www.nabble.com/OpenSER-Business-f26089.html

1 0

[OpenSER-Users] questions about 1.2
by Cseke Tamas 08 Oct '07

08 Oct '07

Hello, We are switching from 1.1 to 1.2, and i have some problems. 1) Could you please tell me how should i handle natted contacts in regisers. i read modparam("registrar", "nat_flag", 6) is not exists any more i guess i should use nat_bfalg in usrloc module, but i don't know how to use this, could you tell me an example? how should i set the branch flag which should be used for this? 2) There is no Calling-Station-Id and Called-Station-Id in radius accounting packets. Although i use the same libradiusclient_ng.so, with the same configuration files, and the same radius server too. I need these attributes, could you please tell me why they are missing? Thanks any help, Tamas

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[OpenSER-Users] How to decode "methods" field in "location" table?
by Iñaki Baz Castillo 08 Oct '07

08 Oct '07

Hi, in case I want to detect which methods allow a registered device I think I should examinate the "methods" field in "location" table, but don't know how to decode it. For example: - 7999 means: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE - 5183 means: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER How could I know if a device allow X method? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[OpenSER-Users] SVN freeze for 1.3 release
by Bogdan-Andrei Iancu 08 Oct '07

08 Oct '07

Hi everybody, According to the plans for 1.3 release, starting with now, the SVN is freeze - no new code or functionalities will be added. The next 2 months are fully allocated to testing & preparing the 1.3 release. Only allowed commits will be for: - bug fixes - warning fixes - comments/logs fixes/beautification - documentation updates - packaging and tools updates I want to thanks to everybody who contributed (with work, code, supports, etc) to the new 1.3 release - it is a really good work ;). Regards, Bogdan

1 0

Re: [OpenSER-Users] Problem with otg proxy when service provider has two registrars
by Robert Dyck 08 Oct '07

08 Oct '07

I had already tried configuring the UA with the address of one of the servers ( both IP and domain name ) as well as altering the openser config to force the address. The peculiar thing there is that the registrar does not challenge or even respond at all. It would seem that it ignores REGISTER requests that do not have callcentric.com as the domain name and realm. The UA can register with this provider without difficulty when the UA is configured to use STUN and no outgoing proxy. The UA does not do a second DNS lookup. It simply uses the same address for both requests. When the UA receives the challenge does it not use the received nonce to encrypt its credentials? I have to admit my knowledge of that subject is shakey. And would this not have to be delivered to the same server that sent the nonce? On Saturday 06 October 2007, you wrote: > Robert Dyck wrote: > > I am more familiar with ethereal. I hope that is OK. Also I have not > > edited the dumps so I am sending them privately. Attached are brief and > > detailed dumps from ethereal. > > Your SIP provider is using DNS round-robin which is why openser is > forwarding the requests to different IP addresses. This is the first > provider I see that is doing DNS RR, this is rather unusual and not what > is described by the SIP RFCs. > > Nevertheless, I still believe that your problem is related to wrong > credentials. Both provider registrars should accept your REGISTER with > Proxy-Auth header. > > You could also configure your SIP client with 204.11.192.22 instead of > the provider's hostname, this will disable DNS RR and let openser > forward the request always to the same host. > > /Christian > > > On Saturday 06 October 2007, you wrote: > >> Robert Dyck wrote: > >>> The second registrar does not send an error code, it simply issues its > >>> own challenge. Openser is definitely alternating between registrars. It > >>> does not send the credentials to the same registrar that requested > >>> them. > >>> > >>> I could send a trace if it would be helpful. > >> > >> Yes, that would be helpful, I'd also like to have a look at the DNS > >> traffic. Can you do > >> > >> tcpdump -i any -s 1500 -w /tmp/trace.pcap > >> > >> /Christian

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[OpenSER-Users] Problem with otg proxy when service provider has two registrars
by Robert Dyck 06 Oct '07

06 Oct '07

I am experimenting with using openser as the basis for a home telephone system. Openser resides on the same node as my NAT. I have also employed the rtp proxy feature. I have had some success with my setup but recently I encountered a problem registering with a particular service provider. My phones are set up to use openser as an outgoing proxy. They can register with various service providers except one. From tracing the messages I believe I know what is going wrong. The REGISTER is sent to openser which does a DNS lookup. This results in two registrars. Openser picks one and forwards the messages. The registrar issues a challenge as expected which is sent back to the phone. The phone sends its credentials to openser. Openser does another DNS lookup and sends the message to the other registrar. This registrar also issues a challenge. Thus it continues forever with the credentials being sent to the wrong registrar. My question is, how do I get openser to send the credentials to the registrar that is requesting them? Thanks in advance for any advice.

2 2

[OpenSER-Users] Stateful matching of all messages within single call leg.
by Alex Balashov 05 Oct '07

05 Oct '07

Also, while I appreciate the 'tm' module's ability to catch replies to messages that are part of the same transaction, i.e. t_check_trans(), but does OpenSER provide a way natively to check if messages associated with *any* transaction are part of a particular logical call leg (i.e. dialogue branch tag, Call-ID GUID)? Right now, I am having to hit my database of stored open Call-IDs via avpops to see if this is the case, and that seems ludicruously expensive. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

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