November 2014 - sr-users

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

rare crash , race condition between 183 and 408 ?
by Dragos Oancea 05 Nov '14

05 Nov '14

Hi all & hi Daniel , We had a crash with version 4.0.5 today . I looks like it happens due to a race condition between a provisional response (183) and the generation of a 408 . We send the INVITE, get a `100 Trying` from upstream , then we get a `183 Session Progress` from upstream , then after exactly 1 minute , we get another `183 Session Progress` from upstream. At this moment kamailio crashed. Some tm params we have : modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) <- our final reply timer is 1 minute . modparam("tm", "restart_fr_on_each_reply", 1) GDB output and logs here: kamailio 4.0.5 crash - in fake_req() - race cond - Pastebin.com kamailio 4.0.5 crash - in fake_req() - race cond - Paste... (gdb) bt #0 0x00007f4d93129a32 in memcpy () from /lib64/libc.so.6 View on pastebin.com Preview by Yahoo In fake_req() from t_reply.c , the pointer shmem_msg->new_uri.s which is passed to memcpy becomes invalid. memcpy( faked_req->new_uri.s, shmem_msg->new_uri.s,faked_req->new_uri.len); (gdb) p shmem_msg->new_uri $3 = {s = 0x0, len = 0} If this bug is not supposed to be fixed in 4.2.0 , we can help with debugging . Regards, Dragos

2 1

AliasDB module & Forking
by Marino Mileti 05 Nov '14

05 Nov '14

Hi guys, i've implemented a parallel forking using aliasdb module. Everything is working good if the first client (first row in dbaliases table) in online. When I try to make a call to a specific number three clients rings together. If the first client isn't online no call is forwarded to other clients :( I know that with alias_db_lookup() the first record from DB is the new R-URI and the others are added like branches. But if the R-URI in unreachable how can try to call the other branches? This is location script.. if(alias_db_lookup("dbaliases")){ $sht(a=>$ci) = $sel(contact.uri); } $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: xlog("L_INFO","No contact found\n"); exit; case -3: send_reply("404", "Erorr"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } Any suggest? My script still log "No contanct found" until timeout :( Many thanks ----- Marino Mileti -- View this message in context: http://sip-router.1086192.n5.nabble.com/AliasDB-module-Forking-tp132128.html Sent from the Users mailing list archive at Nabble.com.

2 3

DTMF Tones question
by Anthony Martin 05 Nov '14

05 Nov '14

HI, Please, excuse me for my bad English Just a question, kamailio does support DTMF Tones? i have a big while searching for documentation about that but without good results. I want to make a server side call transfer method whit DTMF Tones... Thanks so much for your good work Greetings from Venezuela

3 2

DBURL
by shmick＠riseup.net 05 Nov '14

05 Nov '14

[i have no idea about email subject nor anything else for this except prior i configured rtpproxy to work with kamailio on a unix socket and updated tls.cfg to use my cert/key] running kamailio from deb http://deb.kamailio.org/kamailio jessie main $ sudo service kamailio restart [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(15228) INFO: tls [tls_init.c:385]: init_tls_compression(): tls: init_tls: disabling compression... 0(15228) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 30-34: syntax error 0(15228) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 30-34: Invalid arguments 0(15228) : <core> [cfg.y:3439]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 353, column 35: ERROR: bad config file (3 errors) 0(15228) WARNING: <core> [ppcfg.c:219]: pp_ifdef_level_check(): different numbe[FAILpreprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 1 ... failed! failed! /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", *DBURL*) <--------not happy with DBURL here modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif

2 3

dnssec module
by shmick＠riseup.net 05 Nov '14

05 Nov '14

running kamailio from deb http://deb.kamailio.org/kamailio jessie main when i add the dnssec module to /etc/kamailio/kamailio.cfg kamailiop fails to start and gives: [FAIL] Not starting Kamailio SIP Server: invalid configuration file! ... failed! failed! [....] 0(8839) ERROR: <core> [sr_module.c:597]: load_module(): could not open module </usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so>: /usr/lib/x86_64-linux-gnu/libval-threads.so.14: undefined symbol: SSL_get_SSL_CTX 0(8839) : <core> [cfg.y:3436]: yyerror_at(): parse error in config file /etc/kamailio/kam[FAIL.cfg, line 239, column 12-22: failed to load module ERROR: bad config file (1 errors) ... failed! failed! $ sudo ls -la /usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so -rw-r--r-- 1 root root 22912 Oct 17 09:04 /usr/lib/x86_64-linux-gnu/kamailio/modules/dnssec.so any help appreciated thanks

2 4

REGISTRAR location replication (pua_reginfo)
by lejarreta.e＠sarenet.es 04 Nov '14

04 Nov '14

Good morning. We've got a problem with location memory and database replication between "n" kamailios as REGISTRAR servers. 1.- WHAT WE WANNA DO: - To have "n" Kamailios with their DB and memory "user location" replicated. - Each Kamailio owns its own DB.(It could be also with a common DB). - Each kamailio has a PUA_REGINFO, PRESENCE, and REGISTRAR roles. - USRLOC Module with db_mode = 2 (Write-Back scheme). 2.- WHAT WE DON'T WANNA DO: - USRLOC Module with db_mode != 2 (Write-Back scheme). 3.- BACKGROUND: - RFC3265 - RFC3680 - RFC3903 - I attach a small architecture graph. 4.- PROBLEMS: 4.1 Deadlock: Each REGISTER involves a NOTIFY to the other side. Each NOTIFY involves another NOTIFY to the other side. There should be a mechanism to avoid PUBLISH reginfo_handle_notify() function calls but to PUBLISH save() calls on pua_reginfo module. Any idea would be appreciated (also to workaround our problem). 4.2 PUA_REGINFO module notify.c Just FYI. We think It doesn't comply to RFC3680. It should observe "unkown-param" object to aid "Contact" header params such as "Path", "Received", "User_agent" and so on. We've changed code but for a fast startup we've also avoided "unkown-param" and now we're able to see replicated "Path", "Received ... params on each DB. Thanks and best regards. Eduardo Lejarreta.

2 1

how to configure multi-proxy mode for kamailo 4.2.x
by andrew 04 Nov '14

04 Nov '14

hi By default, kamailio.cfg is used for one proxy. How to set up kamailio to support multi-proxy, so that serveral kamaiio servers work together, and each one is responsible for sip clients in some specific area? B.R. Andrew

2 1

"Help with dialog $dlg_var(cseq_diff)"
by Yuriy Gorlichenko 03 Nov '14

03 Nov '14

Hello. I need to increment CSeq value for INVITE with Auth params when use UAC_AUTH for outgoing calls to provider. Kamailio 4.2 may increment this using dialog module http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for… Now I experements with this and var $dlg_var(cseq_diff) and at transaction way int have NULL value. I can not understand why My config is: # ----- dialog params modparam("dialog", "db_url",DBURL) modparam("dialog", "db_mode", 1) modparam("dialog","table_name","dialog") modparam("dialog", "dlg_flag", 4) modparam("dialog", "initial_cbs_inscript", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "default_timeout", 60) modparam("dialog", "track_cseq_updates", 1) route config is at request route if(is_method("INVITE") && !has_totag()){ $dlg_ctx(timeout_route) = "DIALOG_END"; $dlg_ctx(timeout_bye) = 1; dlg_manage(); xlog("L_INFO","Dialog manage is {$ct}\n"); } } t_relay(); next I handle failure reply because 407 reply is recieved failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_check_status("401|407")){ xlog("L_INFO", "Reply from provider on failure: $tU"); xlog("L_ERR","401/407 - Unauthorized. ($ci .) ($rm) from ($fu) (IP:$si:$sp) to ($Ri:$Rp). Must be authorized with digest Auth."); avp_print(); xlog("L_INFO", "CSeq diff: $dlg_var(cseq_diff)"); uac_auth(); xlog("L_INFO", "UAC_AUTH(): $tU"); append_branch(); t_relay(); } if (t_is_canceled()) { exit; } } As you may see i logging $dlg_var(cseq_diff) value and now it NULL. So I can not understand why? What wrong I do? Thanks for advice.

2 3

Problems using htable iterator
by Glenn Larsen 03 Nov '14

03 Nov '14

Hi, I have a hard time using the htable iterator... I have this simple setup: modparam("htable", "htable", "h1=>size=4") ... sht_iterator_start("i1", "h1"); while (sht_iterator_next("i1")) { xlog("L_ERR", " SS: $shtitkey(i1) and $shtitval(i1)\n"); } sht_iterator_end("i1"); ... Now to put some simple data, I use: # sudo kamcmd htable.sets h1 test 123 # sudo kamcmd htable.dump h1 { entry: 8 size: 1 slot: { item: { name: test value: 123 } } } When the route is "executed", I get this in my log file: Nov 3 15:16:47 beast /sbin/kamailio[24865]: ERROR: SS: <null> and <null> The $shtitkey(i1) and $shtitval(i1) is always null. Is it a bug? Or did I manage to do something wrong? -- Best regards, Glenn O. Larsen -- Best regards, *Glenn O. Larsen* *CTO* *SimCom AS* Professor Birkelands vei 24c N-1081 Oslo Direct: +47 21 09 34 12 Mobile: +47 997 00 000 Online profile: http://simcom.no/glenn.larsen *We simplify your communication ...*

2 1

rtpengine does not send RTP packets out
by Juan Perez 03 Nov '14

03 Nov '14

Hi, I have kamilio-4.2 and rtpengine running on the same machine.I have this scenario: softphone --> Kamailio with Rtpengine --> AsteriskThe softphone initiates the call, it is sent to the Asterisk. I can see the SDPs being re-written with the new IP/Ports provided by rtpengine: Invite from Kamailio to Asterisk200 Ok from Kamailio to Softphone However, I take a signaling/media capture on the server where the kamailio/rtpengine are running and see the RTP coming from both endpoints (softphone and asterisk) to the correct ports but there is no packets coming out from the proxy to either direction. I see these 2 lines on the rtpengine log and make me think that something prevents the rtpengine to stream out to the 2 endpoints: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Final packet stats: this is the log from the rtpengine: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686: offer - { "sdp": "v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 99.153.161.148#015#012s=-#015#012c=IN IP4 99.153.161.148#015#012t=0 0#015#012m=audio 16386 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012", "direction": [ "pub", "pub" ], "replace": [ "origin", "session-connection" ], "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", " ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... 99.153.161.148" ], "from-tag": "73rH43p60NeKm", "command": "offer" } Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Creating new call Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Opened ports 35022..35023 for media relay Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Opened ports 35038..35039 for media relay Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Returning to SIP proxy: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate: ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ... NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: Got valid command from 127.0.0.1:46686: answer - { "sdp": "v=0#015#012o=root 439969655 439969655 IN IP4 205.10.248.62#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 205.10.248.62#015#012t=0 0#015#012m=audio 11142 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012", "direction": [ "pub", "pub" ], "replace": [ "origin", "session-connection" ], "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", "205.10.248.62" ], "fr ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: ... om-tag": "73rH43p60NeKm", "to-tag": "as57e7b8e3", "command": "answer" } Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] Returning to SIP proxy: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4 54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 213070 ... Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd] ... 6430 54.173.60.13 35039 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Write error on RTP socket Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Final packet stats: Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] --- Tag 'as57e7b8e3', created 0:00 ago, in dialogue with '73rH43p60NeKm' Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35022 <> 205.10.248.62:11142, 0 p, 0 b, 0 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35023 <> 205.10.248.62:11143 (RTCP), 0 p, 0 b, 0 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] --- Tag '73rH43p60NeKm', created 0:00 ago, in dialogue with 'as57e7b8e3' Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35038 <> 99.153.161.148:16386, 0 p, 0 b, 1 e Nov 1 18:59:26 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] ------ Media #1, port 35039 <> 99.153.161.148:16387 (RTCP), 0 p, 0 b, 0 e Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Got valid command from 127.0.0.1:46686: delete - { "call-id": "0866b358-dc9c-1232-1399-3767db69b8dd", "received-from": [ "IP4", "205.10.248.62" ], "from-tag": "as57e7b8e3", "command": "delete" } Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] [0866b358-dc9c-1232-1399-3767db69b8dd] Call-ID to delete not found Nov 1 18:59:36 ip-10-0-2-68 rtpengine[27764]: [0866b358-dc9c-1232-1399-3767db69b8dd port 35038] Returning to SIP proxy: d7:warning38:Call-ID not found or tags didn't match6:result2:oke this is the log from kamailio (showing only the lines that include the word rtpengine): Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: TO_FS: Starting rtpengine session. Invite Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[TO_FS] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=441 a=25 n=rtpengine_offer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]: check_content_type(): type <application/sdp> found valid Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d3:sdp529:v=0#015#012o=- 314474204173996793 6663595733338147866 IN IP4 54.173.60.13#015#012s=-#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35022 RTP/AVP 114 0 8 101#015#012a=rtpmap:114 ilbc/8000#015#012a=fmtp:114 mode=30#015#012a=rtpmap:0 pcmu/8000#015#012a=rtpmap:8 pcma/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012a=rtcp:35023#015#012a=ice-ufrag:oPm3I7OE#015#012a=ice-pwd:WjZnh7dnhfdkfoXDoqDbnCLLr7N7#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35022 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35023 typ host#015#0126:result2:oke Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: INFO: <script>: mylog: Starting rtpengine session. Answer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:onreply_route=[TO_FS] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=476 a=25 n=rtpengine_answer Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine_funcs.c:148]: check_content_type(): type <application/sdp> found valid Nov 1 18:59:26 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d3:sdp500:v=0#015#012o=root 439969655 439969655 IN IP4 54.173.60.13#015#012s=Asterisk PBX 1.8.10.0#015#012c=IN IP4 54.173.60.13#015#012t=0 0#015#012a=ice-lite#015#012m=audio 35038 RTP/AVP 0 101#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012a=silenceSupp:off - - - -#015#012a=ptime:20#015#012a=sendrecv#015#012a=rtcp:35039#015#012a=ice-ufrag:cBL9SIw9#015#012a=ice-pwd:QHw7VHp83nMsLpD8ZoSqtDxGchN4#015#012a=candidate:NZ1IXXQHYxOora9j 1 UDP 2130706431 54.173.60.13 35038 typ host#015#012a=candidate:NZ1IXXQHYxOora9j 2 UDP 2130706430 54.173.60.13 35039 typ host#015#0126:result2:oke Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: exec: *** cfgtrace:request_route=[LOOSE_ROUTE] c=[/usr/local/kamailio-install-dir//etc/kamailio//kamailio.cfg] l=488 a=24 n=rtpengine_delete Nov 1 18:59:36 ip-10-0-2-68 ./kamailio[28001]: DEBUG: rtpengine [rtpengine.c:1448]: rtpp_function_call(): proxy reply: d7:warning38:Call-ID not found or tags didn't match6:result2:oke

2 4

signaling for NON SIP environment
by fb111＠web.de 03 Nov '14

03 Nov '14

2 1

question
by Javier Ricke 03 Nov '14

03 Nov '14

i work with asterisk, but i have problem at the moment to send audio rtp, because ip is private, kamailio will could solution this problem? regards

2 1

mysql db error
by shmick＠riseup.net 03 Nov '14

03 Nov '14

2 1

Address already in use error when starting Kamailio service
by Mahmoud Ramadan Ali 03 Nov '14

03 Nov '14

Hiiiiii everyone, I'm receiving this error when starting Kamailio saying that it can not resolve my ip addresses used by Kamailio and another "127.0.0.1:5060 : Address already in use " error message...so how can i fix this please ? Nov 2 15:39:26 Kamailio /usr/local/sbin/kamailio[7692]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 35 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.100.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.50.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.100.1 Nov 2 15:39:44 Kamailio kamailio: WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.50.1 Nov 2 15:39:44 Kamailio /usr/local/sbin/kamailio[7726]: ERROR: <core> [tcp_main.c:2876]: tcp_init(): bind(b, 0x7faebd135ca4, 16) on 127.0.0.1:5060 : Address already in use

3 3

Kamailio behind NAT - Wrong VIA Header
by Marko Seidenglanz 03 Nov '14

03 Nov '14

Hello, We are using Kamailio 4.2 and RTPEngine 3.3 hosted on a VM behind NAT in Google Cloud Engine. Asterisk 11 + Kamailio + RTPEngine are on the same machine and Kamailio is used as WebRTC Gateway. INVITES, that are rewritten by Kamailio look like the following: INVITE sip:eTden7KoNSdMVCUNYIwmz@wh2.24dial.com SIP/2.0 Record-Route: <sip:0.0.0.0:5064;lr=on;nat=yes> Via: SIP/2.0/UDP 0.0.0.0:5064 ;branch=z9hG4bK86f5.88fb7f8810c91758b5f3962420d3a4ef.0 Via: SIP/2.0/UDP 10.240.1.95:5060 ;received=127.0.0.1;branch=z9hG4bK05952ee6;rport=5060 Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as7900b63f To: <sip:eTden7KoNSdMVCUNYIwmz@wh2.24dial.com> Contact: <sip:anonymous@10.240.1.95:5060;alias=127.0.0.1~5060~1> Call-ID: 0662d336694c60ed60ed5310571a1aee@10.240.1.95:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Mon, 03 Nov 2014 09:01:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676 P-hint: outbound As you can see, Kamailio uses 0.0.0.0 for Record-Route as well as for VIA Header. It uses the address configured in kamilio.cfg (listen=udp:0.0.0.0:5064) I know, that we could use the advertise parameter to tell kamailio the external IP. The problem is, that IPs in GCE are dynamically assigned to the VMs. Thus static configuration of external IP is not sufficient for our setup. Is there a way, to let Kamailio gather it's external IP dynamically. (E.g. using a STUN server or something like that?) Another possibility would be to tell Kamailio it's external IP via a start parameter. Is that possible? Any help would be grateful. Kind regards, Marko Seidenglanz

2 1

using time stamp
by bob sacamano 03 Nov '14

03 Nov '14

hi, I am trying to see how long kamailio takes to process messages I saw the commands: set_time_stamp diff_time_stamp in older versions of kamailio but they dont seem to be available. I have been searching for info but have been unable to find anything. Is the command no longer available or is there some other way to use? thanks

2 3

FIFO Error When Running Kamctl dispatcher reload
by Mahmoud Ramadan Ali 02 Nov '14

02 Nov '14

Hiiiii Everyone I have encountered that error whenever i try to use the #Kamctl dispatcher reload even thought that i have the "modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")" and loadmodule "mi_fifo.so" configured...any ideas how to solve this issue ?! root@Kamailio:/usr/local/etc/kamailio# kamctl dispatcher reload ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module.

2 1

Re: [Kamailio-Users] /usr/local/sbin/kamdbctl create error
by vivi 01 Nov '14

01 Nov '14

Hi all.. I have solve the problem . Cheers, vivi From: vivi [mailto:vivi.hilton@gmail.com] Sent: Saturday, June 06, 2009 11:29 PM To: 'users(a)lists.kamailio.org' Subject: /usr/local/sbin/kamdbctl create error Hi all I have specify the wanted db type (DBENGINE=MYSQL) in the /usr/local/etc/kamailio/kamctlrc then I using "/usr/local/sbin/kamdbctl create" to create MySQL database, but I got this error: ERROR: database engine not specified, please setup one in the config script root@acer:/usr/local/src/kamailio-1.5.0/sip-server# vim /usr/local/etc/kamailio/kamctlrc root@acer:/usr/local/src/kamailio-1.5.0/sip-server# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Usage: grep [OPTION]... PATTERN [FILE]... Try `grep --help' for more information. /usr/local/lib/kamailio/kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) ERROR: Creating core database and grant privileges failed! Cheers, vivi

2 1

Trouble with LCR removing $ru after upgrading to 4.1.6
by Will Ferrer 01 Nov '14

01 Nov '14

Hi We are having an odd issue with our LCR. This started after we upgraded from kamailio 4.1.0 to 4.1.6. Here is the situation: We populate our gw_uri_avp (stored in: $avp(i:709)) as follows: 16|1||||[gateway ip address]||5060||1|1. The $ru before running next gateway shows: sip:[destination number]@[our domain that the call came into]. This is how it should look before lcr occurs. Then we run next_gw(). We get these message in the logs during the running of next_gw DEBUG: lcr [lcr_mod.c:2156]: generate_uris(): r_uri <*1*[destination number]@[gateway ip address]:5060*1*>, dst_uri <> DEBUG: lcr [lcr_mod.c:2437]: next_gw(): added ruri_user_avp <[destination number]> DEBUG: lcr [lcr_mod.c:2473]: next_gw(): added flags_avp <1> DEBUG: lcr [lcr_mod.c:2480]: next_gw(): added tag_avp <> DEBUG: lcr [lcr_mod.c:2488]: next_gw(): added defunct_gw_avp <1253613379> The main thing to notice there is that there is now a 1 at the start and end of the r_uri shown in generate_uris. I am fairly certain those are new, and maybe related to what happens next. Immediately following the next_gw function running the $ru becomes: <null>, so of course the call doesn't go through. I have looked at the code for the lcr module and it looks like it wasn't changed in version 4.1.6. I found alot of other notes in the change log that had to do with the way uris are handled so it is possible one of those changes caused the problem. I am stumped currently. Any help any one can offer me would be very appreciated. All the best. Will Ferrer Swithsoft Inc

2 4

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sr-users November 2014