sr-users November 2014

sr-users@lists.kamailio.org

91 participants
129 discussions

Re: [SR-Users] RPC uac.reg_info hangs second time
by Dan Christian Bogos 06 Nov '14

06 Nov '14

Hey Daniel, Thanks for your fast reply! I am using the 4.2.0 out of debian packages (not nightly). Will shift on nightly and retest. Not a big deal regarding 150 seconds, maybe it should be a good reason for async reloads (reply with the delay number of seconds when reload will happen). Reloads can be always implemented outside of kamailio but since everybody would need it I assume a more centralized approach would be useful. Anyway, thanks again! DanB On 24.10.2014 12:00, sr-users-request(a)lists.sip-router.org wrote: > Hello, > > are you using branch 4.2 or some tgz/package from 4.2.0 release? There > was a fix related to such issue, so 4.2 branch must be used (or debian > nightly builds for 4.2). > > I will have to check the code how flexible that is, but afaik, the limit > on reload interval was thought to avoid sending again registration > requests while retransmissions from previous one can occur. > > Cheers, > Danel

2 7

Kamailio v4.1.7 Released
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, Kamailio SIP Server v4.1.7 stable release is out. This is a maintenance release of the previous stable branch, 4.1, that includes fixes since release of v4.1.6. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.7. For more details about version 4.1.7 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/11/kamailio-v4-1-7-released/ Note: the latest stable branch is 4.2, at this moment with its latest release v4.2.0. See more details about it at: * http://www.kamailio.org/w/2014/10/kamailio-v4-2-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

how to check if reply is retransmission?
by Juha Heinanen 06 Nov '14

06 Nov '14

is there a means to check if reply to t_relay()ed request is retransmission? there is t_is_retr_async_reply(), but that does not apply to regular transactions. i would like to avoid double calling of rtpengine_answer(). -- juha

4 5

AT&T and UAC cancel after 200 OK
by Yuriy Gorlichenko 06 Nov '14

06 Nov '14

Hello. I use UAC module for trunks and have trouble when call to mobile endpoint of AT&T provider. When endpoint pickup call ans 200 OK reply come into kamailio, kamailio send CANCEL. so at the endpoint I see than call is dropped and at kamailio client I hear voicemail speech form AT&T endpoint. Provider say that his trace is OK... My sip trace bellow: IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1129 E.......@.l. ...6........q..INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 651 E.......*.Z.6... .........}.SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", qop="auth" Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 410 E.......@.o. ...6.........j%ACK sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.0 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com >;tag=04e2a294d0728b89107e081a19babab4.43a7 Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 ACK Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6..........`INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 1402 E.......@.k. ...6.........._INVITE sip:98765432100@phone.myprovider.com SIP/2.0 Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Contact:<sip:12345678@sip.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Mon, 03 Nov 2014 20:21:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 314 Proxy-Authorization: Digest username="12345678", realm="phone.myprovider.com", nonce="VFfk3VRX47HuWpmSTOv4DPvdulTv2Eeq", uri=" sip:98765432100@phone.myprovider.com", qop=auth, nc=00000001, cnonce="2107612791", response="51db231dedb4a74396447729a192a8d3", algorithm=MD5 v=0 o=root 1363870848 1363870848 IN IP4 68.34.22.11 s=Asterisk PBX 12.5.0 c=IN IP4 68.34.22.11 t=0 0 m=audio 30030 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:30031 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xiSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 529 E..-....*.[.6... .........xkSIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102;rport=1024;received=68.34.22.11 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Server: kamailio (4.1.2 (x86_64/linux)) Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 998 E.......*.Y16... .........^.SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 986 E.......*.Y<6... ...........SIP/2.0 180 Ringing Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Length: 0 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1294 E..*....*.X.6... .........:.SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1282 E.......*.X.6... ........ ..SIP/2.0 183 Session Progress Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=UrBQrZp9vSm3p Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 103 INVITE Contact: <sip:98765432100@173.203.60.50:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 Remote-Party-ID: "98765432100" <sip:98765432100@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415020637 1415020638 IN IP4 173.203.60.50 s=FreeSWITCH c=IN IP4 173.203.60.50 t=0 0 m=audio 25430 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W.6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 412 E.......@.o. ...6.........j.CANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1.cs102 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6...........ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.0ec383b6dc8fbb1861dfe644f92ee9ca.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK010bb317;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W~6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........jACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.6a1e71bf5fcd88f137733279fa31770c.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK1f95579c;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 246 E.......*.\.6... ...........OPTIONS sip:68.34.22.11:1024 SIP/2.0 Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 330 E..f....@.p. ...6........R.<SIP/2.0 405 Method not allowed Via: SIP/2.0/UDP 10.202.129.254:5060;branch=0;rport=5060;received=34.43.4.23 From: sip:keepalive@myprovider.com;tag=29ebf5d To: sip:68.34.22.11:1024;tag=c6894543322e2a4942d921e4298ce904.1ec7 Call-ID: 6360a4bb-53bebf7-5686ca6(a)10.202.129.254 CSeq: 1 OPTIONS Server: MS Lync Content-Length: 0 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W|6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........eACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.47488c128d3518b837410e363acc963e.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK5e2d222b;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200 IP 34.43.4.23.5060 > 10.0.2.4.5068: UDP, length 1428 E.......*.W{6... .........H.SIP/2.0 200 OK Via: SIP/2.0/UDP sip.myservice.info:5068 ;rport=1024;received=68.34.22.11;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.2.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK7d9dfadc;rport=50600 Record-Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Record-Route: <sip:sip.myservice.info:5068;nat=yes;ftag=as288f6857;lr=on> From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@phone.myprovider.com>;tag=etQmF8eNZ7NKm Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 104 INVITE Contact: <sip:98765432100@98.129.251.83:5060;transport=udp> User-Agent: myprovider Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-myproviderOutboundGateway: sip:+98765432100@67.231.8.195 X-myproviderOutboundCarrierID: 29927192767592 X-myproviderCarrierRate: 0.00900 X-myproviderCloudRate: 0.00300 Remote-Party-ID: "Outbound Call" <sip:5060@phone.myprovider.com >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1415023959 1415023960 IN IP4 98.129.251.83 s=FreeSWITCH c=IN IP4 98.129.251.83 t=0 0 m=audio 22108 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 647 E.......@.n. ...6..........;ACK sip:98765432100@98.129.251.83:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.00885667802b30315ce7f8b5db31e2bd.0.cs102 Via: SIP/2.0/UDP 10.0.1.41:50600 ;received=10.0.1.41;branch=z9hG4bK6081ac89;rport=50600 Route: <sip:34.43.4.23;lr=on;ftag=as288f6857;did=229.d4f> Max-Forwards: 70 From: "John Connor" <sip:skynet.device-3@10.0.1.41:50600>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068>;tag=etQmF8eNZ7NKm Contact: <sip:skynet.device-3@10.0.1.41:50600> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 105 ACK User-Agent: Asterisk PBX 12.5.0 Content-Length: 0 IP 10.0.2.4.5068 > 34.43.4.23.5060: UDP, length 406 E.......@.o. ...6.........?xCANCEL sip:98765432100@phone.myprovider.com SIP/2.0 Via: SIP/2.0/UDP sip.myservice.info:5068 ;branch=z9hG4bK3f75.164056a067d4ce9fa68c2c2d3bd734b6.1 Max-Forwards: 70 From: "John Connor" <sip:12345678@phone.myprovider.com>;tag=as288f6857 To: <sip:98765432100@ints.myservice.info:5068> Call-ID: 6717947d5e5f445314ba762f154be72f@10.0.1.41:50600 CSeq: 102 CANCEL Content-Length: 0 Reason: SIP;cause=200

1 0

how to use mqueue.
by 张顺通 06 Nov '14

06 Nov '14

whle (mq_fetch("QUEUE")) { if (yes) { do something } else { mq_add("QUEUE"); add back to queue. } } can mq_add use here? I tried it, it do not work. and How to determine the end?

3 6

Releasing v4.1.7
by Daniel-Constantin Mierla 06 Nov '14

06 Nov '14

Hello, I plan to release a new version (4.1.7) of the previous stable branch 4.1. Most probably on Thursday this week. If anyone has remarks/issues about it, write back to mailing lists to discuss. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 24-27, Berlin - http://www.asipto.com

1 1

benchmark module
by bob sacamano 05 Nov '14

05 Nov '14

hello, i was wondering where does this module write the logs? Nothing shows up in my syslog. If i replace the benchmark parts in the route with xlog for example xlog( "L_INFO", "$Ts " ); this shows up fine in syslog heres what u have in my config file: modparam("benchmark", "enable", 1) modparam("benchmark", "granularity", 100) modparam("benchmark", "loglevel", 3) if(is_method("REGISTER")) { bm_start_timer("test"); save("location"); bm_log_timer("test"); exit; } I have been searching but cant find any info on it

2 5

Kamailio with media proxy and two network interfaces
by Julien 05 Nov '14

05 Nov '14

Hello, I have a problem with Kamailio and a setup with two network interfaces (eth0 and eth1) Devices on eth0 CANNOT see devices on eth1 and vice versa, there is no route between both interfaces. Only Kamailio and the media proxy can access both. +--------------------------+ | kamailio + rtpproxy | eth0 | + openvpn | eth1 +---------+ +-----------+ | | | | | +--------------------------+ | +-+----+ +----+-+ | A | | B | | | | | +------+ +------+ A user connected at eth0 shall be able to get the presence status of users on eth1 and users shall be able to call each other. What I did so far: I added listen statements for both interfaces to the Kamailio config file: -listen=udp:eth0_IP:5060 -listen=udp:eth1_IP:5060 What's not working: When I try to make a call from user in A domain to another user in B domain, the call established correctly, but no stream running. On B device I see an incoming call from 'sip:user@A_Domain'. B user doesn't have route for A Domain, so that doesn't work. Is what I want technically possible with Kamailio. If yes, what do I have to change in my config? Do you have any suggestions? Thanks in advance. -- Julien Barreau

3 3

how you make Kamailio with mariadb
by ANTHONY HESNAUX 05 Nov '14

05 Nov '14

hello Daniel I want to install Kamailio with mariadb on CENTOS 7 I already install Kamailio on CENTOS 6.5 without error but on CENTOS 7 with mariadb, What modules do you include for make cfg: make include_modules="db_mysql" cfg thank you best regard Anthony HESNAUX

3 8

UAC error at reply : t_should_relay_response:
by Yuriy Gorlichenko 05 Nov '14

05 Nov '14

ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 Hello. I use UAC module for trunks form 4.2 master branch. I have an error when endpoit at the trunk picked up. OK reply never replied to caller because I see next error: ERROR: t_should_relay_response: status rewrite by UAS: stored: 491, received: 200 So as I see before reply 200 sended to kamailio (from callee) I see packet with 491 SIP code. It is strange because at the another server (same configuration) OK successfully sended to caller and session is Ok...

1 0

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sr-users November 2014