sr-users May 2010

sr-users@lists.kamailio.org

67 participants
106 discussions

by Asim Riaz

Hello List, from the documentation, permissions.allow, allows the calls either by matching From-URI or R-URI and permission.deny denies the call from matching uri. Do I need to divide the file into different sections, for R-URI and for From-URI? Not sure exact syntax could someone explain me that if I want to match in permissions.allow if a phone 10.10.10.10 dial(r-uri) 1234 allow it and everything else is blocked? What is the preference does kamailio check for permissions.deny first if there is no matching then checks permissions.allow? Thanks in advance, Asim

13 years, 11 months

Re: [SR-Users] sipp and stateful transaction problem

by Daniel-Constantin Mierla

Hello, On 5/22/10 2:22 AM, JR Richardson wrote: > On Fri, May 21, 2010 at 4:46 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 5/21/10 10:47 PM, JR Richardson wrote: >> >>> Hi All, >>> >>> I'm doing some testing with kamailio 1.5: >>> >>> kamailio1:/etc/kamailio# kamailio -V >>> version: kamailio 1.5.4-notls (i386/linux) >>> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >>> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> svnrevision: 2:6005M >>> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >>> main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 >>> >>> Using dispatcher module trying to load balance SIP calls across some >>> Asterisk servers. I have it working fine when I test in this >>> scenario: >>> >>> sip phone dial out><asterisk><kamailio><round robin to several asterisk >>> servers >>> >>> This works stateful and stateless, handles everything gracefully. >>> >>> This scenario is giving me fits: >>> >>> sipp dial out><kamailio><round robin to several asterisk servers >>> >>> I get retransmits on every call back to sipp with errors like >>> "Discarding message which can't be mapped to a known SIPp call" and >>> "SIP/2.0 481 Call leg/transaction does not exist" >>> >>> This happens with kamailio setup stateful or stateless. I'm wondering >>> if sipp is the problem or just doesn't play well with kamailio? >>> >>> I've kept the config as simple as possible for testing, it is listed >>> here http://pastebin.com/BZ8hJvJv >>> >>> Here is my sipp usage: >>> >>> sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 >>> -trace_err >>> >>> Any insight would be appriciated. >>> >>> >>> >> the problem is in your sipp scenario. The uac calls do not map to uas. >> kamailio does not reply 481, check the uas scenario, that is the one that >> sends back the 481. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio (OpenSER) Advanced Training >> Miami, Fl, USA - June 21-23, 2010 >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> >> >> > Thanks Daniel, I reveiwed the sipp docs, '-sn uas' just sits there as > a responder, it will not initiate calls to kamailio. I don't > understand what you are getting at? How would I use this type of > scenario to test? > keeping the mailing list cc-ed is recommended, since others can respond faster and new people can benefit of the discussion. What I wanted to say is that kamailio does not reply 481. So the problem is in the responder of requests sent by UAC and forwarded by Kamailio. Somehow, the dialog is destroyed before the BYE (or other in-dialog request) is sent by UAS. If you can grab the SIP trace of such call (e.g., using ngrep on kamailio server), I can give more hits (try to select the sip flow for one such call only, sending full sip trace will be too big). Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

13 years, 11 months

Re: [SR-Users] kamailio 3.0.2 for Thu, May 27, 2010

by Daniel-Constantin Mierla

On 5/25/10 8:51 PM, SIP wrote: > Daniel-Constantin Mierla wrote: > >> Hello, >> >> I am planning to release kamailio 3.0.2 this Thursday. There were some >> fixes since 3.0.1 that worth to be packaged. If you have major reports >> for current stable version, please write to sr-dev(a)lists.sip-router.org >> >> Cheers, >> Daniel >> >> > Is this a drop-in upgrade from 3.0.1 (i.e. recompile modules and core > and keep the rest in place)? > > yes, changes to last number in version string mean that config language and database structure are the same. This is a minor release, bringing only fixes to the code. So upgrading from 3.0.0 or 3.0.1 to 3.0.2 is exactly what you said, just recompile and reinstall modules and core, keep the rest in place. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

13 years, 11 months

could not bind to the TM-Module

by Javier Menendez

Hi, I have been configuring an Kamailio 3.0.1, with the dispatcher module. Everything seems to work just fine but when a gateway is in Probing mode it remains in that state for ever. checking the logs I found some weird messages, I think that is the rrot cause of the problem but I have no clue how to solve it. These are the messages I mentioned before May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:516]: find_export_record: <load_tm> not found May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: WARNING: dispatcher [dispatcher.c:358]: could not bind to the TM-Module, automatic re-activation disabled. May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:871]: DEBUG: init_mod: sl May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [statistics.c:105]: statistics manager successfully initialized May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [md5utils.c:87]: DEBUG: MD5 calculated: f8f2ab2c1295e90ed7dbb499b30f44b2 May 21 14:56:34 localhost /usr/local/sbin/kamailio[21539]: DEBUG: <core> [sr_module.c:506]: find_export_record: found <load_tm> in module tm [/usr/local/lib/kamailio/modules/tm.so] As you may see load_tm is not found in the dispatcher but it is found in md5utils?? These are the modules included mpath="/usr/local/lib/kamailio/modules/:/usr/local/lib/kamailio/modules_k/" loadmodule "tm.so" loadmodule "dispatcher.so" loadmodule "sl.so" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "permissions.so" loadmodule "drouting.so Any clue?

13 years, 11 months

could not open module kamailio

by Bruno Bresciani

I compiled the route.so module to the kamailio but when I start kamailio the follow error is generated in the log file: "ERROR:core:sr_load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/route.so>: /home2/local/kamailio/lib/kamailio/modules/route.so: undefined symbol: add_hash" Someone can help me? Cheers, Bruno

13 years, 11 months

kamailio 3.0.1 rpm for centos 5.5

by Daniel-Constantin Mierla

Hello, for those interested, kamailio 3.0.1 rpm packages for centos 5.5 are available for download at: http://www.kamailio.org/pub/kamailio/3.0.1/packages/centos-5.5/ Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

13 years, 11 months

R: SIP Router devel meeting, Berlin, June 8, 2010

by Daniel-Constantin Mierla

Hello, a short reminder for those that want to participate to SIP Router devel meeting and haven't been in contact with me yet, you need to send email to: *registration(a)lists.sip-router.org* See more details at: http://sip-router.org/2010/05/12/sip-router-devel-meeting-june-8-2010-berli… Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

13 years, 11 months

no Contact in REFER generated with dlg_bridge

by Anton Roman

Hi all, I'm trying to implement a Click2Dial service by using the dlg_bridge command from the dialog module. Although it works in this case, I found two problems in the scenario and I would like to read your opinion about them. 1) Fisrt problem: the REFER generated by Kamailio doesn't contain a contact header. As per RFC 3515, this request should have a Contact header. This REFER is not being accepted by a proprietary device. This is can be worked around with the append_hf() command from the textops module, but I don know if it is a good solution. 2) The second one arises when the ACK generated by Kamailio, goes through Kamailio (caller and calle involved in click2dial are registered on the Kamailio server): the branch of the second via header is not correctly populated. Its content is '0' when it must be tha same as in the INVITE. Below you can find the traces of the whole scenario. Could you take a look at them and give me your feedback? Thanks in advance, best regards Antón Román T +8.969957 127.0.0.1:44785 -> 127.0.0.1:5060 [AP] POST HTTP/1.0 User-Agent: PEAR XML_RPC Host: 127.0.0.1 Content-Type: text/xml Content-Length: 331 <?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>mi_fifo</methodName> <params> <param> <value><string>dlg_bridge</string></value> </param> <param> <value><string>sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103></string></value> </param> <param> <value><string>sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103></string></value> </param> </params> </methodCall> ## U +0.001779 10.1.20.103:5060 -> 10.1.20.103:5060 INVITE sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 147 User-Agent: kamailio (3.0.0 (i386/linux)) Contact: <sip:kamailio.org:5060> Content-Type: application/sdp v=0 o=kamailio-bridge 0 0 IN IP4 0.0.0.0 s=kamailio c=IN IP4 0.0.0.0 t=0 0 m=audio 9 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 # U +0.001330 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7160 req_src_ip=10.1.20.103 req_src_port=5060 in_uri=sip:111@10.1.20.103<sip%3A111(a)10.1.20.103>out_uri= sip:111@10.1.3.15:5061 via_cnt==1" # U +0.000727 10.1.20.103:5060 -> 10.1.3.15:5061 INVITE sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 10 INVITE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 147 User-Agent: kamailio (3.0.0 (i386/linux)) Contact: <sip:kamailio.org:5060> Content-Type: application/sdp v=0 o=kamailio-bridge 0 0 IN IP4 0.0.0.0 s=kamailio c=IN IP4 0.0.0.0 t=0 0 m=audio 9 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 # T +0.000442 127.0.0.1:5060 -> 127.0.0.1:44785 [AP] HTTP/1.0 200 OK Via: SIP/2.0/TCP 127.0.0.1:44785 Server: kamailio (3.0.0 (i386/linux)) Content-Length: 211 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=7172 req_src_ip=127.0.0.1 req_src_port=44785 in_uri=sip:127.0.0.1:9 out_uri=sip: 127.0.0.1:9 via_cnt==1" <?xml version="1.0"?> <methodResponse> <params> <param> <value><array><data> <value><string>200 OK </string></value> <value><string> </string></value> </data></array></value> </param> </params> </methodResponse> ##### U +0.021063 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 100 Trying To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.030585 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000384 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +2.440228 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.31dbafc4.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761338 761338 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000315 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 200 OK To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 10 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 Record-Route: <sip:10.1.20.103;lr=on;did=768.90707151> Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761338 761338 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001190 10.1.20.103:5060 -> 10.1.20.103:5060 ACK sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 CSeq: 10 ACK Route: <sip:10.1.20.103;lr=on;did=768.90707151> User-Agent: kamailio (3.0.0 (i386/linux)) Content-Length: 0 # U +0.001249 10.1.20.103:5060 -> 10.1.3.15:5061 ACK sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 *Via: SIP/2.0/UDP 10.1.20.103;branch=0* Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKe8d6.21dbafc4.0 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 CSeq: 10 ACK User-Agent: kamailio (3.0.0 (i386/linux)) Content-Length: 0 # U +0.000418 10.1.20.103:5060 -> 10.1.20.103:5060 REFER sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 REFER Call-ID: 3d3d4ede-7172(a)127.0.0.1 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> Refer-To: sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> * No Contact!*! # U +0.005133 10.1.20.103:5060 -> 10.1.3.15:5061 REFER sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.1e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 REFER Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> Refer-To: sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> # U +0.046925 10.1.3.15:5061 -> 10.1.20.103:5060 INVITE sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 INVITE Max-Forwards: 70 Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000520 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d44b Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 INVITE Proxy-Authenticate: Digest realm="10.1.20.103", nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813" Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7162 req_src_ip=10.1.3.15 req_src_port=5061 in_uri=sip:112@10.1.20.103<sip%3A112(a)10.1.20.103>out_uri= sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> via_cnt==1" # U +0.004377 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 202 Accepted To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 REFER Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.1e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000193 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 202 Accepted To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 REFER Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.0e0719e2.0 Contact: "111" <sip:111@10.1.3.15:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000286 10.1.20.103:5060 -> 10.1.20.103:5060 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 BYE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) # U +0.003618 10.1.20.103:5060 -> 10.1.3.15:5061 BYE sip:111@10.1.3.15:5061 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.3e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd CSeq: 11 BYE Call-ID: 3d3d4ede-7172(a)127.0.0.1 Content-Length: 0 User-Agent: kamailio (3.0.0 (i386/linux)) # U +0.005223 10.1.3.15:5061 -> 10.1.20.103:5060 ACK sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-9ea2814a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> >;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d44b Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 ACK Max-Forwards: 70 Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.010954 10.1.3.15:5061 -> 10.1.20.103:5060 INVITE sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103> ",algorithm=MD5,response="2baeb40875c0cc7040857daa6e8192fa" Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.001353 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Server: kamailio (3.0.0 (i386/linux)) Content-Length: 0 Warning: 392 10.1.20.103:5060 "Noisy feedback tells: pid=7159 req_src_ip=10.1.3.15 req_src_port=5061 in_uri=sip:112@10.1.20.103<sip%3A112(a)10.1.20.103>out_uri= sip:112@10.1.3.18:5061 via_cnt==1" # U +0.000226 10.1.20.103:5060 -> 10.1.3.18:5061 INVITE sip:112@10.1.3.18:5061 SIP/2.0 Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Max-Forwards: 69 Contact: "111" <sip:111@10.1.3.15:5061> Expires: 240 Referred-By: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 389 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761586 761586 IN IP4 10.1.3.15 s=- c=IN IP4 10.1.3.15 t=0 0 m=audio 16434 RTP/AVP 8 0 2 4 18 96 97 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.007697 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 500 Internal Server Error To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.3e0719e2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000280 10.1.20.103:5060 -> 10.1.20.103:5060 SIP/2.0 500 Internal Server Error To: sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>;tag=d45e4a92cc93b70fi1 From: sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> ;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 11 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKf8d6.2e0719e2.0 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.012355 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 100 Trying To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>> From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.034919 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 180 Ringing To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000260 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 180 Ringing To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.032160 10.1.3.15:5061 -> 10.1.20.103:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.002343 10.1.20.103:5060 -> 193.22.119.66:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK2455.f8178c15.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 69 Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.461009 10.1.20.103:5060 -> 193.22.119.66:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK2455.f8178c15.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 69 Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.028104 10.1.3.15:5061 -> 10.1.20.103:5060 NOTIFY sip:kamailio.org:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-77e77cb8 From: <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103>>;tag=d45e4a92cc93b70fi1 To: <sip:controller@kamailio.org <sip%3Acontroller(a)kamailio.org> >;tag=533cb9e91f4b999cf76861cbb9ed54ed-2abd Call-ID: 3d3d4ede-7172(a)127.0.0.1 CSeq: 101 NOTIFY Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=768.90707151> Event: refer User-Agent: Linksys/SPA941-5.1.5 Subscription-State: active Content-Length: 21 Content-Type: message/sipfrag;version=2.0 SIP/2.0 180 Ringing # U +0.760273 10.1.3.18:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bK7d0c.d071633.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Contact: "112" <sip:112@10.1.3.18:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761170 761170 IN IP4 10.1.3.18 s=- c=IN IP4 10.1.3.18 t=0 0 m=audio 16464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.000246 10.1.20.103:5060 -> 10.1.3.15:5061 SIP/2.0 200 OK To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-ee6fa05e Record-Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Contact: "112" <sip:112@10.1.3.18:5061> Server: Linksys/SPA941-5.1.5 Content-Length: 200 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 761170 761170 IN IP4 10.1.3.18 s=- c=IN IP4 10.1.3.18 t=0 0 m=audio 16464 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv # U +0.022157 10.1.3.15:5061 -> 10.1.20.103:5060 ACK sip:112@10.1.3.18:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-a8e0a92a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 ACK Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.3.18:5061 ",algorithm=MD5,response="3d72eeb2b6b5b2c161685ac3435b657c" Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.009321 10.1.20.103:5060 -> 10.1.3.18:5061 ACK sip:112@10.1.3.18:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=0 Via: SIP/2.0/UDP 10.1.3.15:5061;branch=z9hG4bK-a8e0a92a From: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 To: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 102 ACK Max-Forwards: 69 Proxy-Authorization: Digest username="111",realm="10.1.20.103",nonce="4b7179d300000000c65419875a5dfc25067a1fc4b173b813",uri=" sip:112@10.1.3.18:5061 ",algorithm=MD5,response="3d72eeb2b6b5b2c161685ac3435b657c" Contact: "111" <sip:111@10.1.3.15:5061> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.447617 10.1.3.18:5061 -> 10.1.20.103:5060 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Max-Forwards: 70 Route: <sip:10.1.20.103;lr=on;did=a4d.ef1941a3> User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.007101 10.1.20.103:5060 -> 10.1.3.15:5061 BYE sip:111@10.1.3.15:5061 SIP/2.0 Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKad0c.d0bfffc.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Max-Forwards: 69 User-Agent: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.014754 10.1.3.15:5061 -> 10.1.20.103:5060 SIP/2.0 200 OK To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Via: SIP/2.0/UDP 10.1.20.103;branch=z9hG4bKad0c.d0bfffc.0 Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 Server: Linksys/SPA941-5.1.5 Content-Length: 0 # U +0.000198 10.1.20.103:5060 -> 10.1.3.18:5061 SIP/2.0 200 OK To: "111" <sip:111@10.1.20.103 <sip%3A111(a)10.1.20.103> >;tag=58af654bc0e634a2o1 From: <sip:112@10.1.20.103 <sip%3A112(a)10.1.20.103>>;tag=d091cd3cd91903c5i1 Call-ID: f4177d3f-86d1dd3e(a)10.1.3.15 CSeq: 101 BYE Via: SIP/2.0/UDP 10.1.3.18:5061;branch=z9hG4bK-73a35564 Server: Linksys/SPA941-5.1.5 Content-Length: 0

13 years, 11 months

sipp and stateful transaction problem

by JR Richardson

Hi All, I'm doing some testing with kamailio 1.5: kamailio1:/etc/kamailio# kamailio -V version: kamailio 1.5.4-notls (i386/linux) flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:6005M @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 Using dispatcher module trying to load balance SIP calls across some Asterisk servers. I have it working fine when I test in this scenario: sip phone dial out><asterisk><kamailio><round robin to several asterisk servers This works stateful and stateless, handles everything gracefully. This scenario is giving me fits: sipp dial out><kamailio><round robin to several asterisk servers I get retransmits on every call back to sipp with errors like "Discarding message which can't be mapped to a known SIPp call" and "SIP/2.0 481 Call leg/transaction does not exist" This happens with kamailio setup stateful or stateless. I'm wondering if sipp is the problem or just doesn't play well with kamailio? I've kept the config as simple as possible for testing, it is listed here http://pastebin.com/BZ8hJvJv Here is my sipp usage: sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 -trace_err Any insight would be appriciated. Thanks. JR -- JR Richardson Engineering for the Masses

13 years, 11 months

Need help with route statement

by Nelson Pereira

Hi all... Well I have made some progress... Bellow is my routing statement: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (src_ip==10.98.6.5) { if (dst_port==5065) { t_relay_to_tcp("10.98.118.20", "5065"); } else if (dst_port==5066) { t_relay_to_tcp("10.98.118.20", "5066"); } else if (dst_port==5067) { t_relay_to_tcp("10.98.118.20", "5067"); } else { t_relay_to_tcp("10.98.118.20", "5060"); } } else { t_relay_to_udp("10.98.6.5", "5060"); }; } When asterisk sends a call to kamailio, Kamailio then sends the invite to 10.98.118.20 via TCP on port 5061. INVITE sip:1989@10.98.6.5:5061 SIP/2.0 Record-Route: <sip:10.98.6.5:5065;transport=tcp;r2=on;lr=on> Record-Route: <sip:10.98.6.5:5061;r2=on;lr=on> Via: SIP/2.0/TCP 10.98.6.5:5065;branch=z9hG4bK74fd.a4578a84.0 Via: SIP/2.0/UDP 10.98.6.5:5060;branch=z9hG4bK7b9bb22d;rport=5060 From: "1103" <sip:1103@10.98.6.5>;tag=as4ae41ccf To: <sip:1989@10.98.6.5:5061> Contact: <sip:1103@10.98.6.5> Call-ID: 44073b911e86b0a96c9104cb7a5ec389(a)10.98.6.5 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date: Fri, 21 May 2010 12:23:58 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 232 P-hint: usrloc applied The problem is that the Kamailio receives a 302 Moved Temporarily with a contact field of CONTACT: <sip:1989@10.98.6.5:5065;transport=TCP> I need to have Kamailio, use this contact field and re-send the invite. How can this be done? Nelson Pereira Senior Network Specialist Protus<http://www.protus.com/> npereira(a)protus.com<mailto:name@protus.com> phone: 613.733.0000 ext.528 MyFax: 613.822.5083 MyFax.com<http://www.myfax.com/> | my1voice.com<http://www.my1voice.com/> | Campaigner.com<http://www.campaigner.com/>

13 years, 11 months

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sr-users May 2010