[SR-Users] TLS connection with short lifetime - how to handle it?
Greetings. I have the next problem: iOS based clients connect via TLS to kamailio server. They run mostly in background mode - it means connection refresh interval is ~10 minutes. Some of clients reside behind paranoidal routers which considers such idle connections as lost and closes them. I see only way to resolve it is to send OPTIONS/NOTIFY/CRLF from kamailio to clients. Tried to use nat_traversal module by calling nat_keepalive() to every REGISTER message. But there is no incoming keepalive (OPTIONS or NOTIFY) on client side. Is there another way to make heartbeat in SIP/TLS from server side? Thank you. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
Hello, Thanks Juha, that will be a good thing to investigate more when I get my simple unrealistic scenario working. :) I tried compiling rtpengine on Centos 6.5, I wonder do I need to change the Makefile somehow for CentOs? Remove Debian specific flags like mentioned in the github page? Below is the output from the make: In file included from call.c:25: ../kernel-module/xt_MEDIAPROXY.h:23: warning: declaration does not declare anything call.c: In function 'kernelize': call.c:342: error: 'struct mp_address' has no member named 'ipv4' call.c:344: error: 'struct mp_address' has no member named 'ipv4' call.c:348: error: 'struct mp_address' has no member named 'ipv6' call.c:348: error: 'struct mp_address' has no member named 'ipv6' call.c:350: error: 'struct mp_address' has no member named 'ipv6' call.c:350: error: 'struct mp_address' has no member named 'ipv6' call.c: In function 'call_destroy': call.c:1961: warning: format '%lu' expects type 'long unsigned int', but argument 8 has type 'u_int64_t' call.c:1961: warning: format '%lu' expects type 'long unsigned int', but argument 9 has type 'u_int64_t' call.c:1961: warning: format '%lu' expects type 'long unsigned int', but argument 10 has type 'u_int64_t' make[1]: *** [call.o] Error 1 make[1]: Leaving directory `/usr/local/src/rtpengine/daemon' make: *** [all] Error 2 cheers, Olli 2014-04-06 21:58 GMT+03:00 Juha Heinanen j...@tutpro.com: Olli Heiskanen writes: Thanks, I'll look into the rtpengine, had a busy weekend but next week I'll have better time. what comes to peter's slideshare failure_route example, i think it only works in very simple unrealistic scenario when there is no forking or serial routing. also, its nathelper handling is unnecessary when websocket sip ua, such as jssip, supports gruu. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Show Domain-Name in Siremis
Hello, which page are you looking at? Cheers, Daniel On 08/04/14 06:59, aawaise wrote: Hello, I am running kamailio server in my set up. And have got siremis working for monitoring purposes. Problem is that in web interface, domain name of users is not shown in siremis. If a user is online, I get following credentials infront of Client column, Client.Name@Client's.IP How can I show domain name in siremis ?? Any help will be highly appreciated. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp126602.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio / rtpproxy - remove codec
Hello, you are using it with some rtp relay function. Remove the codec, then do msg_apply_changes() (all this before doing record route), then do the rtp relaying. Cheers, Daniel On 07/04/14 12:01, Oliver Roth wrote: Here we go IN INVITE sip:41442742931@81.7.235.180:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 82.197.185.185;branch=z9hG4bKa368.02f63ac6.0 Via: SIP/2.0/UDP 195.216.67.103:5060;branch=z9hG4bK008082590C36C1313E749FCBA4C6 Route: sip:82.197.185.186;lr=on From: sip:+41446512001@195.216.67.103;user=phone;tag=008082590C36C1313BF6EF12E7F2 To: sip:+41442742931@82.197.185.185;user=phone;tag=snl_0014532334 Call-ID: e43880005082-533eb654-10953444-55d4a80-6bc5dd@127.0.0.1 CSeq: 5466 INVITE Contact: sip:+41446512001@195.216.67.103:5060 Max-Forwards: 69 Content-Type: application/sdp Session-Expires: 1800;refresher=uas Supported: 100rel, timer, replaces Content-Length: 330 v=0 o=- 82545395 2 IN IP4 195.216.67.103 s=session t=0 0 m=audio 4550 RTP/AVP 8 0 c=IN IP4 195.216.67.120 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv m=image 4552 udptl t38 c=IN IP4 195.216.67.120 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF OUT INVITE sip:41442742931@81.7.235.180:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 82.197.185.186;branch=z9hG4bKa368.aca64b5.0 Via: SIP/2.0/UDP 82.197.185.185;branch=z9hG4bKa368.02f63ac6.0 Via: SIP/2.0/UDP 195.216.67.103:5060;branch=z9hG4bK008082590C36C1313E749FCBA4C6 From: sip:+41446512001@195.216.67.103;user=phone;tag=008082590C36C1313BF6EF12E7F2 To: sip:+41442742931@82.197.185.185;user=phone;tag=snl_0014532334 Call-ID: e43880005082-533eb654-10953444-55d4a80-6bc5dd@127.0.0.1 CSeq: 5466 INVITE Contact: sip:+41446512001@195.216.67.103:5060 Max-Forwards: 68 Content-Type: application/sdp Session-Expires: 1800;refresher=uas Supported: 100rel, timer, replaces Content-Length: 188 v=0 o=- 82545395 2 IN IP4 82.197.185.186 s=session t=0 0 m=audio 64748 RTP/AVP 8 0 c=IN IP4 82.197.185.186 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv 5037682.197.185.186 *Von:*sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] *Im Auftrag von *Daniel-Constantin Mierla *Gesendet:* Montag, 7. April 2014 10:40 *An:* Kamailio (SER) - Users Mailing List *Betreff:* Re: [SR-Users] kamailio / rtpproxy - remove codec Hello, can you give the incoming sdp and outgoing sdp (both as text) when you are using sdp_remove_media, to see what gets malformed there? Cheers, Daniel On 07/04/14 08:37, Oliver Roth wrote: Hi all We use kamailio 3.3.7 and rtpproxy for enduser call-termination. In case of a fax call, we get an invite from our carrier for codecs G711a/u and T38. As our termination carrier does not support T38 and because the invite contains G711 and T38 we get back error 488. How is it possible to remove the whole T38 part of this invite? We tried sdp_remove_codecs_by_name(list) without success – what “name” should we use for T38? [T38, t38, t.38, T.38, …] sdp_remove_line_by_prefix(string) sdp_remove_media(type) None of these functions did really work – best was the last one with type=image but then the sip header is malformed. As we saw with Kamailio version 4.1.x there are a lot of new functions within sdpops. Would an upgrade help? So basically the question is: How to remove the t38 part of the fax invite? (see attachment) KR, Oli ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - April 2-4, 2014, Berlin, Germany http://www.kamailioworld.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] sql_xquery() and xavp checks
On Saturday 05 April 2014 17:32:18 Alex Balashov wrote: When using sql_xquery() like this: sql_xquery(ca, SELECT * FROM gateways, gateways); ... what's a good way to check if any rows were returned? Since one does not have a $dbr(gateways=rows) value in this scenario, what should one do? All sqlops query functions have (undocumented) return values: -1: error in parameters or query execution 1: query successful, at least one row in resultset (for SELECTs) 2: query successful, no rows returned It might be useful to extend $sqlrows() to return the number of rows in the resultset. -- Alex Hermann ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] cnxcc
i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Show Domain-Name in Siremis
I am looking at the LOCATION LIST under SER MONITOR MENU. I am looking at the online users. The user's IP adress can been seen the table infront of Contact but there is nothing present infront of Domain. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp126602p126613.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cnxcc
Hi Kelvin, probably the dialog matching parameter is wrong, or something in the parameter configuration of cnxcc. The value of max_amount is 0, because the call wasn't established or was established and cnxcc couldn't detect it (probably because of the dlg matching, as I mentioned). It will retain this value for as long as the call is in the early state. Check this example [1], it may help you. [1] http://caruizdiaz.com/2014/04/04/cnxcc-prepaid-module-workshop-kamailio-world/ Regards, On Tue, Apr 8, 2014 at 3:04 AM, Kelvin Chua kel...@gmail.com wrote: i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cnxcc
Hi Carlos, please remember: You're on your Honeymoon! :-) Enjoy Rome, Carsten 2014-04-08 10:26 GMT+02:00 Carlos Ruiz Díaz carlos.ruizd...@gmail.com: Hi Kelvin, probably the dialog matching parameter is wrong, or something in the parameter configuration of cnxcc. The value of max_amount is 0, because the call wasn't established or was established and cnxcc couldn't detect it (probably because of the dlg matching, as I mentioned). It will retain this value for as long as the call is in the early state. Check this example [1], it may help you. [1] http://caruizdiaz.com/2014/04/04/cnxcc-prepaid-module-workshop-kamailio-world/ Regards, On Tue, Apr 8, 2014 at 3:04 AM, Kelvin Chua kel...@gmail.com wrote: i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cnxcc
Hi Carsten, that's right, I'm replying to this while I wait for a tour to the Vatican City :-D ;-) Cheers, On Tue, Apr 8, 2014 at 3:27 AM, Carsten Bock cars...@ng-voice.com wrote: Hi Carlos, please remember: You're on your Honeymoon! :-) Enjoy Rome, Carsten 2014-04-08 10:26 GMT+02:00 Carlos Ruiz Díaz carlos.ruizd...@gmail.com: Hi Kelvin, probably the dialog matching parameter is wrong, or something in the parameter configuration of cnxcc. The value of max_amount is 0, because the call wasn't established or was established and cnxcc couldn't detect it (probably because of the dlg matching, as I mentioned). It will retain this value for as long as the call is in the early state. Check this example [1], it may help you. [1] http://caruizdiaz.com/2014/04/04/cnxcc-prepaid-module-workshop-kamailio-world/ Regards, On Tue, Apr 8, 2014 at 3:04 AM, Kelvin Chua kel...@gmail.com wrote: i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dialog: Keep-alive issue:
Thanks Kelvin, but I already have tried this. This is exactly what I wrote that the issue is: Despite having set those options, and the keep alives are sent, the dialog does not timeout as it should when no reply is sent to the keep-alive OPTIONS msg by the dst peer. I am using the master branch and I am trying to follow the source code to see how and where the dialog module should be called back when the timeout for the OPTIONS message occurs, but I am a little lost. If you think I am missing something and there is a simple configuration solution please do point it to me. Thanks a lot. On Mon, Apr 7, 2014 at 10:02 PM, Kelvin Chua kel...@gmail.com wrote: try this modparam(dialog, ka_timer, 3) modparam(dialog, ka_interval, 10) these 2 will make sure it will disconnect within 30 seconds dlg_set_property(ka-src); dlg_set_property(ka-dst); thesse 2 will make sure that keep alive timers will apply to the current dialog Kelvin Chua On Mon, Apr 7, 2014 at 1:43 AM, Vassilis Radis rad...@gmail.com wrote: Hello, I am trying to use the dialog module for dealing with some cases where clients do not send BYE messages correctly and just disappear. For this reason i use the ka_timer and ka_interval along with ka-dst and ka-src attributes. I am using the master branch. I have setup the following test: I have setup a sipp instance as a callee uas configured to discard OPTIONS messages. Kamailio serves as a proxy between this sipp instance and a VoIP device registered to kamailio. Kamailio is also configured to send every call to the sipp uas with stateful proxing and dialog support: CALLER -- KAMAILIO - CALLEE (sipp / ignores OPTIONS msg) I have the following issue: 1. Dialog does send the OPTIONS messages to caller and callee, and as expected, only the caller responds (with a 200 OK response). But the dialog module never terminates the dialog as it should (because the callee never responds to the OPTIONS msg). Instead it keeps sending those OPTIONS for ever. It is like there is no timer entry in the timer list for those OPTIONS. I looked a bit in the source code and I am trying to find where a timeout for those keep alives is a)set and b)handled . I see that the dlg_timer_routine is called every second but after turning debugging on, i see that the only timer reported is the default timeout for the call which i ve set to 3600 secs in the module parameters. Any insight or advice? Thanks a lot. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] No audio issue
This is webrtc, using Kamailio with websocket relay to Asterisk. I am not using rtpproxy 07.04.2014, 22:49, Kelvin Chua kel...@gmail.com: is this webrtc?are you using rtpproxy? Kelvin Chua On Mon, Apr 7, 2014 at 6:37 AM, jaflong jaflong jafl...@yandex.com wrote: Hi, I am at the point where connection is established and no apparent errors are reported. However audio is not output. The rtp traffic seems to be transfering between the points as conclueded because Asterisk debug log shows Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021868, ts 221760, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001383, ts 1917269534, len 000160) Sent RTP packet to 10.1.xxx.xxx41143 (via ICE) (type 08, seq 021869, ts 221920, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001384, ts 1917269694, len 000160) Sent RTP packet to 10.1.xxx.xxx:41143 (via ICE) (type 08, seq 021870, ts 222080, len 4294967284) Got RTP packet from 10.1.xxx.xxx:41143 (type 08, seq 001385, ts 1917269854, len 000160) And the browser machine on the other endpoint on a tcpdump does shows traffic on the port (41143) What could be causing there to be no audio? This is the connected sdp =0 o=root 350315728 350315728 IN IP4 10.31.xxx.xxx s=Asterisk PBX 12.2.0-rc1 c=IN IP4 10.31.xxx.xxx t=0 0 m=audio 24316 UDP/TLS/RTP/SAVPF 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=maxptime:150 a=ice-ufrag:1c5c5d52130f06fd70e1e23f0d6323f2 a=ice-pwd:12611b8146599a9019d59b4b649a7970 a=candidate:Ha1f026f 1 UDP 2130706431 10.31.xxx.xxx 24316 typ host a=candidate:Ha1f026f 2 UDP 2130706430 10.31.xxx.xxx 24317 typ host a=connection:new a=setup:active a=fingerprint:SHA-256 13:BB:CF:88:C4:75:9B:F0:DA:36:0A:6D:5D:37:C9:26:6B:3C:82:3E:F6:92:AE:A7:AE:CF:FF:78:F5:86:D9:E8 a=sendrecv ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users , ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Show Domain-Name in Siremis
The contact is the address of the device. To get the domain part set, you have to set use_domain to usrloc module in kamailio.cfg Cheers, Daniel On 08/04/14 10:08, aawaise wrote: I am looking at the LOCATION LIST under SER MONITOR MENU. I am looking at the online users. The user's IP adress can been seen the table infront of Contact but there is nothing present infront of Domain. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Show-Domain-Name-in-Siremis-tp126602p126613.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Caller ID number being passed to Gateway
FreeSWITCH shows nobody@IP I just need to know the proper way to pass did information Sent from my iPad On Apr 7, 2014, at 7:33 PM, Joel E White joelewh...@gmail.com wrote: Kamailio Version 4.1 No, when I do the uac_replace_from the username or the did (specified in the db) do not show in FreeSWITCH On 4/7/2014 2:46 PM, Kelvin Chua wrote: hi joel, what kamailio version are you using? can you verify if you see your old username at the end of the domain? for example: us...@domain.com after calling uac_replace_from you wll get user2@domain.comuser1 Kelvin Chua On Mon, Apr 7, 2014 at 11:36 AM, Joel White joelewh...@gmail.com wrote: What I am getting on the FreeSWITCH side is this... Name passes, but DID is set without @ IP From: Test Phone 2 1XXX Call does go through, but not with the information I would like On Mon, Apr 7, 2014 at 12:13 PM, Joel E White joelewh...@gmail.com wrote: I have been playing with this for a couple days. What I have pulls from the usr_preferences DB and Should insert CID Name and CID Number What I am ending up with is that the name portion is being sent, not the number. Is there a certain format that I should add the DID in usr_preferences? Also when I changed the AVP type in the DB from a 0 to a 1 the value for CID DID changes... can someone help me with correcting this behavior? avp_db_load($from,$avp(s:callerid-name)); avp_db_load($from,$avp(s:caller-did)); uac_replace_from($avp(s:callerid-name),$avp(s:callerid-did)); xlog(SIP From Header returned $hdr(From)\n); xlog(AVP CallerID-Name returned $avp(s:callerid-name)\n); xlog(AVP CallerID-Ext returned $avp(s:callerid-did)\n); Thank you, Joel --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users This email is free from viruses and malware because avast! Antivirus protection is active. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] RTP not coming via RTPProxy
HI All, I am using the Kamailio 4.1.2 with RTPProxy 1.2.1 as a SIP Proxy. I want to ensure that all my sip and media (RTP) passes through the SIP Proxy (Kamailio + RTP Proxy) while we my SIP Client sends any SIP and RTP to SIP Server (asterisk). So it is like below: SIP Agent SIP+RTP--SIP Proxy(Kamailio+RTPProxy)--SIP+RTPSIP Server(Asterisk). Now it is working fine with my current configuration on Kamailio with RTPProxy with SIP and RTP both if my SIP Agent is on the 802.11 LAN that is my PC have local IP address like 192.168.1.2. Now I am facing the problem in the case where my PC is accessing the internet through the USB data card (or Internet USB dongle). Here my PC IP is like 116.203.51.209. In this case my SIP is successfully passes through the SIP Proxy and works good. But in this case my RTP is does not passes through the SIP Proxy (RTPProxy). The SIP Proxy(Kamailio + RTPPRoxy) and SIP Server is running on the public IP. Please find attached my kamailio.cfg. Please let me know what is the issue. Thanks and Regards Varun kamailio.cfg Description: Binary data ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cnxcc
Rome! Great choice! :) Daniel On 04/08/2014 10:31 AM, Carlos Ruiz Díaz wrote: Hi Carsten, that's right, I'm replying to this while I wait for a tour to the Vatican City :-D ;-) Cheers, On Tue, Apr 8, 2014 at 3:27 AM, Carsten Bock cars...@ng-voice.com mailto:cars...@ng-voice.com wrote: Hi Carlos, please remember: You're on your Honeymoon! :-) Enjoy Rome, Carsten 2014-04-08 10:26 GMT+02:00 Carlos Ruiz Díaz carlos.ruizd...@gmail.com mailto:carlos.ruizd...@gmail.com: Hi Kelvin, probably the dialog matching parameter is wrong, or something in the parameter configuration of cnxcc. The value of max_amount is 0, because the call wasn't established or was established and cnxcc couldn't detect it (probably because of the dlg matching, as I mentioned). It will retain this value for as long as the call is in the early state. Check this example [1], it may help you. [1] http://caruizdiaz.com/2014/04/04/cnxcc-prepaid-module-workshop-kamailio-world/ Regards, On Tue, Apr 8, 2014 at 3:04 AM, Kelvin Chua kel...@gmail.com mailto:kel...@gmail.com wrote: i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 tel:%2B595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 tel:%2B49%2040%2034927219 Fax +49 40 34927220 tel:%2B49%2040%2034927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
On 04/08/14 03:00, Olli Heiskanen wrote: Hello, Thanks Juha, that will be a good thing to investigate more when I get my simple unrealistic scenario working. :) I tried compiling rtpengine on Centos 6.5, I wonder do I need to change the Makefile somehow for CentOs? Remove Debian specific flags like mentioned in the github page? Below is the output from the make: In file included from call.c:25: ../kernel-module/xt_MEDIAPROXY.h:23: warning: declaration does not declare anything call.c: In function ‘kernelize’: call.c:342: error: ‘struct mp_address’ has no member named ‘ipv4’ call.c:344: error: ‘struct mp_address’ has no member named ‘ipv4’ call.c:348: error: ‘struct mp_address’ has no member named ‘ipv6’ call.c:348: error: ‘struct mp_address’ has no member named ‘ipv6’ call.c:350: error: ‘struct mp_address’ has no member named ‘ipv6’ call.c:350: error: ‘struct mp_address’ has no member named ‘ipv6’ call.c: In function ‘call_destroy’: call.c:1961: warning: format ‘%lu’ expects type ‘long unsigned int’, but argument 8 has type ‘u_int64_t’ call.c:1961: warning: format ‘%lu’ expects type ‘long unsigned int’, but argument 9 has type ‘u_int64_t’ call.c:1961: warning: format ‘%lu’ expects type ‘long unsigned int’, but argument 10 has type ‘u_int64_t’ make[1]: *** [call.o] Error 1 make[1]: Leaving directory `/usr/local/src/rtpengine/daemon' make: *** [all] Error 2 Those should be fixed now. cheers signature.asc Description: OpenPGP digital signature ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] uac_replace_from
Hi!
I had similar problem and I got it finally thanks to the mailing list
support. In my case I was trying to modify TO header and when it went to the
failure route it was modified just as you told us with the FROM.
I got it by applying the modification (uac_replace_to) only once on t_branch
and in all other places updating the variable. I copy here Castern tip which
put me in the right direction. Try just the same but with from!
I hope it works for you as well as for me!
Helena
route[FROMPHONE] {
[...]
if(!ds_select_domain(1, 8))
{
send_reply(404, No destination);
exit;
}
t_on_failure(FAILURE_ROUTE);
t_on_branch(MODIFY_TO);
subst_uri('/^sip:(.*)/sip:0199\1/i'); # add prefix to URI
$avp(s:new_to) = sip:0199+$rU+@+$rd;
route(RELAY);
exit;
}
branch_route[MODIFY_TO] {
uac_replace_to(, $avp(s:new_to));
}
failure_route[FAILURE_ROUTE] {
if (t_is_canceled()) {
exit;
}
if (t_check_status(500) or (t_branch_timeout() and
!t_branch_replied()))
{
if(ds_next_domain())
{
t_on_failure(RTF_DISPATCH); #in case of
t_on_branch(MODIFY_TO);
$avp(s:new_to) = sip:+$rU+@+$rd; #ru already has
the prefix
route(RELAY);
exit;
}
}
}
This way, uac_replace_to() gets only called once per destination.
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel Tryba
Sent: viernes, 04 de abril de 2014 10:46
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] uac_replace_from
On Friday 04 April 2014 09:53:48 Alex Balashov wrote:
I've seen this when calling uac_replace_from twice, which you cannot do.
I ran into this before, and based upon recommendations here I decided to
store the changes in avps and commit the changes in route[RELAY].
That works fine until something ends up in a failure route (redirects after
a fr_inv_timeout). It appears I get double changes even though I reset the
avps after calling uac_* in RELAY so uac_* shouldn't get called for a second
time anywhere, but this is something I'll have to debug further before
making more statements.
--
POCOS B.V. - Croy 9c - 5653 LC Eindhoven
Telefoon: 040 293 8661 - Fax: 040 293 8658
http://www.pocos.nl/ - http://www.sipo.nl/
K.v.K. Eindhoven 17097024
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Re: [SR-Users] Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP
Hi, Thanks, it compiled nicely, I'll continue with more testing tomorrow. - Olli 2014-04-08 15:36 GMT+03:00 Richard Fuchs rfu...@sipwise.com: On 04/08/14 03:00, Olli Heiskanen wrote: Hello, Thanks Juha, that will be a good thing to investigate more when I get my simple unrealistic scenario working. :) I tried compiling rtpengine on Centos 6.5, I wonder do I need to change the Makefile somehow for CentOs? Remove Debian specific flags like mentioned in the github page? Below is the output from the make: In file included from call.c:25: ../kernel-module/xt_MEDIAPROXY.h:23: warning: declaration does not declare anything call.c: In function 'kernelize': call.c:342: error: 'struct mp_address' has no member named 'ipv4' call.c:344: error: 'struct mp_address' has no member named 'ipv4' call.c:348: error: 'struct mp_address' has no member named 'ipv6' call.c:348: error: 'struct mp_address' has no member named 'ipv6' call.c:350: error: 'struct mp_address' has no member named 'ipv6' call.c:350: error: 'struct mp_address' has no member named 'ipv6' call.c: In function 'call_destroy': call.c:1961: warning: format '%lu' expects type 'long unsigned int', but argument 8 has type 'u_int64_t' call.c:1961: warning: format '%lu' expects type 'long unsigned int', but argument 9 has type 'u_int64_t' call.c:1961: warning: format '%lu' expects type 'long unsigned int', but argument 10 has type 'u_int64_t' make[1]: *** [call.o] Error 1 make[1]: Leaving directory `/usr/local/src/rtpengine/daemon' make: *** [all] Error 2 Those should be fixed now. cheers ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] cnxcc
Wow rome! great choice carlos! enjoy your honeymoon! :) here is what i did: #!define FLD_CNXCC 5 modparam(cnxcc, dlg_flag, FLD_CNXCC) cnxcc_set_max_credit(1 , 196.9485 , 0.011000 , 6 , 6); number_of_calls increments for every call and i never see max_amount going non-zero client_id:6,number_of_calls:2,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; Kelvin Chua On Tue, Apr 8, 2014 at 5:19 AM, Daniel Grotti dgro...@sipwise.com wrote: Rome! Great choice! :) Daniel On 04/08/2014 10:31 AM, Carlos Ruiz Díaz wrote: Hi Carsten, that's right, I'm replying to this while I wait for a tour to the Vatican City :-D ;-) Cheers, On Tue, Apr 8, 2014 at 3:27 AM, Carsten Bock cars...@ng-voice.com mailto:cars...@ng-voice.com wrote: Hi Carlos, please remember: You're on your Honeymoon! :-) Enjoy Rome, Carsten 2014-04-08 10:26 GMT+02:00 Carlos Ruiz Díaz carlos.ruizd...@gmail.com mailto:carlos.ruizd...@gmail.com: Hi Kelvin, probably the dialog matching parameter is wrong, or something in the parameter configuration of cnxcc. The value of max_amount is 0, because the call wasn't established or was established and cnxcc couldn't detect it (probably because of the dlg matching, as I mentioned). It will retain this value for as long as the call is in the early state. Check this example [1], it may help you. [1] http://caruizdiaz.com/2014/04/04/cnxcc-prepaid-module-workshop-kamailio-world/ Regards, On Tue, Apr 8, 2014 at 3:04 AM, Kelvin Chua kel...@gmail.com mailto:kel...@gmail.com wrote: i am trying to use cnxcc for the first time. kamctl kamcmd cnxcc.active_clients client_id:6,number_of_calls:1,concurrent_calls:0,type:1,max_amount:0.00,consumed_amount:0.00; i don't think this is correct, seems like the dialog is not disengaging after i hang up. i created dialogs alongside cnxcc just to compare, all dialogs were disengaged. is this an expected behavior? It's also weird that the max_amount is 0 while i checked that there is a value for credit being passed as an argument. Kelvin Chua ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto: sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 tel:%2B595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto: sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:cars...@ng-voice.com mailto:cars...@ng-voice.com Office +49 40 34927219 tel:%2B49%2040%2034927219 Fax +49 40 34927220 tel:%2B49%2040%2034927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Carlos http://caruizdiaz.com http://ngvoice.com +595981146623 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dialog: Keep-alive issue:
1. Ok I think I found the cause for the problem:
First of all I noticed that the dialog doesn't timeout if the ka_timer
module parameter is less or equal to 10 secs. If it is above 10 secs then
everything works.
I use the master branch and in the dialog module (
https://github.com/kamailio/kamailio/blob/master/modules/dialog/dlg_req_within.c)
in line 264 there is this code:
if(ps-code==408 || ps-code==481) {
if(update_dlg_timer(dlg-tl, 10)0) {
LM_ERR(failed to update dialog lifetime\n);
goto done;
}
dlg-lifetime = 10;
dlg-dflags |= DLG_FLAG_CHANGED;
}
This code is for updating the dialog's lifetime when a timeout occurs (by a
fake 408 reply) or a 481 is received.
But in the call of update_dlg_timer function above, the second argument is
a hardcoded 10, which causes the dialog to refresh its lifetime to 10 more
seconds, every time a timeout occurs, and because when ka_timer is 10
secs this gets refreshed again before the dialog expires. I did a test with
a value of 1 and it correctly works for values of ka_timer = 2
I cant say the correct value, propably should be 1, so that the dialog gets
expired the next second that the dlg_timer_routine runs. Also this value
should be less than the ka_timer value so there should be a minimum valid
ka_timer value.
I also looked in the 4.1.2 release and it is there too.
2. Is it possible to manually define the OPTIONS timeout when sending it?
Currently is around 30 secs, and I would like to make it less for this
specific use (dialog keep-alives)
Thanks.
On Tue, Apr 8, 2014 at 12:12 PM, Vassilis Radis rad...@gmail.com wrote:
Thanks Kelvin, but I already have tried this. This is exactly what I wrote
that the issue is: Despite having set those options, and the keep alives
are sent, the dialog does not timeout as it should when no reply is sent to
the keep-alive OPTIONS msg by the dst peer.
I am using the master branch and I am trying to follow the source code to
see how and where the dialog module should be called back when the timeout
for the OPTIONS message occurs, but I am a little lost. If you think I am
missing something and there is a simple configuration solution please do
point it to me.
Thanks a lot.
On Mon, Apr 7, 2014 at 10:02 PM, Kelvin Chua kel...@gmail.com wrote:
try this
modparam(dialog, ka_timer, 3)
modparam(dialog, ka_interval, 10)
these 2 will make sure it will disconnect within 30 seconds
dlg_set_property(ka-src);
dlg_set_property(ka-dst);
thesse 2 will make sure that keep alive timers will apply to the current
dialog
Kelvin Chua
On Mon, Apr 7, 2014 at 1:43 AM, Vassilis Radis rad...@gmail.com wrote:
Hello,
I am trying to use the dialog module for dealing with some cases where
clients do not send BYE messages correctly and just disappear. For this
reason i use the ka_timer and ka_interval along with ka-dst and ka-src
attributes. I am using the master branch. I have setup the following test:
I have setup a sipp instance as a callee uas configured to discard OPTIONS
messages. Kamailio serves as a proxy between this sipp instance and a VoIP
device registered to kamailio. Kamailio is also configured to send every
call to the sipp uas with stateful proxing and dialog support:
CALLER -- KAMAILIO - CALLEE (sipp / ignores OPTIONS msg)
I have the following issue:
1. Dialog does send the OPTIONS messages to caller and callee, and as
expected, only the caller responds (with a 200 OK response). But the dialog
module never terminates the dialog as it should (because the callee never
responds to the OPTIONS msg). Instead it keeps sending those OPTIONS for
ever. It is like there is no timer entry in the timer list for those
OPTIONS. I looked a bit in the source code and I am trying to find where a
timeout for those keep alives is a)set and b)handled . I see that the
dlg_timer_routine is called every second but after turning debugging on, i
see that the only timer reported is the default timeout for the call which
i ve set to 3600 secs in the module parameters.
Any insight or advice?
Thanks a lot.
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Re: [SR-Users] Segfault when trying to send MESSAGE through websocket with jsSIP
El 04/04/14 16:26, Alex Villacís Lasso escribió: I am currently trying to replace a pure-Asterisk implementation of SIP messaging through Websockets, with a Kamailio-4.1.2-based implementation. However, when I try to send a message with jsSIP, Kamailio crashes: Program terminated with signal 11, Segmentation fault. #0 0x7f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 350slot = reg_get_entry(hash, _reg_htable-htsize); Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.132.el6.x86_64 keyutils-libs-1.4-4.el6.x86_64 krb5-libs-1.10.3-10.el6_4.6.x86_64 libcom_err-1.41.12-18.el6.x86_64 libgcc-4.4.7-4.el6.x86_64 libselinux-2.0.94-5.3.el6_4.1.x86_64 libtool-ltdl-2.2.6-15.5.el6.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-14.el6.x86_64 mysql-connector-odbc-5.1.5r1144-7.el6.x86_64 mysql-libs-5.1.73-3.el6_5.x86_64 nss-softokn-freebl-3.14.3-9.el6.x86_64 openssl-1.0.1e-16.el6_5.4.x86_64 unixODBC-2.2.14-12.el6_3.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x7f0e5cf31be3 in reg_ht_get_byuuid (uuid=0x7fff59734b00) at uac_reg.c:350 #1 0x7f0e5cf36c71 in uac_reg_lookup (msg=0x7f0e6271e790, src=0x7fff59734b00, dst=0x7f0e6267c950, mode=0) at uac_reg.c:924 #2 0x7f0e5cf2f991 in w_uac_reg_lookup (msg=0x7f0e6271e790, src=0x7f0e6267e0d0 \a, dst=0x7f0e6267c950 \006) at uac.c:560 #3 0x00419bf6 in do_action (h=0x7fff59735690, a=0x7f0e6267f1e0, msg=0x7f0e6271e790) at action.c: #4 0x00422878 in run_actions (h=0x7fff59735690, a=0x7f0e6267e510, msg=0x7f0e6271e790) at action.c:1599 #5 0x00417900 in do_action (h=0x7fff59735690, a=0x7f0e62664aa0, msg=0x7f0e6271e790) at action.c:715 #6 0x00422878 in run_actions (h=0x7fff59735690, a=0x7f0e6265d3b8, msg=0x7f0e6271e790) at action.c:1599 #7 0x00423017 in run_top_route (a=0x7f0e6265d3b8, msg=0x7f0e6271e790, c=0x0) at action.c:1685 #8 0x004a5153 in receive_msg ( buf=0x7f0e570d0168 MESSAGE sip:ad...@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: sip:ad...@gatitos.com\r\nFrom: \Alex Villacís Lasso\ sip:avillacisIM@gatitos, len=585, rcv_info=0x7f0e570cfe90) at receive.c:212 #9 0x7f0e5c8e5802 in ws_frame_receive (data=0x7fff59735a40) at ws_frame.c:652 #10 0x0045531c in sr_event_exec (type=10, data=0x7fff59735a40) at events.c:254 #11 0x0052e04e in ws_process_msg ( tcpbuf=0x7f0e570d0160 \201\376\002I\032\327\302\344MESSAGE sip:ad...@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: sip:ad...@gatitos.com\r\nFrom: \Alex Villacís Lasso\ sip:avillacisIM@..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1146 #12 0x0052e21f in receive_tcp_msg ( tcpbuf=0x7f0e570d0160 \201\376\002I\032\327\302\344MESSAGE sip:ad...@gatitos.com SIP/2.0\r\nVia: SIP/2.0/WS ftisg2dgtaqe.invalid;branch=z9hG4bK2656184\r\nMax-Forwards: 16\r\nTo: sip:ad...@gatitos.com\r\nFrom: \Alex Villacís Lasso\ sip:avillacisIM@..., len=593, rcv_info=0x7f0e570cfe90, con=0x7f0e570cfe78) at tcp_read.c:1182 #13 0x0052f2f3 in tcp_read_req (con=0x7f0e570cfe78, bytes_read=0x7fff59735cac, read_flags=0x7fff59735ca4) at tcp_read.c:1383 #14 0x00530d27 in handle_io (fm=0x7f0e62724d30, events=1, idx=-1) at tcp_read.c:1617 #15 0x005296c0 in io_wait_loop_epoll (h=0x8df220, t=2, repeat=0) at io_wait.h:1092 #16 0x00531650 in tcp_receive_loop (unix_sock=53) at tcp_read.c:1728 #17 0x00523c21 in tcp_init_children () at tcp_main.c:4959 #18 0x0046d6a3 in main_loop () at main.c:1702 #19 0x0047030b in main (argc=13, argv=0x7fff59736178) at main.c:2533 Is this a known bug? For additional information, the segfault stems from dereferencing a NULL pointer at global variable '_reg_htable' declared at modules/uac/uac_reg.c , which in turn looks as if the process (or its parents before the fork()) failed to call uac_reg_init_ht() . This is confirmed by a patch to return NULL from reg_ht_get_byuuid() if _reg_htable is NULL - but this variable should not be uninitialized in the first place. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
