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37518 discussions

Help Asterisk with Kamailio unable to register with remote VOIP providers
by Manoj Gupta 02 Jan '17

02 Jan '17

Request to all - Please help we are BADLY stuck in this asterisk 11.13 and Kamailio 4.1 integration provided with Elastix MT Our Configuration is like this: Asterisk IP - 10x.3x.6x.236 port 5080 Kamailio IP - 10x.3x.6x.236 port 5060 Please see our kamailio.cfg attached. Issue 1: Issue is REGISTER messages sent by asterisk are not responded by kamailio asterisk is constantly doing a re-transmit on REGISTER messages. Retransmitting #1 (NAT) to 10x.3x.6x.236:5060: REGISTER sip:ims.airtel.in SIP/2.0 Via: SIP/2.0/UDP 10x.3x.6x.236:5080;branch=z9hG4bK40ac1ed8;rport Max-Forwards: 70 From: <sip:+91224400@ims.airtel.in>;tag=as565ef860 To: <sip:+91224400@ims.airtel.in> Call-ID: 7dd361ec25cb07344ea9df526e89ca8a(a)10x.3x.6x.236 CSeq: 182 REGISTER User-Agent: Asterisk PBX 11.13.0 Expires: 120 Contact: <sip:+91224400@10x.3x.6x.236:5080> Content-Length: 0 Issue 2: We are not able to see any log messages in /var/log/kamailio.log. We have done all kind of changes suggested on online blogs Why does kamailio not have a pre-set configuration as it is mostly used as sip proxy? Manoj K. Gupta

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Freezing for release v5.0.0
by Daniel-Constantin Mierla 02 Jan '17

02 Jan '17

Hello, I propose to freeze the development of v5.0.0 on January 4, 2017. Hopefully it is just in time for getting the frozen version in next Debian stable. Meanwhile, I hope we can get rid of mi code to have a proper fresh start with cleaned up code in 5.0.0. The big code tree restructuring was already done, config files in embedded languages is there as well (it needs more exports, but that's only about available time to fill some structures, not new feature development). If anyone wants to make other proposals, reply to mailing lists and let's discuss. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Happy 2017!
by Daniel-Constantin Mierla 01 Jan '17

01 Jan '17

A fresh year ahead, let it be a great one for all Kamailio friends! Wishing a prosperous and safe 2017 everywhere! Happy New Year! -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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TOPOS contact issue on RE-INVITE
by Daren FERREIRA 30 Dec '16

30 Dec '16

Hello, I made lots of tests in order to make Kamailio act as a topology hiding proxy, but i always got problems when using topos to store needed Route and Record-Route informations. I thought that was because of possible error(s) playing with : - socket binding - destination tweaking that may have lead to transaction breaking, so i decided to use NAT mode (even if that only can be temporary). My problem is the same. I have not found any way to influence TOPOS on the domain it use on the « Contact » header. Everything is getting well, iPs on Contact Header are toggled between external and internal sides, until RE-INVITE. From there, Contact domain are no more toggled, and so: - Public IP appear on Private side : i can manage my divide to force it to reply to the sender of the request instead of the contact - Private IP appear on Public side : i can’t do anything to make the external side ignore contact’s domain I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers? PUBLIC NETWORK SIDE RFC6598 NETWORK SIDE ————————— | | PROVIDER Y.Y.Y.Y———————X.X.X.X—| KAMAILIO ]—100.64.10.1———————— 100.64.10.2 ASTERISK (or anything else) | | ————————— 20:58:46.215911 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 1004 INVITE sip:+0598@100.64.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 246 v=0 o=root 79038203 79038203 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=audio 13564 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv 20:58:46.216642 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 375 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport=5060;received=100.64.10.2 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 0 20:58:46.222391 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 1040 INVITE sip:+0598@Y.Y.Y.Y:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 Max-Forwards: 69 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: <sip:+9997@voip-connect.fr;user=phone> Content-Type: application/sdp Content-Length: 264 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=root 79038203 79038203 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=audio 57810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=nortpproxy:yes 20:58:46.234592 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 290 SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE 20:58:46.548771 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 718 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:46.554903 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 719 SIP/2.0 183 Session Progress From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:47.735919 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:47.741229 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:48.050762 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:48.056175 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.811066 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 774 SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - 20:58:52.818381 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 775 SIP/2.0 200 OK From: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 To: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes 20:58:52.819315 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 446 ACK sip:atpsh-58656af1-769f-2@100.64.10.1 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK18c31028;rport Max-Forwards: 70 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Contact: <sip:+9997@100.64.10.2:5060> Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 20:58:52.825894 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 491 ACK sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.d149220aa711a5230da51bfea457d8bc.0 Max-Forwards: 69 From: "FAX" <sip:+9997@100.64.10.2>;tag=as4cfa9905 To: <sip:+0598@100.64.10.1;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> 20:59:00.388481 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 910 INVITE sip:btpsh-58656af1-769f-1@X.X.X.X <sip:btpsh-58656af1-769f-1@X.X.X.X> SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 273 Max-Forwards: 69 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: <sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp> Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=- 1510281183 168103905 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=image 17818 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.390679 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 420 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1;rport=5060 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 0 20:59:00.395723 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 890 INVITE sip:+9997@100.64.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 289 Max-Forwards: 68 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Contact: <sip:btpsh-58656af1-769f-1@X.X.X.X> v=0 o=- 1510281183 168103905 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=image 62130 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.397160 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 580 SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Length: 0 20:59:00.397344 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.405004 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 SIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes 20:59:00.896864 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:+9997@100.64.10.2:5060> Content-Type: application/sdp Content-Length: 266 v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy 20:59:00.904049 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 TSIP/2.0 200 OK From: <sip:+0598@Y.Y.Y.Y;user=phone>;tag=SDc8s1299-016500890000adc2 To: "FAX" <sip:+9997@X.X.X.X>;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: <sip:atpsh-58656af1-769f-2@100.64.10.1> v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes And so on until timeout and BYE

3 4

Astersik kamailio integration.
by Deep Pratap Singh Tomar 29 Dec '16

29 Dec '16

Hi I have modified the file so that if i send register request to kamailio it sud register on asterisk also. i have shared you the code but i cant see anyting happening in the asterisk server. please help me ASAP. L&T Technology Services Ltd www.LntTechservices.com<http://www.lnttechservices.com/> This Email may contain confidential or privileged information for the intended recipient (s). If you are not the intended recipient, please do not use or disseminate the information, notify the sender and delete it from your system.

2 1

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

3 4

String split function
by Grant Bagdasarian 29 Dec '16

29 Dec '16

Hi, Is there a string split function built into Kamailio for use in the cfg script? I'd like to split a string of key/value pairs inside the cfg script. For example: Key1=Value1;Key2=Value2; First split on ; Then on = I think I can use the String Transformation {s.select,index,separator}, but I was hoping there is a built in function. {param.valueat,index[, delimiter]} may also be an option. Regards, Grant

2 2

Re: [SR-Users] Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 29 Dec '16

29 Dec '16

I am attaching latest config along with kamailio log file and client trace. Please let me know how I can get Kamailio to advertise public ip in record route. I followed all the steps mentioned here - http://www.fredposner.com/1457/kamailio-behind-nat/ Thanks Pranathi From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 5:09 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Cc: Mick McGrellis <MMcGrellis(a)cyracom.com>; 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: Kamailio behind NAT, ACK to private IP not advertised public IP. I implemented full NAT logic as per the sample config. Still unable to resolve the issue. How do I let Kamailio change record_route header to use public ip address? Please help!!! (attached are latest scripts) From: Pranathi Venkatayogi Sent: Wednesday, December 28, 2016 12:39 PM To: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org>> Subject: Kamailio behind NAT, ACK to private IP not advertised public IP. Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

Kamailio behind NAT, ACK to private IP not advertised public IP.
by Pranathi Venkatayogi 28 Dec '16

28 Dec '16

Hi, I am encountering the same problem described in google groups<https://groups.google.com/forum/#!topic/2600hz-dev/-xvUZUrv4Y4>. However I dint not find any resolution hence writing again. 200 OK sent from the server has private Ip in its record route. As you see below, though the message is received on public IP (63.149.103.72) , the record route is set to private IP (172.31.211.31) I used listen with advertise of public IP, it did not work. Please find attached the config I am using. How do I change it send public ip only when talking to external world. Can someone point to me clear documentation how to configure Kamailio for NAT traversal. The following message is sent from Kamailio behind NAT to the public computer. 2016-12-27 17:19:24.526875 [blink.exe 5652]: RECEIVED: Packet 123, +0:08:42.690309 63.149.103.72:5061 -(SIP over TLS)-> 10.0.0.6:62912 SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.0.6:62912;rport=62912;received=50.175.10.190;branch=z9hG4bKPj2e381a96979945bd969989ffe9dca3a9;alias Record-Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> Call-ID: eb8670eec4354acdb69fd26f5625b75c From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 CSeq: 4665 INVITE Server: Blink 3.0.0 (Windows) Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Contact: <sip:75329410@10.0.27.108:61381;transport=tls> Supported: 100rel, replaces, norefersub, gruu Content-Type: application/sdp Content-Length: 355 v=0 o=- 3691844303 3691844304 IN IP4 10.0.27.108 s=Blink 3.0.0 (Windows) t=0 0 m=message 2855 TCP/TLS/MSRP * c=IN IP4 10.0.27.108 a=path:msrps://10.0.27.108:2855/261d3f47be25612cc77c;tcp a=accept-types:message/cpim text/* image/* application/im-iscomposing+xml a=accept-wrapped-types:text/* image/* application/im-iscomposing+xml a=setup:active -- The following is the ACK sent by public computer in reply to the above message. Note this message never reaches the Kamailio server as it is sent to private IP. 2016-12-27 17:19:24.526875 [blink.exe 5652]: SENDING: Packet 124, +0:08:42.690309 10.0.0.6:62944 -(SIP over TLS)-> 172.31.211.31:5061 ACK sip:75329410@10.0.27.108:61381;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.0.6:62944;rport;branch=z9hG4bKPj7df757862e6546beba18a646cb965ba2;alias Max-Forwards: 70 From: "cust1" <sip:cust1@translation.sms-test.cyracom.com>;tag=2f25d2ae690747c48c874e0b415ca03c To: <sip:spanish@translation.sms-test.cyracom.com>;tag=1c33ad41f6f44cae8ae8e060f30fe119 Call-ID: eb8670eec4354acdb69fd26f5625b75c CSeq: 4665 ACK Route: <sip:172.31.211.31:5061;transport=tls;lr;did=5.5e5> User-Agent: Blink 3.0.0 (Windows) Content-Length: 0 Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

1 0

BYE issue kamailio + rtpproxy "481 call leg transaction doesn't exists".
by Nihar Ranjan Deb 28 Dec '16

28 Dec '16

Hi All, I am a new user of kamailio. We have setup a kamilio server with RTP proxy. Most of the features are working fine except BYE. Say User A and B are registered in kamailio server. A (caller) connected to B(called), call established. 1. If Caller (A) disconnects the call, everything goes fine. B gets disconnected and call releases. 2. But if Called party (B) disconnects the call, A never gets disconnected. I checked trace as well, in this case caller (A) gets a BYE with B's callid. So A rejects call with "481 call leg transaction doesn't exists". Basically A party and B party has separate dialog established with Kamilio. So they have two separate callids . All messages are exchanged correctly. Only when B party disconnects call, kamailio sends BYE to A party with B party callid. Please help me to resolve this issue. Do let me know if any more information required. ------ INVITE-------------------------- INVITE sip:966268@*** SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:63238;branch=z9hG4bK-524287-1---e64d171ec3b0b54e;rport Max-Forwards: 70 Contact: <sip:334031@***:8906;rinstance=ea9b5f48eb342a98> To: <sip:966268@***> From: "334031"<sip:334031@****>;tag=4406552c Call-ID: 82158NzljY2NiYTA5YmI3M2U1MzQyODgwMWYwYTI3ZWJjNzc CSeq: 2 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE Content-Type: application/sdp Proxy-Authorization: Digest username="334031",realm="*****",nonce="****",uri="sip:966268@***",response=" ***",algorithm=MD5 Supported: replaces User-Agent: X-Lite release 4.9.6 stamp 82158 Content-Length: 334 v=0 o=- 13127393523589215 1 IN IP4 192.168.1.36 s=X-Lite release 4.9.6 stamp 82158 c=IN IP4 192.168.1.36 t=0 0 m=audio 57116 RTP/AVP 9 8 120 0 84 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:84 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv -------------------------BYE-------------------------------- BYE sip:334031@192.168.1.36:63238;rinstance=ea9b5f48eb342a98 SIP/2.0 Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* From: <sip:966268@***>;tag=1665200125 To: "334031" <sip:334031@***>;tag=4406552c Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Contact: <sip:10.0.0.1;line=sr-mcsjIlJcYlxcIPVo-0-8Ylb8-9bKUlPjIlxEYwbw> Max-Forwards: 69 User-Agent: Yealink SIP-T20P 9.72.0.80 Content-Length: 0 -------------------------- 481--------------------- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP ***:5060;branch=z9hG4bKa28b.a2ac8c26802cdcc8c619784643f82000.0;received=35.1 64.228.12 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKsr-HqsDUwx8-phRByVr-9VwUlbcUlPcIpKo-0noY0rjI95gmMhg10 qgY9mc-w5He0EJbTbkb3AZt2Yna6n4uPmqbJWEYwVKYlDTYlH* To: "334031" <sip:334031@***>;tag=4406552c From: <sip:966268@***>;tag=1665200125 Call-ID: !!:xePve9XKtEJ4NJJT-2hM9lsXe6RemqZPIRAa1PZf1wJ4DRx5ePRce0OwbHAY-RD4Hlya-HRwe MH4N6RemHYMI9Y0tV** CSeq: 3 BYE Accept-Language: en Content-Length: 0 Regards, Nihar

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