sr-users October 2013

sr-users@lists.kamailio.org

78 participants
152 discussions

dispatcher: how to set the transport protocol
by Karimi-Cherkandi, Bizhan 01 Nov '13

01 Nov '13

kamailio dispatcher module in my environment uses only UDP I have not been able to find how I can configure kamailio dispatcher module to use TCP Would appreciate any help Bijan

3 6

Kamailio - RTPProxy - Asymmetric RTP
by Lucas Girard 01 Nov '13

01 Nov '13

Hi all, I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is used. I have the system running, lines registered in Kamailio are able to call to several destinations, but I have a problem with one provider that use Asymmetric RTP (others providers use Symmetric RTP and I dont have any issues). A -----> Kamailio & RTPProxy ----------> Trunk -------> Provider (Asymmetric RTP) -------> B A is calling B. The problem is that line in kamailio A can listen to B, but B can not listen what A says. I have captured the sip dialog and rtp traffic and I can see that: Kamailio tells to provider that is using the port 52388 (media port in SDP). Providers tells Kamailio that is using the port 5394 (media port in SDP). RTP traffic from Providers to kamailio goes from port 5392 to 52388. RTP traffic from Kamailio to Provider goes from port 52388 to 5392. I have not found an RFC regarding Asymmetric RTP. The question are, why RTP is sent from Kamailio (rtpproxy) to Provider to destination port 5392? It should use 5394 as SDP indicate? or, port 5392 is learned when RTP traffic was received from 5392 to 52388? # kamailio -V version: kamailio 3.3.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 22:17:07 Jun 18 2012 with gcc 4.4.5 # rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications I hope to be clear with this explanation. Regards, Lucas Girard

3 7

Help : Kamailio RTP Proxy Issue...
by Mahmoud Ramadan Ali 31 Oct '13

31 Oct '13

Hello , This is the first time to put a question on sr-users mailing list and i hope to help me to solve my issue...i'm following this post http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html to setup RTP Proxy for kamailio 4 but it's NOT working as predicted and i'm receiving this error message on asterisk... [2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)", ...): Name or service not known [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873 process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4 192.168.1.80192.168.1.80' [2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464 process_sdp: Insufficient information in SDP (c=)... The problem is NOT at the asterisk side but at the proxy side...when kamailio rewrites the SDP message it adds the public IP twice in the "C" filed of the SDP message instead of only one time like this 'IN IP4 192.168.1.80192.168.1.80' this is because the scripte is calling rtpproxy_manage() in the routes TOASTERISK /FROMASTERISK and in MANAGE_BRANCH / NATMANAGE. It should be either or,not both....but i don't know how to do this since i don't have any scripting skills so if anyone can modify the config file in the attachment i will be thankful... Best Regards' Mahmoud Ramadan Ali

2 2

Hardware requirements/recommandations
by Igor Potjevlesch 30 Oct '13

30 Oct '13

Hello, I'm looking for the best practises for setting Kamailio and the associated hardware requirements. Kamailio must deliver registrar and routing functions for 500k aliases, 5k registered accounts and ~10k routes set in carrierroute module or similar. My worries mainly concern DB integration. Is it better to setup Kamailio DB on the same server (no network issue, etc.) or on another server? Your opinion/feedback experience would be appreciated. Thanks! Regards, Igor. --- Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active. http://www.avast.com

2 7

Upgrade v4.0.x to 4.1.0 guidelines
by Daniel-Constantin Mierla 30 Oct '13

30 Oct '13

Hello, this is a call for developers and community users to update the page collecting upgrade guidelines from current stable (4.0.x) to upcoming major release (4.1.0): - https://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0 If you are aware of a change in the behaviour for some parameters, functions or statements, list them there. I added the sql statements for MySQL that will allow to get to the new database structure. Would be good to have the same for other popular sql servers (maybe postgres, sqlite ...). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

1 0

Inser Delayed in MySQL
by Gus Monti 29 Oct '13

29 Oct '13

Hi, I can see that we are missing a lot of Sip messages in or db, so, checking Homer FAQ there is a way to compile MySql with Insert Delayed for partitioning table. The problem is that I don't understand how to recompile MySql and make the necessary changes. The URL is https://code.google.com/p/homer/wiki/FAQ#Q:_I_have_a_lot_of_SIP_traffic_an d_some_SIP_messages_are_missing The question is, can you send me a step by step how to do this? I really appreciate your help. I have CentOS 6.4: [root@Homerdb02 ~]# kamailio -v version: kamailio 4.1.0-dev9 (x86_64/linux) f8f3d3 [root@Homerdb02 ~]# cat /etc/redhat-release Red Hat Enterprise Linux Server release 6.4 (Santiago) [root@Homerdb02 ~]# uname -a Linux Homerdb02 2.6.32-358.18.1.el6.x86_64 #1 SMP Fri Aug 2 17:04:38 EDT 2013 x86_64 x86_64 x86_64 GNU/Linux [root@Homerdb02 ~]# mysql -V mysql Ver 14.14 Distrib 5.5.34, for Linux (x86_64) using readline 5.1 Thanks, Gus

2 2

Kamailio Tech Admin Group
by Daniel-Constantin Mierla 29 Oct '13

29 Oct '13

Hello, being discussed during last Devel IRC Meeting, we are planing to build a Kamailio Project Technical Administration Group: https://www.kamailio.org/wiki/devel/irc-meetings/2013blog#technical_adminis… Its goal is to get a bunch of people that volunteer to do administration tasks for the project, such as: - helping with releases (e.g., patch backports, packaging, uploading files for download, etc) - doing sysadmin tasks for our servers (e.g., performing upgrades to wiki, web site, etc) - preparing technical decisions and doing them (e.g., what applications to use to make operations easier, cloning git repository to github, ...) From the devel meeting, so far we have Victor Seva, Fred Posner, Peter Dunkley and Olle Johansson. Existing people doing admin tasks will probably stay in (if they don't opt out): me, Elena-Ramona Modroiu, Henning Westerholt (owner of devel.kamailio.org hardware), Jan Janak (owner of sip-router.org hardware), Jesus Rodriguez and Oriol Capsada (owners of kamailio.org hardware). Requirements for candidates and other details: - volunteer to do the work, it is not a paid job - an existing record of activity within the project is a plus (e.g., developer, active mailing list member) - reply to the lists detailing where and how you can help - possibility to spend 1-4 hours a week for project administration (more is welcome, sometime is not necessary at all) Rewards: - you will be listed as part of project administration on the website - get to interact more with the project and the nice guys around it ;-) - more spam - admin list address will be public and the list open so everyone can send in case of critical situations (content/archive will be kept private) Note that we will try to build a group of an adequate size, thus not everyone willing to participate may get in (at least on the first phase). One criteria is to have skills that complement existing team knowledge. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Trainings - Berlin, Nov 25-28 - more details about Kamailio trainings at http://www.asipto.com -

5 5

ACK not sending to the invite to header but to the contact.
by anfecora 29 Oct '13

29 Oct '13

Hi all, can anyone help me to find out what is wrong with my setup, i have an asterisk behind a kamailio, kamailio is proxying all packages to the outside. when the call is bridge it gets disconnected after a few seconds, it seems that our voip carrier is sending a bye because we didn't answer to their 200 ok propperly, but as the trace shows we did only that kamailio is answering to the contact header ip not the ip that is sending the ok. any help is apreciated . thanks. my setup request_route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { # send reply for each options request sl_send_reply("200", "ok"); exit(); } if(method=="BYE") { #Account BYE transactions }; if (method=="CANCEL") { if (t_check_trans()) t_relay(); exit; }; if (loose_route()) { t_relay(); exit; } if (is_method("INVITE")) { record_route(); } f (!t_relay_to_udp("3.1.1.1", "5060")) { sl_reply_error(); exit; }; exit }; here is a trace to a call made to a hotel. i had changed the real ips for obvious reasons. thanks. asterisk ip 1.1.1.1 kamailio internal 1.1.1.2 kamailio external 2.0.0.1 Voip Carrier 3.1.1.1 voip contact ip 3.1.1.2 U 2013/10/23 17:26:03.920163 1.1.1.1:5060 -> 1.1.1.2:5060 INVITE sip:23276341079@2.0.0.1 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.921355 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: kamailio (4.0.4 (x86_64/linux)). Content-Length: 0. . U 2013/10/23 17:26:03.921544 1.1.1.2:5060 -> 3.1.1.1:5060 INVITE sip:76890723276341079@3.1.1.1:5060 SIP/2.0. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. User-Agent: Asterisk PBX 1.8.15-cert2. Date: Wed, 23 Oct 2013 21:26:46 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Privacy: off. P-Asserted-Identity: sip:+19812457865@1.1.1.1. Cisco-Guid: 25655507-3591552378-379709 Content-Type: application/sdp. Content-Length: 333. . v=0. o=root 519803789 519803789 IN IP4 1.1.1.1. s=Asterisk PBX 1.8.15-cert2. c=IN IP4 1.1.1.1. t=0 0. m=audio 49926 RTP/AVP 0 18 3 8 101. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2013/10/23 17:26:03.955394 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.1>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Server: gProxy (1.8.3 (i386/Linux)). Content-Length: 0. . U 2013/10/23 17:26:04.424330 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:04.424521 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846067 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:16.846201 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. T 2013/10/23 17:26:16.846287 1.1.1.2:55305 -> 10.0.3.54:3306 [AP] .....insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,kekuintid,type_call,dst_ip,carriercode,callmode ) values ('INVITE','as4bc322e9','3591552407-393967',' 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060','200','OK','2013-10-23 17:26:16','sip:+19812457865@1.1.1.1','sip:23276341079@2.0.0.1 ','+19812457865','1.1.1.1','sip:76890723276341079@3.1.1.1:5060',' sip:23276341079@2.0.0.1','OUT') U 2013/10/23 17:26:16.847421 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:16.847651 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK05b1c5df;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.346094 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.346262 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:17.349001 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:17.349223 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK675a9141;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.347584 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.347767 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:18.348867 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:18.349133 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK3fc3c548;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.352624 3.1.1.1:5060 -> 1.1.1.2:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bK887c.94fdcd27.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.353056 1.1.1.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. Session-Expires: 3600;refresher=uas. Require: timer. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK4bd52990;rport=5060. Record-Route: <sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Record-Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 INVITE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Contact: <sip:76890723276341079@3.1.1.2:5060>. Call-Info: <sip:3.1.1.2>;method="NOTIFY;Event=telephone-event;Duration=1000". Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 202. . v=0. o=MSXB 4755 8544 IN IP4 3.1.1.2. s=sip call. c=IN IP4 204.15.40.111. t=0 0. m=audio 33408 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2013/10/23 17:26:20.354026 1.1.1.1:5060 -> 1.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 70. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:23276341079@2.0.0.1>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:20.354248 1.1.1.2:5060 -> 3.1.1.2:5060 ACK sip:76890723276341079@3.1.1.2:5060 SIP/2.0. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>. Max-Forwards: 16. From: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. To: <sip:76890723276341079@3.1.1.2>;tag=3591552407-393967. Contact: <sip:+19812457865@1.1.1.1:5060>. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 102 ACK. User-Agent: Asterisk PBX 1.8.15-cert2. Content-Length: 0. . U 2013/10/23 17:26:36.355580 3.1.1.1:5060 -> 1.1.1.2:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 69. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:+19812457865@1.1.1.1>;tag=as4bc322e9. From: <sip:76890723276341079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. . U 2013/10/23 17:26:36.355995 1.1.1.2:5060 -> 1.1.1.1:5060 BYE sip:+19812457865@1.1.1.1:5060 SIP/2.0. Max-Forwards: 16. Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>. To: "+19812457865" <sip:176822213@1.1.1.1>;tag=as4bc322e9. From: 0 <sip:079@3.1.1.1>;tag=3591552407-393967. Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060. CSeq: 2 BYE. Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE. Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKce8a.52d22d63.0. Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0. Via: SIP/2.0/UDP 3.1.1.2:5060 ;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952. Contact: <sip:76890723276341079@3.1.1.2:5060>. Content-Length: 0. .

2 2

Dialog callback after restart
by Kristian F. Høgh 29 Oct '13

29 Oct '13

Hi, After restarting kamailio, pua_dialoginfo doesn't get updated when active dialogs change status. I see the dialogs are read from the database, but when they receive eg. BYE the callback are not called. I use db_mode 0 in pua, but tried mode 2 with no luck. Should pua_dialoginfo register DLGCB_LOADED? Regards, Kristian Høgh.

2 2

Re: [SR-Users] Kamailio network edge for registration and rtp pass-through
by JR Richardson 29 Oct '13

29 Oct '13

Date: Fri, 25 Oct 2013 17:48:13 -0400 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: SR-Users <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio network edge for registration and rtp pass-through Message-ID: <6D231DE3-5BED-4601-A097-5A79CEF452F0(a)palner.com> Content-Type: text/plain; charset=us-ascii Do you need the registration be local to the asterisk? I would have all the asterisks send calls to the Kamailio. You can have a lookup on endpoint outbound to decide which asterisk should handle the outbound call for that did. Also a lookup for incoming DIDs, etc. ---Fred > On Oct 25, 2013, at 5:43 PM, Jr Richardson <jmr.richardson(a)gmail.com> wrote: > > Hi All, > > Starting a new project, roll your own SBC, not a full SBC, just need some minor functionality. I'm interested in deploying Kamailio as a edge device on a VSP for single entry point for hosted PBX's, Asterisk based. I had some wonderful and informative conversations at Astricon 2013, several folks assuring me Kamailio w/rtpproxy was the tool for the job, so this is a follow up to delve more into details. > > I've been researching configs, topology, modules needed, ect... Most of the examples I'm reading about for this scenario are spreading the registrations across many PBX's without distinction. One concept I'm struggling with is having a specific phone register to a specific PBX. > > phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101 > phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101 > phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101 > > I could add a unique identifier to some part of the registration of each phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect. What I'm not clear on is when the request comes to kamailio, where would I identify what PBX the phone should register to and how to re-write the 'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct PBX and ensure rtp flows through kamailio. > > Could this function be derived using dbaliases or possibly using dispatcher with group number for each customer PBX? > > So assuming I can get the registrations to work properly, would standard invite for calling just work or would I also have to have specific config in place to ensure an invite from customer A phone also reaches the correct customer A PBX? > > A point in the right direction? > > Thanks. > > JR I would like kamailio to be transparent in the process, i.e. phones register locally to their respective PBX. Thanks. JR

1 0

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2003

2002

sr-users October 2013