- sr-users - kamailio.org

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

1 0

rtpengine destroys voice branch on multiple branch call
by Yuriy Gorlichenko 27 Dec '16

27 Dec '16

Hi. I have 2 endpoints 1001 and 1002 1001 registered from 1 device 1002 registered from 2 devices at the same time When I called to 1002 kamailio makes 2 branches rtpengine_manage command called from branch rout for handling every branch directly (it can be different endoint types (ws/tls/udp) for each branch) When i picking up at the 1002 on one device server sends CANCEL (answered elswhere) to another devices of 1002 rtpengine_manage reacts at the CANCEL and deletes this brach after 30 seconds My question is: What is solution for use rtpengine_manage and call branches together?

1 0

set_body_multipart and append_body_part
by Diego Nadares 26 Dec '16

26 Dec '16

Hi guys, I'm trying to append a content type to a message that has only one content type. I tried using just set_body_multipart and now I'm using set_body multipart and append_body_part with no success too. The result is a message with multipart but has "two" sdp and no gtd. It has the old sdp with no ip changed for rtpproxy and the new sdp with ip changed BUT with no content type. And the message has not the end boundary (--unique-boundary-1.) Kamailio log has an info and a warning Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: INFO: <core> [msg_translator.c:1693]: get_boundary(): Content-Type hdr has no params <application/sdp> Dec 26 11:36:22 dwrfsd01 ./kamailio[8847]: WARNING: <core> [msg_translator.c:1959]: build_req_buf_from_sip_req(): check_boundaries error kamailio.cfg ...... route[RELAY]{ $avp(ch) = 1 .... if ($avp(ch)){ set_body_multipart(); msg_apply_changes(); $var(gtd) ="IAM,\r\nGCI,asdafsdfasd\r\n\r\n"; append_body_part("$var(gtd)", "application/gtd", "signal;handling=optional"); } .... } Ngrep U 2016/12/26 11:36:22.496634 172.16.213.21:5060 -> 172.16.213.38:5060 ... Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. U 2016/12/26 11:36:22.688458 172.16.213.38:5060 -> 172.16.208.11:5060 .... Content-Type: multipart/mixed;boundary="unique-boundary-1". Mime-Version: 1.0. Remote-Party-ID: <sip:1152565064@172.16.213.38:5060 >;party=calling;privacy=off;screen=no. . --unique-boundary-1. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. . --unique-boundary-1. v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 58724 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=sendrecv. a=rtcp:58725. a=ice-ufrag:hhXDwrco. a=ice-pwd:5temIXP8veOGuD8Dsllm5HXkWU. Any help will be very very appreciated! Thanks in advance. Diego

2 4

help configure dispatcher in Amazon ec2
by huan nguyen duy 26 Dec '16

26 Dec '16

Hi, I’m very appreciated if someone can figure out this problem for me. My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system like this: soft phone 1(Linphone) —————> | Load balancer —————> | Sip Server huancomputer | private IP: 172.31.11.35 | 172.31.13.113 soft phone 2(Linphone) —————> | public IP: 35.166.179.244 | 35.166.152.104 huannguyen | | The dispatcher.lst config file in /usr/local/etc/kamailio as below: # gatewayss 1 sip:172.31.13.113:5060;transport=udp <sip:172.31.13.113:5060;transport=udp> 0 0 abc=xyz User records subscriber table: ——————————————————————————————————————————————————— id | username | domain ——————————————————————————————————————————————————— 70 | nguyenhuan | nguyenhuan(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> 71 | huancomputer | huancomputer(a)ec2-35-166-179-244.us-west-2.compute.amazonaws.com <mailto:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> Location records from location table: ——————————————————————————————————————————————————— username | domain | contact ——————————————————————————————————————————————————— huancomputer | NULL | sip: huancomputer(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:64166;transport=udp nguyenhuan | NULL | sip: nguyenhuan(a)14.161.41.70 <mailto:huancomputer@14.161.41.70>:33267;transport=udp But soft phone 1 cannot make a call to soft phone 2, below is data is extracted from wireshark: Load balancer: -------------------------------- No. Time Source Destination Protocol Length Info 177 4.463129 14.161.41.70 172.31.11.35 SIP/SDP 1433 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 179 4.463367 172.31.11.35 14.161.41.70 SIP 452 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 180 4.463421 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1215 32.552942 14.161.41.70 172.31.11.35 SIP 1105 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1216 32.553170 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1219 32.555588 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1220 32.555706 172.31.11.35 14.161.41.70 SIP 593 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1609 42.603150 14.161.41.70 172.31.11.35 SIP 1113 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1610 42.603364 172.31.11.35 172.31.13.113 SIP 1269 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1613 42.605800 172.31.13.113 172.31.11.35 SIP 749 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1614 42.605932 172.31.11.35 14.161.41.70 SIP 599 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1688 44.425237 172.31.11.35 14.161.41.70 SIP 471 Status: 408 Request Timeout | No. Time Source Destination Protocol Length Info 1689 44.425285 172.31.11.35 172.31.13.113 SIP 491 Request: CANCEL sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 1691 44.426018 172.31.13.113 172.31.11.35 SIP 519 Status: 200 canceling | Data from Sip Server: ------------------------------- No. Time Source Destination Protocol Length Info 237 6.242928 172.31.11.35 172.31.13.113 SIP/SDP 1589 Request: INVITE sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:nguyenhuan@ec2-35-166-179-244.us-west-2.compute.amazonaws.com> | No. Time Source Destination Protocol Length Info 243 6.244409 172.31.13.113 172.31.11.35 SIP 602 Status: 100 trying -- your call is important to us | No. Time Source Destination Protocol Length Info 244 6.244541 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 288 6.710414 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 328 7.710454 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 396 9.710411 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 547 13.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 688 17.710406 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 835 21.710381 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 985 25.710385 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1130 29.710407 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1279 33.710400 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1303 34.332395 172.31.11.35 172.31.13.113 SIP 1261 Request: REGISTER sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com <sip:ec2-35-166-179-244.us-west-2.compute.amazonaws.com> (1 binding) | No. Time Source Destination Protocol Length Info 1312 34.334415 172.31.13.113 172.31.11.35 SIP 743 Status: 200 OK (1 binding) | No. Time Source Destination Protocol Length Info 1459 37.710402 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | No. Time Source Destination Protocol Length Info 1609 41.710395 172.31.13.113 14.161.41.70 SIP/SDP 1720 Request: INVITE sip:nguyenhuan@14.161.41.70:33267;transport=udp <sip:nguyenhuan@14.161.41.70:33267;transport=udp> | Huan Nguyen, --------------------------------------- Mobile app developer Phan Thi Company Sai Gon, VietNam Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 1

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 24 Dec '16

24 Dec '16

While waiting for Santa, it's a good time to also express my thanks and greetings to all the friends, developers and community members that made 2016 another amazing year for Kamailio project. Enjoy the winter holidays together with your beloved ones! Merry Christmas! Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

Merry Christmas to all Kamailians
by Olle E. Johansson 24 Dec '16

24 Dec '16

Friends, Trying to calm down and prepare for Christmas with the family. It’s been another great year for Kamailio and I’m proud to be part of the Kamailio development community. We’ve made great releases, had a great conference and overall done good stuff :-) For all of you that celebrate Christmas - a Merry Christmas! To the rest of you: Happy Holidays! A big Thank You to all developers and a special giga-Thank You to Daniel for all the time you spend working with the code, helping out in discussions on the lists and in the bug tracker. And to all of you in the community - Thank You for being part of this! Greetings from a cold Sweden, currently without snow in my area just north of Stockholm. We celebrate on Christmas Eve (tomorrow) so there’s a lot of food being prepared right now and the last minute shopping is just about to start... Keep your Kamailio running! /Olle PS. If you are not currently following the project on Twitter - it’s time! Find us as @kamailioproject

3 2

baresip version 0.5.0 released
by Alfred E. Heggestad 23 Dec '16

23 Dec '16

Hi, my apologies for the off-topic posting but I think this could be interesting for the SIP Community. Baresip version 0.5.0 is now released, you can find the announcement email here: http://lists.creytiv.com/pipermail/re-devel/2016-December/001154.html Merry Xmas from Norway! /alfred

1 0

Re: [SR-Users] rtjson + mhomed and record route (corrected)
by Diego Nadares 23 Dec '16

23 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The call is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header? ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } .... Thanks in advance. Diego 2016-12-22 12:08 GMT-03:00 Diego Nadares <dnadares(a)gmail.com>: > Hi list, > > We have two eth, one with a private ip and the other with a public ip. We > have MHOMED configured. The call comes from private network and then is > routed to a public network via rtjson routing. The is routed but the thing > is that the record route is being set with the private one (It's the same > when calls come from public and is routed to a private network, record > route has the public and not the private). > > With mhomed configured shouldn't Record Route be updated as VIA header > > ".... When activated, sip-router will select a socket that can reach the > destination (to be able to connect to the remote address). (sip-router > opens a UDP socket to the destination, then it retrieves the local IP which > was assigned by the operating system to the new UDP socket. Then this > socket will be closed and the retrieved IP address will be used as IP > address in the Via/Record-Route headers)..." > > This is part of ngrep > > U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 > INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. > To: <sip:1111111111@172.16.213.38:5060>. > Call-ID: 1-13024(a)172.16.213.21. > CSeq: 1 INVITE. > Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. > .... > > U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. > .... > > U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 > INVITE sip:1111111111 <011%201111-1111>@XXX.XXX.XXX.02:5060 SIP/2.0. > *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* > Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f > Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 > > This is my cfg > > mhomed=1 > > request_route { > > > # per request initial checks > route(REQINIT); > > # NAT detection > #route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # handle retransmissions > if(t_precheck_trans()) { > t_check_trans(); > exit; > } > t_check_trans(); > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are > routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) { > record_route(); > } > > > Thanks in advance. > > Diego. > >

1 1

kamailio dispatcher call flow example
by huan nguyen duy 23 Dec '16

23 Dec '16

Hi, I want to load balance multiple kamailio server using dispatcher module but not success at configuration. I’m not clearly how it work. Someone can show me an example call flow like below i’m very appreciated. Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| | | | 407 F2 | | | |<---------------| | | | ACK F3 | | | |--------------->| | | | INVITE F4 | | | |--------------->| INVITE F5 | | | 100 F6 |--------------->| INVITE F7 | |<---------------| 100 F8 |--------------->| | |<---------------| | | | | 180 F9 | | | 180 F10 |<---------------| | 180 F11 |<---------------| | |<---------------| | 200 F12 | | | 200 F13 |<---------------| | 200 F14 |<---------------| | |<---------------| | | | ACK F15 | | | |--------------->| ACK F16 | | | |--------------->| ACK F17 | | | |--------------->| | Both Way RTP Media | |<================================================>| | | | BYE F18 | | | BYE F19 |<---------------| | BYE F20 |<---------------| | |<---------------| | | | 200 F21 | | | |--------------->| 200 F22 | | | |--------------->| 200 F23 | | | |--------------->| | | | | Huan Nguyen, ——————————————————— Mobile app developer Phan Thi Company Sai Gon, VietNam

2 1

rtpproxy/rtpengine. Help to understand.
by Igor Olhovskiy 22 Dec '16

22 Dec '16

Hi! Issue I can’t figure out. Or all working ok and that’s just me who not understands. I have situation softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4) All addresses are on same public network. rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3 I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage(). But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3 If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working) So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something? If any - can provide pcap’s with both cases. Regards, Igor

3 5

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } Thanks in advance. Diego.

1 0

Socket in rtjson module
by Diego Nadares 22 Dec '16

22 Dec '16

Hi Guys, Anybody knows if each 'socket' specified in routes struct should be use to send the messages? The socket is configured in kamailio. In kamailio.cfg listen=udp:172.16.213.38:5060 json struct "routes": [ { "uri": "sip:127.0.0.1:5080", "dst_uri": "sip:127.0.0.1:5082", "path": "<sip:127.0.0.1:5084>, <sip:127.0.0.1:5086>", ----> "socket": "udp:172.16.213.38:5060",<---- "headers": { "from": { "display": "Alice", "uri": "sip:alice@127.. In our kamailio has two eth and I want to select which one to use but I can't make it work through rtjson. In the code of the module I can't find where is being assigned the socket. Thanks in advance! Diego

2 3

kamailio dispatcher can not forward call in Amazon EC2
by huan nguyen duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information. Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 0

kamailio dispatcher can not forward call in Amazon EC2
by Huân Nguyễn Duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information.

1 0

Memory leak in Kamailio 4.4.4 TLS Module
by raja raja 22 Dec '16

22 Dec '16

Hi, shared memory not freed once TLS connection close means shm available memory decreasing for new connection that leads to the following error *tls: ssl bug #1491 workaround: not enough memory for safe operation* *TestCase:* 1. start kamailio 2. use sample program to establish tls connection and close imediately. 3. Kamailio Log is For first TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63604096 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63545192 For second TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63545192 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63489352 *Memory Leak:* for tls connection connection 63604096 - 63545192 = 58904 bytes not freed. for tls connection connection 63545192 - 63489352 = 55840 bytes not freed. I tested same scenario on kamailio - 4.2.4, 4.3.4 versions result is same. Could you please provide the fix? Thanks & Regards, Raja Chegondi.

2 1

register_timer cb function does not have access to shared memoery var
by Andy 22 Dec '16

22 Dec '16

in mod_init I have something like list = (struct list_t*) shm_malloc ..... this list is available from any functions in the module, now there is a timer registered, in the same module register_timer(callback_f, 0 , 5) the callback_f function when tries to use list, always sees as empty -- View this message in context: http://sip-router.1086192.n5.nabble.com/register-timer-cb-function-does-not… Sent from the Users mailing list archive at Nabble.com.

2 2

new URI [] shorter than old URI - error
by Nelson Migliaro 21 Dec '16

21 Dec '16

Hello, My SIP vendor request me to replace the from in order to authenticate calls. I use: uac_replace_from("111111111", "sip:111111111@vendor-domain.com"); And when finishing calls I have this error: ERROR: uac [replace.c:607]: restore_uri(): new URI [] shorter than old URI [sip:34912345678@vendor-public-ip] Information ------------------------------------------------------------------ 34912345678 -> original CLI 111111111 -> my vendor contract vendor-domain.com -> domain of the vendor ------------------------------------------------------------------ I use the function uac_replace_from in the request_route My Kamailio version is: 4.2.8-1.1 Thank you Nelson.-

1 0

kamailio 4.4 segmentation fault when dp_translate match dialplan with ftmatch operator and empty match expression
by Julia Boudniatsky 21 Dec '16

21 Dec '16

version: kamailio 4.4.4 id:852c47 ENTRY: { PRIO: 16765956 MATCHOP: 2 *MATCHEXP: * MATCHLEN: 0 SUBSTEXP: REPLEXP: ATTRS: 1 } Please see generated core file in extension. Thank you, Julia.

2 7

nathelper - variable natping_interval per registration
by Phil Lavin 20 Dec '16

20 Dec '16

Hi all, Is it possible to set the natping_interval to a different value, per registration, at the time the user registers? For example, user A has 10 seconds and user B has 60 seconds. Thanks

3 3

auth result -3
by Jon Bonilla (Manwe) 19 Dec '16

19 Dec '16

Hi all I'm getting an auth challenge result of -3 and I don't know what it means as it's not listed in the auth module doc. Here's syslog' output: db_postgres [km_dbase.c:230]: db_postgres_submit_query(): sending query ok: 0x7f36d1f75f00 (2) - [select secret from sip_buddies where username='48481628780'] db_postgres [km_res.c:108]: db_postgres_get_columns(): 1 columns returned from the query db_postgres [km_res.c:133]: db_postgres_get_columns(): RES_NAMES(0x7f36d1f92238)[0]=[secret] db_postgres [km_res.c:166]: db_postgres_get_columns(): use DB1_STRING result type db_postgres [km_res.c:211]: db_postgres_convert_rows(): no rows returned from the query auth_db [authorize.c:199]: get_ha1(): no result for user '48481628780@' INFO: <script>: Ha fallado la auth con resultado -3 cheers Jon

3 4

INFO: jsonrpcs used instead of mi_fifo in the default kamailio.cfg and kamctl
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, with MI code planned to be removed, I updated the kamctl and default kamailio.cfg to use jsonrpcs (jsonrpc server) module instead of mi_fifo to interact with kamailio at runtime. The default fifo file for rpc command is /var/run/kamailio/kamailio_rpc.fifo and expects JSONRPC commands as described at: - http://kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idp31525708 Switching back to mi_fifo is still possible by loading the module and editing kamctlrc, but the module might not make it to 5.0, so for new installations and testing is better not to use it. Few rpc commands are missing in carrierroute, otherwise the kamctl commands should work just as before, but be aware that the output is now in json format. Testing of the master branch with the latest kamctl and providing feedback will be really appreciated -- I could do tests only for the modules I typically use. 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 0

Forking with 404 replies
by José Seabra 19 Dec '16

19 Dec '16

Hello there, I have a Kamailio server acting as registrar server and when the subscriber has more than 1 AOR, kamailio sends an INVITES to each AOR, if one of them replies 404 what should be the kamailio behavior: 1. should Kamailio reply with an ACK and not pass the 404 to UAC? 2. should Kamailio forward the 404 to UAC? I'm asking this because i'm facing the second behavior, that is, kamailio is forwarding the 404 to UAC and the call is terminated. -- Regards José Seabra

2 3

RFC: renaming the modules with - (dash) in name
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, given the work for 5.0 which will bring major changes on source tree structure, I am considering to remove the - (dash) from the modules that have it in the name. There are 4 of them now: cpl-c janssonrpc-c jsonrpc-c jsonrpc-s One of the reasons is that this kind of name doesn't work for the feature of defining MOD_modname for each loaded module, because - is a separator for ids in config language grammar, so for example, MOD_jsonrpc-c cannot be used in an #!ifdef. More over, for embedded interpreters, most of exported functions there use kamailio module name as package name, and again this will break. Of course, a different package name can be defined, but won't be coherent anymore. Anyone against this renaming? 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

Re: [SR-Users] Question on multiple 200OK received after forking
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, well, the specs say that all 200ok must be forwarded to the caller, because a middle point cannot decide for the user making the call. Even sending an ACK for the 200ok from a middle point is not enough because the callee of that branch will think it is in an active call. The caller can take different actions, the usual is to ACK the 2nd 200ok and then immediately send BYE. But it can also decide to end previous branch and switch to this one (e.g., uses better codecs, or has video, etc...). Cheers, Daniel On 14/12/2016 15:05, Marino Mileti wrote: > > Hi guys, > > > > I've a Kamailio instance listening on both eth0 and wlan0. > > Kamailio forks call to two branches, so both clients rings together. > > > > In case (very very very rare J) of simultaneous answers is there any > way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to > arrange a semaphore with lock/unlock capabilities of cfgutils, I've > used the drop command but client still sending the 200OK because it's > waiting for an ACK L > > > > Any help will be very appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it/ <mailto:marino.mileti@alice.it>// > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 0

(Correction) unusual characters added to reINVITE message
by AL S 19 Dec '16

19 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

2 1

Limiting calls based on CLI and dst number
by Cibin Paul 19 Dec '16

19 Dec '16

Hi, Is there a way to limit the number of active calls per cli or dst number basis? Yesterday my system had a ddos attack in which 123456 (CLI) was hitting 2345 (DID) at a rate of 100-200 calls per minute. I would like to restrict to 2-4 simultaneous calls per CLI/DST. Any help would be appreciated. Thanks, Cibin

4 9

http_async_client - queries
by Ramachandran, Agalya (Contractor) 17 Dec '16

17 Dec '16

Hi team, We are trying to use kamailio for our project purpose, which will be used as a proxy server. Am trying to use "http_async_client" module to send my rest queries asynchronously. I have few queries regarding the usage of http_async_query() and using PUT method. http_append_header("Authorization: Bearer $var(auth)"); http_append_header("Content-Type: application/json"); http_set_method("PUT"); http_async_query("http://sdk-evm-wcdc-c-001.rtc.sys.comcast.net/events/createxmpprooteventfor…", "{'from_tn':'$fU','to_tn':'$tU','callid':'ci'}", "HTTP_REPLY"); Here am passing from, to, callid in post data field. Passing as 'key':'value' is mandatory or I can pass only value- ? In the file, async_http.c - in async_send_query(), if I modify the post data to the format I need, I see the value is truncated up to three fields only. How can I achieve that my payload format whatever I form in async_send_query(),doesn't get truncated and will be passed to the server-? Can you please explain how this works-? Also, I observed that kamailio is restricting the content-length of HTTP request to 256. Is this something modifiable-? If so please guide me how this can be achieved. Regards, Agalya

2 2

sipp dead calls
by Diego Nadares 16 Dec '16

16 Dec '16

Hi guys, We are testing kamailio with sipp. We are running it with 20cps and some calls do the following. When Kamailio is processing the 'ringing' a '200ok' arrives in the middle. First, kamailio forwards the 200 ok and then the ringing. 'ACK' arrives and, I suppose, the call is established. The thing is than when the 'BYE' arrives kamailio responds with a 404. This is the summary of the call Id Time Source Destination Protocol Len Info 6483 2016-12-14 11:26:06.264697 SIPP-A KAMAILIO SIP/SDP 692 Request: INVITE sip:11111111@172.16.213.38:5060 | 6484 2016-12-14 11:26:06.264937 KAMAILIO SIPP-A SIP 367 Status: 100 trying -- your call is important to us | 6485 2016-12-14 11:26:06.266327 KAMAILIO SIPP-B SIP/SDP 1179 Request: INVITE sip:22222222@172.16.213.31:5060 | *6486 2016-12-14 11:26:06.267217 SIPP-B KAMAILIO SIP 566 Status: 180 Ringing | * 6487 2016-12-14 11:26:06.267268 SIPP-B KAMAILIO SIP/SDP 733 Status: 200 OK | 6488 2016-12-14 11:26:06.267758 KAMAILIO SIPP-A SIP/SDP 788 Status: 200 OK | *6489 2016-12-14 11:26:06.267833 *KAMAILIO SIPP-A* SIP 442 Status: 180 Ringing | * 6490 2016-12-14 11:26:06.268868 SIPP-A KAMAILIO SIP 493 Request: ACK sip:127.0.0.8;line=sr-N6IAzBFwMJZfWJZLM.M7MlF-W.y6Mx14NEt7Nw05NhPQKjaP | 6491 2016-12-14 11:26:06.269162 KAMAILIO SIPP-B SIP 609 Request: ACK sip:172.16.213.31:5060;transport=UDP | 6492 2016-12-14 11:26:06.269614 SIPP-A KAMAILIO SIP 404 Request: BYE sip:11111111@172.16.213.38:5060 | 6493 2016-12-14 11:26:06.269782 KAMAILIO SIPP-A SIP 348 Status: *404 Not here* | We are using modules rtjson, evapi, uac, topoh, rtpproxy for all calls. My debug level is -1. With higher levels this behavior increase. Kamailio is running in a virtual machine with centos7 with 8 cores and 8gb of ram. Do you need any further information? I can send you a pcap or ngrep file. Best regards, Diego.

2 3

unusual characters added to reINVITE message by Kamailio
by AL S 16 Dec '16

16 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

1 0

ACK not routed to destination
by Wim Van Cauwenbergh 16 Dec '16

16 Dec '16

Hi, I have setup kamailio with the dispatcher module and have a problem with the ACK in response to 200OK not being forwarded by kamailio: Scenario is very simple: sipp(.219) --> kamailio (.220) --> sipp(.239) SIP trace: root@DNS-enum-8Gb-3:/var/log# ngrep -W byline -d eth0 port 5060 interface: eth0 (10.57.26.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # U 10.57.26.219:5060 -> 10.57.26.220:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.239:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Record-Route: <sip:10.57.26.220;lr>. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.219:5060 -> 10.57.26.220:5060 ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 ACK. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. XContactinOK: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # *U 10.57.26.219:5060 <http://10.57.26.219:5060> -> 10.57.26.220:5060 <http://10.57.26.220:5060>* *ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0.* *Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5.* *From: sipp <sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>>;tag=15698SIPpTag021.* *To: sut <sip:service@10.57.26.220:5060 <http://sip:service@10.57.26.220:5060>>;tag=14783SIPpTag018.* *Call-ID: 1-15698(a)10.57.26.219 <1-15698(a)10.57.26.219>.* *CSeq: 1 ACK.* *Contact: sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>.* *Max-Forwards: 70.* *Subject: Performance Test.* *XContactinOK: <sip:10.57.26.239:5060;transport=UDP>.* *Content-Length: 0.* . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@10.57.26.219:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. *kamailio.conf:* # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", # "mysql://openser:openserro@localhost/openser") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; } # 1= only check the header RURI, 1126 checks some other general stuff if(!sanity_check("1126", "1")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing xlog("#######WITHINDLG:hastotag $si:$sp\n"); if (loose_route()) { xlog("#######WITHINDLG:looseroute $si:$sp\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("#######WITHINDLG:ACK transaction found $si:$sp\n"); # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. xlog("#######WITHINDLG:ACK no transaction found so discard $si:$sp\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Debug: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:10.57.26.239:5060 ;transport=UDP> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-15698-1-5>; state=16 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sl [sl_funcs.c:412]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=14783SIPpTag018 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [57]; uri=[sip:service@10.57.26.220:5060] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [sut < sip:service@10.57.26.220:5060>] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: siputils [checks.c:106]: has_totag(): totag found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:hastotag 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:929]: loose_route(): There is no Route HF Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=8 T start=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=53919, isACK=1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=9 T end=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:ACK no transaction found so discard 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)

3 3

Siremis installation with postgresql.
by José Seabra 16 Dec '16

16 Dec '16

Hi there, I'm facing some issues trying to install siremis with postgresql. The file Config.xml has the correct configuration for postgresql driver but It seems that during installation phase siremis can't create database structure for postgresql. Anyone have installed siremis with postgresql, if yes can you share the steps needed for this installation. -- Cumprimentos José Seabra

4 6

kamcli - transferring the github repository to kamailio organization
by Daniel-Constantin Mierla 16 Dec '16

16 Dec '16

Hello, kamcli (kamailio command line interface tool written in python) got a consistent set of features trying to become in pair with kamctl. It is currently hosted by asipto organization on github, but I am considering to transfer (move) it to kamailio organization in order to manage easier the access for kamailio devs as well as increase its exposure to community. It interacts with kamailio using rpc commands via jsonrpcs module. You can read more about what it can do at: - https://github.com/asipto/kamcli The commands are implemented in a plugin-like fashion -- see the ones done so far: - https://github.com/asipto/kamcli/tree/master/kamcli/commands Given that it can handle better the jsonrpc responses, provides more options to format the output and the help is embedded in each command, I think it is a good candidate to replace the old shell(bash)-based kamctl. Probably we can adjust kamctl for rpc when mi code will be removed, but the options to build and extend in long term a good cli for kamailio are quite limited in shell. For the sake of an example, here is the implementation of 'ps' command, handling the json rpc to display a very compact list of kamailio processes: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_ps.py Dealing with database records is also rather trivial: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_group.py Anyone with comments/suggestions/alternatives? 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

Forwarding Emergency calls based on caller IP address
by João Vitor Torres 15 Dec '16

15 Dec '16

Hello all, I am looking for a solution to forward emergency calls inside an enterprise network. Is there any way to forward calls based on the IP address of caller device? The requirement is that calls to a unique emergency number must be forwarded to specific local numbers in each building, and the IP address of the caller is the criteria for choosing the destination. A table maps IP nets to specific local numbers. The SIP contact header has the caller IP address, but I am new to Kamailio and I am not sure where to start. Any suggestions? Thanks in advance. João Vitor.

2 1

Re: [SR-Users] htable lock - unlock
by Daniel-Constantin Mierla 15 Dec '16

15 Dec '16

Hello, the log message you refer to is inside the if block: if ($sht(pht=>$ci)!=$null) { ... } but you don't set the item in memory, no assignment like: $sht(pht=>$ci) = ...; The sht_lock(pht=>$ci) is just locking the slot where the value of $sht(pht=>$ci) will be stored, but don't store any valye for that variable. Cheers, Daniel On 15/12/2016 11:46, Marino Mileti wrote: > > Hi guys, > > > > I'm going out of mind J > > I've this piece of script inside an "on_reply_route" that captures > only 200OK of INVITE (i would like to save in ht the contact of who > has answered a call) > > > > sht_lock("pht=>$ci"); > > if ($sht(pht=>$ci)!=$null){ > > xlog("L_INFO","Before assignment.."); > > $sht(pht=>$ci)=$var(myvar); > > xlog("L_INFO","Assigned done..."); > > sht_unlock("pht=>$ci"); > > } > > else{ > > xlog("L_INFO","Skipping.."); > > } > > sht_unlock("pht=>$(hdr(Call-ID))"); > > > > The situation is that I never see the "Assigned done" message in log > file... > > > > Is this the right use of sht_lock/sht_unlock method? > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

2 1

Updating to SIREMIS 4.3
by Kevin Greene 15 Dec '16

15 Dec '16

Hello fellow SIP people: I upgraded from SIREMIS 4.2 to 4.3 for my Kamailio system and I now get the error message shown below. Any ideas on how to solve it? I am assuming there was a change on how SIREMIS did a date query and my version of Linux (6.8) or my PHP does not support it as expected. Any help would be useful. Thanks. Kevin [{"target":"ERROR","content":"\n[2016-12-14 22:59:55 (GMT)] An exception occurred while executing this script: \nError message: #2, date(): It is not safe to rely on the system's timezone settings. You are *required* to use the date.timezone setting or the date_default_timezone_set() function. In case you used any of those methods and you are still getting this warning, you most likely misspelled the timezone identifier. We selected 'UTC' for 'UTC\/0.0\/no DST' instead<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php:160<\/font> \n *function:<\/b> errorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/sysheader.inc 117\n *function:<\/b> userErrorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \n *function:<\/b> date ( \"m\/d\/Y\" ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php 160\n The "date" command returns: $ date Thu Dec 15 01:13:41 UTC 2016 The /etc/localtime entry is: $ more /etc/localtime TZif2 UTC $ which php /usr/bin/php $ php -version PHP 5.3.3 (cli) (built: Aug 11 2016 13:38:15) Copyright (c) 1997-2010 The PHP Group Zend Engine v2.3.0, Copyright (c) 1998-2010 Zend Technologies PHP I am using is the latest patched version I have. ***

3 2

htable lock - unlock
by Marino Mileti 15 Dec '16

15 Dec '16

Hi guys, I'm going out of mind J I've this piece of script inside an "on_reply_route" that captures only 200OK of INVITE (i would like to save in ht the contact of who has answered a call) sht_lock("pht=>$ci"); if ($sht(pht=>$ci)!=$null){ xlog("L_INFO","Before assignment.."); $sht(pht=>$ci)=$var(myvar); xlog("L_INFO","Assigned done..."); sht_unlock("pht=>$ci"); } else{ xlog("L_INFO","Skipping.."); } sht_unlock("pht=>$(hdr(Call-ID))"); The situation is that I never see the "Assigned done" message in log file... Is this the right use of sht_lock/sht_unlock method? Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Kamailio 4.3 Rewriting From and db_alias values
by Gonzalo Gasca Meza 15 Dec '16

15 Dec '16

Hi forum, This is the scenario: VM <- Kamailio -> SP1 ----- SP2 ---- PhoneA (john) PhoneB PhoneA is configured in *dbaliases* table: alias_username, alias_domain, username, domain 11111111, sp1.com, john, sp2.com When PhoneB dials 1111111, Kamailio converts 111111(a)sp1.com to john(a)sp2.com Now I need to provide VM services to PhoneA. (Opposite direction) PhoneA calls voicemail but the calling number is john(a)sp2.com I need to have SP1 use the 11111111(a)sp1.com format How can I overwrite the From? Pseudo algorithm ------------------------ If destination is VM and from domain ($fU) is sp2.com: Find user URI: ( In this case john extracted from john(a)sp2.com in dbaliases) If user is found: rewrite From URI: use alias_username@alias_domain I'm using default kamailio.cfg and calls successfully are forwarded to VM Thank you

2 1

Question on multiple 200OK received after forking
by Marino Mileti 14 Dec '16

14 Dec '16

Hi guys, I've a Kamailio instance listening on both eth0 and wlan0. Kamailio forks call to two branches, so both clients rings together. In case (very very very rare J) of simultaneous answers is there any way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to arrange a semaphore with lock/unlock capabilities of cfgutils, I've used the drop command but client still sending the 200OK because it's waiting for an ACK L Any help will be very appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

question on call cancel
by Kelvin Chua 14 Dec '16

14 Dec '16

I have a peculiar scenario, maybe you guys have an easy solution: 1. a call comes in through an SBC (which is also kamailio) 2. SBC passes call to core kamailio server 3. call reaches 180 ringing state 4. client suddenly changes from wifi to 4G (using 4G IP, VIA headers are all updated) 5. client sends cancel and receives cancelling msg from SBC 6. SBC sends cancel to kamailio core 7. kamailio core sends back Request Terminated after termination the issue is, this message was sent to the wifi IP not the 4G when checking the network dumps, looks like this was because the INVITE was received on kamailio core from the old wifi IP. Is there maybe a way to update the transaction perhaps? Kelvin Chua

2 2

Multi-Domain dispatcher
by Satish Patel 14 Dec '16

14 Dec '16

We have currently dispatcher running for single domain, but in future we have more domain coming so i want to do multi domain dispatching for example dispatcher redirect request for foo.com to foo.registar and bar.com to bar.registar SIP server currently we have following code in dispatcher logic. what should we do to handle multi-domain if(!ds_select_dst("1", "2")) { # if we are here that means no destination is available. We notify the user by 404 and exit the script. xlog("L_NOTICE", "No destination available!"); send_reply("404", "No destination"); exit; }

2 1

regular expression question
by Satish Patel 14 Dec '16

14 Dec '16

I am trying to block SIP scanner so i am trying to use following logic only allow "My-UserAgent" and block rest but its throwing error, but if i use =~ regular expression which works! why negative match doesn't work? if($ua !~ "(My-UserAgent") { xlog("L_INFO","On more scriptkiddie, coming from $si, blocking"); exit; } what could be wrong here?

3 3

event for new tcp connection from client
by Andy 14 Dec '16

14 Dec '16

similar to tcp closed, can this be achieved when a client is connect to tcp -- View this message in context: http://sip-router.1086192.n5.nabble.com/event-for-new-tcp-connection-from-c… Sent from the Users mailing list archive at Nabble.com.

3 6

Kamailio TM - t_set_fr
by José Seabra 13 Dec '16

13 Dec '16

Hello there, I'm using t_set_fr(0,2000) on calls to registered phones but by any reason this isn't working, the call only enters on the failure route after 30 seconds in case of no response from SBC/Phone, anyone has any idea what can be the reason for that? if(lookup("location")) { xlog("L_INFO", "routing to registered phone ruid=$ruid R=$ru - ID=$ci \n"); t_set_fr(0, 2000); t_on_failure("ASBC_FAILURE"); t_on_branch("ASBC_BRANCH"); route(RELAY); exit; } My RELAY route: route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } version: kamailio 4.4.2 (x86_64/linux) Thanks -- Cumprimentos José Seabra

2 11

Accessing SDP changes
by Pete Kelly 13 Dec '16

13 Dec '16

Hi I am using Kamailio 4.4. and calling the function sdp_remove_transport in order to remove some SRTP media lines. The function works (of course!) however I would like to get access to the modified SDP within the cfg. I thought this would do the trick: sdp_remove_transport("RTP/SAVP"); sdp_get("$avp(foo)"); xlog("$avp(foo)"); However $avp(foo) does not contain the newly modified SDP. Is there a way for me to access it? I have been looking at things like t_save_lumps() and msg_apply_changes(). Any pointers or confirmation that this is or isn't possible would be very welcome! Thanks Pete

3 5

Tsilo iOS Push Notifications with Kamailio/Asterisk Integration
by Benjamin Fitzgerald 12 Dec '16

12 Dec '16

Hello, my goal is to address iOS 10's removal of periodic TCP connections in iOS apps with their replacement PushKit, a push-based alternative all VoIP apps should implement. In order to have iOS apps receive SIP calls in the background, I believe I need to use tsilo to suspend the INVITE transaction send a push notification to the device. I have been following the presentation here ( http://www.kamailio.org/events/2015-KamailioWorld/Day2/20-Federico.Cabiddu-…) however I am not sure how to integrate it with Asterisk ( http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb ). My questions can be summarized below: (1) What changes need to be made from Federico's presentation to use with the Kamailio/Asterisk integration? (2) How can I modify the config to send push notifications for EVERY invite? This is to address the scenario where a device goes offline before the SIP registration expires and a call is received within that window. iOS is very aggressive about killing background applications so it's likely as soon as user leaves the VoIP app, the device will become unreachable. Thanks for any help. Benjamin Fitzgerald LETS Corporation (925) 235-1154 ben(a)letscorp.us

2 3

DTMF Using INFO
by Nahum Nir 12 Dec '16

12 Dec '16

Hello, I tried Googling but didn'tfind anything. Is it possible to support DTMF using INFO? Thanks, Nir

4 8

msilo timestamp
by Slava Bendersky 12 Dec '16

12 Dec '16

Hello Everyone, What format stored inc_time in msilo module ? I am trying convert it to human readable format, but time looks UTC SELECT FROM_UNIXTIME($(xavp(dbquery[$var(i)]=>inc_time)), '%M %D %Y %h:%i:%s') Slava.

2 1

Error using Siremis 4.3.0 with Kamailio 4.4 and PHP 7
by Diogenes Aviles 09 Dec '16

09 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in page "Administration-User Management". When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/ openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 I verified and the file "/var/www/siremis-4.3.0/openbiz/bin/Resource.php" exist. But I don't have any idea where the error is. Some help will be appreciate. Regards -- Diogenes

2 2

Asterisk Security Advisory (AST-2016-009)
by Matthew Jordan 09 Dec '16

09 Dec '16

Hey all - The Asterisk project just released a security advisory for a security vulnerability in which Asterisk using chan_sip with a proxy can allow for unauthenticated calls. This affects all supported versions of Asterisk (11, 13, 14). Since that may be relevant to those on this mailing list who are not also on the asterisk-users mailing list, I thought it prudent to mention it here as well. A description of the vulnerability follows: Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue does not affect you. The announcement can be seen here: http://lists.digium.com/pipermail/asterisk-announce/2016-December/000662.ht… Thanks again to Walter Doekes for reporting the vulnerability and providing the patch to fix it. Matt -- Matthew Jordan Digium, Inc. | CTO 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org

3 2

IP-based security + registrations
by JLL 09 Dec '16

09 Dec '16

Hello, I would like to know if anyone can help me accomplish the following: I have a Kamailio server (4.2.3 on CentOS 7 x64) and 5 Asterisk servers (1.6.2.13 . I know, it's old but we have a lot of custom modules written for it). I need for certain IP's to be "trusted" (able to send INVITE's to K from the internet - these would be our origination providers), but any other IP's that come along would need to register before sending an INVITE (the REGISTER would be sent to any of the Asterisk servers for validation, if this is possible). Can this be done? Thank you!

2 1

Siremis manual to learn
by Saranga Nagesh 08 Dec '16

08 Dec '16

Hi, I have installed kamailio and Siremis in centos but I don't know how to use Siremis. Can you forward me manual or link to learn about Siremis. Thanks & Best Regards, Nagesh Saranga

2 1

Action: restructuring the source code tree
by Daniel-Constantin Mierla 07 Dec '16

07 Dec '16

Hello, today I am going to do the restructuring of the source code tree as discussed previously. - http://lists.sip-router.org/pipermail/sr-dev/2016-December/037715.html If anyone has patches to push using existing structure, do it before 12:00UTC today. If you need more time, write to sr-dev and let's see how we can find an easy solution. Once I finish with the restructuring, I will send another notification, from that moment the commits can go on as usual. 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

Send SIP Info within a dialog using $uac_req(method)="INFO"
by Jonathan Hunter 07 Dec '16

07 Dec '16

Hi Guys, Is it still the case that when using uac_req_send, you cant send withing a specific dialog?I can modify call-id, but I presume tags may be more of a problem?See old post below from 2015; > I am familiar with uac_req_send. but how do I send it with in a > specific dialog and with data in the INFO req ? sending a new request inside a dialog is not possible with uac_req_send(). It is not easy over all because you change the sequence order (CSeq value). Practically, you need to track how many requests you sent from the middle to update (and restore in reply) when caller or callee sends a new request. dialog module can track changes in CSeq for requests sent to callee, being used now for authentication of INVITE to another provider, when Kamailio adds the credentials. But for more you would need to extend the dialog module. I just need to send a SIP info within an established dialog to stop some function up stream, so wondered if this is still a blocker? Many thanks Jon

2 40

Error using Siremis 4.3.0
by Diogenes Aviles 07 Dec '16

07 Dec '16

I'm using Siremis 4.3.0 and PHP7. Installation was OK, but running siremis I have some problems in Administration-User Management. When I select an user there is the following error: [2016-12-06 23:18:47 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/siremis-4.3.0/openbiz/bin/Resource.php:283 *function:* errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/siremis-4.3.0/openbiz/bin/sysheader.inc 117 *function:* userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/siremis-4.3.0/openbiz/bin/Resource.php", 283, Array(11) ) @ *function:* trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/siremis-4.3.0/openbiz/bin/Resource.php 283 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/openbiz/bin/BizSystem.php 544 *function:* getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/siremis-4.3.0/siremis/modules/service/userEmailService.php 108 *function:* SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/siremis-4.3.0/siremis/modules/common/form/ErrorForm.php 34 *function:* Report ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 310 *function:* invoke ( ) @ /var/www/siremis-4.3.0/openbiz/bin/BizController.php 110 Regards -- Diogenes

2 1

Getting destination ip
by David Villasmil 06 Dec '16

06 Dec '16

Hello Guys, I'm wondering how do I get destination IP on a reply? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 7

Re: [SR-Users] Local messages are malformed in Kamailio 4.4.X
by g.Helio 06 Dec '16

06 Dec '16

2 1

Limiting Registration Contacts
by Colin Morelli 05 Dec '16

05 Dec '16

Hey all, I know Kamailio's registrar module has a max_contacts parameter that will limit the number of active contacts for an AOR. However, is there any mechanism to control what happens when that number is hit? When using a web client where every page refresh registers a new contact with Kamailio, it would be useful to automatically roll off the oldest registrations and replace to make room for the new ones. I know this can be manually accomplished with sqlops, but I was hoping something like this already existed. Best, Colin

3 5

INFO: modules purple and iptrtpproxy moved to obsolete folder
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, be aware that the modules purple and iptrtpproxy were moved to obsolete folder. They were not compiling for several years and nobody showed interest in updating them. If anyone wast to rescue them, she/he is welcome to get them back and upgrade the code to work with current versions of their dependencies. 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 0

INFO: kcore internal lib removed
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, mainly for those having private modules or other components for kamailio, take note that the internal library kcore has been removed. It was supposed to be a termporary place where to keep some kamailio specific functions at the time of merging with ser code back in 2008. The main code was related to internal statistics, which was re-implemented as fast counters several years ago, in libkcore being only some wrappers. You have to include now directly the counters.h from core. There were several files with some str operations helpers, not they are exported via strutils.h from core. The sst parser was used only by sst module and moved to it, same being done with some stats helper functions used only by snmpstats module. 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 0

kamailio bridge internal and external network
by zhang90501 05 Dec '16

05 Dec '16

My kamailio listen at internal ip and external ip(advertise parameter), it send message to freeswitch from internal ip and to client from external ip. How can I configure my kamailio to auto generate correct record route header?

3 4

Re: [SR-Users] Kamailio Processes Stop Responding
by Peter DiSanto 05 Dec '16

05 Dec '16

OK thank you Alex. I will look into your suggestions next time the problem arises. Each call does generate one SQL LCR lookup, one CDR INSERT, and also a timer route that runs 3 queries every few minutes. Once Kamailio stops responding it will continue that way indefinitely until a process restart. Will report back. Thanks. Pete D

3 2

Restructuring source code tree for v5.0
by Daniel-Constantin Mierla 05 Dec '16

05 Dec '16

Hello, I already started playing with various scripts to do the restructuring of the source code tree for v5.0 as we discussed in the past months and I am considering to do it quite soon, in the next few days. The result will look like: /Makefile (only-forwarding make commands to various subdirs) /README|INSTAL|LICENSE|... /src/Makefile* (build makefiles) /src/main.c (main c file to generate the binary) /src/core/*.{c,h} (c code for core, including subfolders such as parser, mem, atomic, ...) /src/lib/* (subfolders with the internal libraries) /src/modules/* (subfolders with the modules) Other subfolders from root directory may be relocated, but to be decided along the way. At the time of doing the restructuring, I will try to merge all the pull requests that look ok. Therefore it would be good that anyone interested in open pull requests to do the review (and merge) meanwhile. Also, I think it would be good to disable travis and nightly master builds during few days when the restructuring happens in order to avoid flooding sr-dev with alerts. If anyone has suggestions to add, do not hesitate to write on mailing lists. 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 0

Re: [SR-Users] msilo
by Slava Bendersky 05 Dec '16

05 Dec '16

Hello Daniel, I updated OFFLINE MESSAGE store based on username. https://paste.fedoraproject.org/499377/14809030/ Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 10:59:45 Subject: Re: [SR-Users] msilo Hello, the order you use to store and update the record is not safe, other messages from different users can be stored by another kamailio process. Use callid and username to select the last record inserted matching them, eventually protected with lock/unlock from cfgutils. Cheers, Daniel On 24/11/2016 14:34, Slava Bendersky wrote: Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: BQ_BEGIN Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

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Re: [SR-Users] RTP Pyload
by amar Smart Telecom 04 Dec '16

04 Dec '16

Dear all, We are seeing that some RTP Payload are with 14 bytes and some are with 15 bytes. 1. 14 byte is correct or 15 byte is correct? 2. Is TOC present in all payloads i.e. in all RTP messages? Following example with 14 byte payload. Regards, Amar

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Re: [SR-Users] msilo
by Slava Bendersky 04 Dec '16

04 Dec '16

Hello Everyone, I need help with while loop. Can't find why it try go over only first value from db search and not each value. https://paste.fedoraproject.org/497911/48081139/ Any help thank you. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 02:12:21 Subject: Re: [SR-Users] msilo Hello Daniel, I am trying more simple example of $dbr in REGISTER first. Insert into location temporary data. The problem is not looping though all entries in database only last one. Which compare same entry 100 times. if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query("cb", "SELECT * FROM location", "result"); xlog("rows: $dbr(result=>rows) cols: $dbr(result=>cols)\n"); $var(result) = 0; while($var(i)<$dbr(result=>rows)) { sql_pvquery("cb", "SELECT ruid FROM location WHERE callid = '$ci'", "$var(dbruid)"); if(fnmatch("$var(dbruid)", "$var(ruid)")) { xlog("L_INFO", "AOR: Found match --> [$var(dbruid)] = [$var(ruid)]\n"); } else { sql_query_async("cb","DELETE FROM location WHERE ruid = '$var(dbruid)'"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', NOW(), '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } } } From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:58:29 Subject: Re: [SR-Users] msilo Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Rtpproxy settings without NAT in Kamailio
by shantanu saha 03 Dec '16

03 Dec '16

Hello, I am trying to configure rtpproxy with Kamailio. kamailio.cfg is already configured (default) for rtpproxy. I just add the following line at the top of the cfg file. *#!define WITH_NAT* After that rtpproxy is working in local network. Both client and sip server are running in local network. But when I run this in public network, rtpproxy didn't trigger. Rtppoxy and Kamailio are running in same public IP, Sip client also running in another *public IP. *So there is *no NAT* *.* Both local and public version was Centos 6 Kamailio version : 4.4 rtpproxy version: 2.1 Note: udp connection between rtpproxy and Kamailio was established (both in public and private network) Thanks shantanu saha

2 1

Dual Stack Environments
by Colin Morelli 02 Dec '16

02 Dec '16

Hey all, Sorry for two questions out at the same time! I'm using kamailio in front of FS, and I'm about to enable IPv6, but I have a question that I'm not entirely clear on. What's the proper way to forward traffic using the same interface that it came in on? Specifically, if a client connects to kamailio on IPv6, it should connect to FS on IPv6, so that FS knows to advertise its v6 address in the SDP. The same should be true for v4. Is this possible? Best, Colin

2 1

Local messages are malformed in Kamailio 4.4.X
by Helio Okuyama 01 Dec '16

01 Dec '16

Local messages are malformed in Kamailio 4.4.X. For example after receiving an error response (603) to an INVITE message, the ACK is generated with header Max-Forwards concatenated to header CSeq and double CRLF, which is wrong. Via: SIP/2.0/TLS 173.16.10.61:5061 ;branch=z9hG4bKc782.5b960c7c48a4363a0126963daa797eb0.0;i=7 From: "22861"<sip:22861@173.16.10.61;transport=tls>;tag=snl_OIPEOzJUNy Call-ID: 61b500a71b510a1Pj61qq1k9dPW To: <sip:551138174108@174.16.4.108:5061;transport=tls>;tag=435012226 CSeq: 2351 ACKMax-Forwards: 70 User-Agent:... The fix for that seems to be just the order of append_str in function build_local() from tm\t_msgbuilder.c. append_str( p, method, method_len ); + append_str( p, CRLF, CRLF_LEN ); append_str( p, MAXFWD_HEADER, MAXFWD_HEADER_LEN ); - append_str( p, CRLF, CRLF_LEN );

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help with string modification
by Sergey Basov 01 Dec '16

01 Dec '16

Hello All. I have some troubles with upstream sip switch. It ignores SIP packets which contains: User-Agent: FPBX-2.11.0(11.17.1) or Server: User-Agent: FPBX-2.11.0(11.17.1) If space is present before first "(" then sip switch works as expected So my question is: how corektly make analyze and modify this headers using kamailio? In which routes i must parse it? i need such modifications: User-Agent: FPBX-2.11.0(11.17.1) -> User-Agent: FPBX-2.11.0 (11.17.1) Server: User-Agent: FPBX-2.11.0(11.17.1) -> Server: User-Agent: FPBX-2.11.0 (11.17.1) Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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Re: [SR-Users] TSILO question
by Federico Cabiddu 01 Dec '16

01 Dec '16

Hi, about your questions: Q1: yes it's feasible as long as the two Registrars (Kamailio) have the location information. Please see my reply on a thread on this topic: http://lists.sip-router.org/pipermail/sr-users/2016-February/091612.html Q2: given that TSILO depends on the users' location you have to put it in the Registrar. Best regards, Federico On Wed, Nov 30, 2016 at 5:32 AM, C. J Hsu (徐熙杰) <CJ.Hsu(a)quantatw.com> wrote: > hi Federico > > i'm stuck at TSILO and wondering if you could give me some advises. > > Q1: > > in lab1, I'm having two Kamailios with proxy and registrar together, UA1 > and UA2. > UAs go thru hardware load-balancing device (F5) to register themselves in > Kamailio, > > I wonder, UA1 calling to UA2, invite suspended in Kamailio 1, waiting for > UA2; > and when UA2 come to register in Kamailio 2, TSILO in Kamailio 1 resumes > the invite and sending it to Kamailio 2, is this a no problem scenario? > > Q2: > > if we decide to separate Proxy and Registrar in different Kamailio, when > should I put TSILO in? > > > RGDS > > C J >

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Real error?
by Olvera, Victor 01 Dec '16

01 Dec '16

Is this a real error? 9(1899) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

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In dialog messages cannot be routed after re-registration with outbound and gruu
by Robert 01 Dec '16

01 Dec '16

Hi, I'm using outbound with gruu enabled and with edge proxy, registrar and auth proxy all running as separate entities. My problem is once a dialog is formed between two UAs, if one side re-registers and gets assigned a new flow token, all subsequent in-dialog messages cannot be routed because the flow token in the Route set is no longer valid. What is the proper way to update the binding? I read in this thread <http://sip-router.1086192.n5.nabble.com/Outbound-and-Registrar-No-binding-u…> that "using GRUU means that Outbound is also used, so the Contact has a +sip.instance and reg-id params which are inspected by the registrar to create/update/delete a binding". How is the binding updated exactly? Any help would be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/In-dialog-messages-cannot-be-routed… Sent from the Users mailing list archive at Nabble.com.

1 0

Kamailio Processes Stop Responding
by Peter DiSanto 30 Nov '16

30 Nov '16

We have a Kamailio 4.2 machine with a custom config file we've been using for the better part of two years without issue. Suddenly we've run into an issue where Kamailio just stops responding to any INVITEs, and stops any logging. The processes keep running but stop doing anything. It is a full production machine which naturally causes problems. I have the memlog from the restart after a hang if that helps, but I presume it is too large for this list. We have increased shared memory to 1536 and Pkg memory to 32 but that didn't help, and in any case the shmem stats look OK to us. It is a full production machine doing about 200 CPS, the load is barely measurable but turning logging to debug will surely kill it. We use TOPOH and do get lots of log entries like : /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <▒> on state 100 /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2738]: parse_via(): ERROR: parse_via on: <▒#010d▒▒_▒#0117Jt▒#015#012X> /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2744]: parse_via(): ERROR: parse_via: via parse error /usr/sbin/kamailio[17865]: ERROR: topoh [th_msg.c:438]: th_unmask_via(): cannot find cookie in via2 /usr/sbin/kamailio[17865]: ERROR: topoh [th_mask.c:165]: th_mask_decode(): invalid input string"642525435_130081688(a)xxx.xxx.xxx.xxx" /usr/sbin/kamailio[17865]: ERROR: topoh [th_msg.c:484]: th_unmask_callid(): cannot decode callid and CRITICAL: sl [../../forward.h:279]: msg_send(): unknown proto 0 However, I believe these are not new and do not seem to affect overall call processing. Anyone have any idea what to look for or how to figure this out? shmem:fragments = 4019 shmem:free_size = 1538943936 shmem:max_used_size = 84349120 shmem:real_used_size = 71668800 shmem:total_size = 1610612736 shmem:used_size = 54855384

2 1

Offload SSL from backends with Kamailio
by Vladyslav Zakhozhai 30 Nov '16

30 Nov '16

Hi, I'm trying to understand the best (or reasonable) approach of offloading SSL encryption from backend to Kamailio. Let me simplify a little bit: UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH My main problem is in Contact header of SIP packet which passes through Kamailio SIP proxy and remains unmodified. For example, REGISTER request. There is FreeSWITCH backend which is registrar server as well. UAC send REGISTER request to it through Kamailio SIP proxy via SIP/TLS. This request dispatches to backend(s) by Kamailio with dispatcher module. Backend does not configured to support TLS. In this case everything works fine: I see REGISTER requests on FreeSWITCH. But Contact header of SIP message which is passing Kamailio remains unmodified. And as result I see on FreeSWITCH something like the following: Call-ID: Jpmjp4ruHI User: user_name@domain_name Contact: "" <sip:user_name@uac_ip :27026;transport=tls;fs_path=sip%3Akamailio_ip%3A5060> Agent: Linphone/3.10.2 (belle-sip/1.5.0) Status: Registered(TLS)(unknown) EXP(2016-11-28 11:48:28) EXPSECS(110) Ping-Status: Reachable Ping-Time: 0.00 Host: kamailio_host IP: kamailio_ip Port: 5060 Auth-User: unknown Auth-Realm: domain_name MWI-Account: user_name@domain_name As a result FreeSWITCH tries to originate call over SIP/TLS and it fails because FreeSWITCH does not configured to work with TLS. I want to understand what is correct workaraound of this issue. Do I need to modify Contact header manually on kamailio host and this is right approach? Or kamailio in case of correct config rewrites this header itself? If parts of my kamailio config would be useful I will post it later. Thanks in advance. -- С уважением, Владислав Захожай

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kazoo module
by Ján Füri 30 Nov '16

30 Nov '16

Hello, I am trying to install kazoo module. According to documentation I have installed libraries : librabbitmq, libjson, libuuid, but It always fails on missing json.h : CC (gcc) [M kazoo.so] kz_trans.o In file included from kz_trans.c:53:0: kz_json.h:12:18: fatal error: json.h: No such file or directory #include <json.h> ^ compilation terminated. ../../Makefile.rules:97: recipe for target 'kz_trans.o' failed make[1]: *** [kz_trans.o] Error 1 Makefile:511: recipe for target 'modules' failed make: *** [modules] Error 1 Can anybody help me ? BR Jan

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Re: [SR-Users] Kamailio IMS virtual machine
by Franz Edler 29 Nov '16

29 Nov '16

Hi Rafael, There should not be a problem with the VMware mage. Regarding your question: you have to check kamailio-pcscf.cfg to recognize when and it how forward the SIP message towards the I-CSCF. I guess it is the statement "t_relay();" in line 896. You may provide a core tracefile (generated with tcpdump), then I may have aquick look ate your ise. BR Franz Von: Rafi Aizenstain [mailto:rafi_a@cellowireless.com] Gesendet: Montag, 28. November 2016 16:12 An: franz.edler(a)technikum-wien.at Betreff: Kamailio IMS virtual machine Hello Mr. Franz I'm trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ But when running the machine as-is I keep getting errors from the P-CSCF module and it doesn't forward the SIP message towards the I-CSCF. Please see the following log: root@server:~# kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2096) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: server.net1.test [10.0.0.9]:4060 Aliases: *: pcscf.net1.test:* WARNING: no fork mode 0(2096) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(2096) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(2096) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 0(2096) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(2096) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(2096) WARNING: ims_usrloc_pcscf [usrloc_db.c:59]: connect_db(): DB connection already open... continuing 0(2096) INFO: <script>: [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] 0(2096) ERROR: <script>: REGISTER (sip:bob@net1.test (10.0.0.200:59721) to sip:bob@net1.test, 7013b8fc-11a0-2b70-3654-9ce68424e9a7) Looking into the configuration files I couldn't understand how does the P-CSCF determine that it has to forward the SIP message towards the I-CSCF module. Are you aware of any issue with the VMware machine you have shared. Please advise. Best Regards, Rafael A.

2 1

LCR modules troubles
by Laura 29 Nov '16

29 Nov '16

Dear list, i have a strange situation with module LCR on kamailio 4.3.6. A customer send a call to my first kamailio and i forward it with some tech prefix to my supplier. Here the 2 correct invites.. customer is sending to me with prefix 9001 14:34:29.155719 CUSTOMER_IP.101:5060 -> EU_KMAILIO_IP.40:5060 INVITE sip:900133139864069@EU_KMAILIO_IP.40 SIP/2.0. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP CUSTOMER_IP.101;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. To: <sip:900133139864069@EU_KMAILIO_IP.40>. Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 69. ..... and i forward to my supplier U 2016/11/28 14:34:29.157516 EU_KMAILIO_IP.40:5060 -> EU_SUPPLIER_IP.198:5060 INVITE sip:00233139864069@EU_SUPPLIER_IP.198 SIP/2.0. Record-Route: <sip:EU_KMAILIO_IP.40;lr;ftag=33F431FC-2103;did=ff5.6f01;vst=AAAAAAkAAgIAAgIKAQ4CBAYPeXIECQAABAcAAwQIAAUJOA--;nat=yes>. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP EU_KMAILIO_IP.40:5060;branch=z9hG4bK2d4.782830bfdde5e873d3dc6a71bcbba70f.0. Via: SIP/2.0/UDP CUSTOMER_IP.101;rport=5060;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. To: <sip:00233139864069@EU_SUPPLIER_IP.198>. Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 68. Timestamp: 1480343669. Contact: <sip:441753825186@CUSTOMER_IP.108:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. On both invites the To field respect the correct information from INVITE so that the kamailio CDR is working correct. After that the supplier reply me 503 and call is come back to my kamailio. LCR modules send the call to second supplier that on this case is my second box kamailio on another countrie.. Here there is this problem that i signed on RED U 2016/11/28 14:34:43.191618 EU_KMAILIO_IP.40:5060 -> EU_SUPPLIER2_IP.242:5060 *INVITE sip:905233139864069@EU_SUPPLIER2_IP.242 SIP/2.0.* Record-Route: <sip:EU_KMAILIO_IP.40;lr;ftag=33F431FC-2103;did=ff5.6f01>. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP EU_KMAILIO_IP.40:5060;branch=z9hG4bK2d4.782830bfdde5e873d3dc6a71bcbba70f.1. Via: SIP/2.0/UDP CUSTOMER_IP.101;rport=5060;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. *To: <sip:900133139864069@EU_KMAILIO_IP.40>.* Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 68. Timestamp: 1480343669. Contact: <sip:441753825186@CUSTOMER_IP.108:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. .... The LCR modules make the new INVITE using the original To field and don't rewrite it like on the first request.. How is that possibile ? Any idea ? Best regards Laura

3 2

XAVP Null
by Linux Vince 29 Nov '16

29 Nov '16

Hi, I am sure i am doing something very basic wrongly. Here is my config file snippet $xavp(caller_peer[0]=>accountname) = "AAA"; xlog("L_WARN", "XAVP ACCOUNT NAME $xavp(caller_peer[0]=>accountname) \n"); No matter what i set this XAVP at, it always prints null in log file as below WARNING: <script>: XAVP ACCOUNT NAME <null> Please help how i can set XAVP and use it further in config file.

3 2

MSRP auth does not work with blink
by Pranathi Venkatayogi 29 Nov '16

29 Nov '16

Hi, I am trying to use MSRP feature of blink client. In the server config when auth is challenged, I see the following error in the client. Can you please advise if I am configuring the server right? Note: the return value of pv_www_authenticate is "-5". 13 2016-11-07 17:57:26.494124 [blink.exe 24976]: Closed connection to 10.0.16.51:2855 ('NotificationProxyLogger' object has no attribute 'received_illegal_data') Find attached full set of client logs. Here is the relevant portion of kamailio.cfg: if (!pv_www_authenticate("MY_DOMAIN", "$var(passwd)", "0", "$msrp(method)")) { www_challenge("MY_DOMAIN", "1"); xlog("L_ERR", "MSRP auth failed with: $var(retValue)\n"); exit; } 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.

2 4

Problems with Siremis and Kamailio - PHP7
by Diogenes Aviles 29 Nov '16

29 Nov '16

I have Kamailio 4.4 installed in ubuntu 16.04.03, with Siremis 4.2.0, PHP7 and apache2. All is OK, but when I try to install the Siremis Wizard, the step 2 (Data Base Configuration) doesn't work, and there is the message "Something Went Wrong". I tried all the option on Step 2, but the problem continuous. Then I can't go to step 3 Application Configuration. The Siremis DataBase is created and the user also. I appreciate any help. -- Diogenes

2 1

Siremis v4.3.0 Released
by Daniel-Constantin Mierla 29 Nov '16

29 Nov '16

Hello, a new release of Siremis, the web management interface for Kamailio, has been released recently. The main work was towards making it compatible with PHP 7 and the new constraints from MySQL 5.7+. The announcement is available at: - http://siremis.asipto.com/2016/11/26/siremis-v4-3-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World 2017 - May 8-10, in Berlin - www.kamailioworld.com

1 0

custom version database/table
by David Villasmil 28 Nov '16

28 Nov '16

Hello all, I have a databse where i created a "version" table only with a version, a dispatcher records. I only want to use dispatcher on a db. I don't know if kamailio is looking for the version tables somewhere else, or it does find it in my database (not named kamailio) but the format is wrong. I used: CREATE TABLE `version` ( `table_name` VARCHAR(32) NOT NULL, `table_version` INT UNSIGNED DEFAULT 0 NOT NULL, CONSTRAINT table_name_idx UNIQUE (`table_name`) ); INSERT INTO version (table_name, table_version) values ('version','1'); INSERT INTO version (table_name, table_version) values ('dispatcher','4'); To create an populate the table, but kamailio returns: mod_init(): could not initiate a connect to the database but it does connect properly... any thoughts? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 3

Recent problem with fix_nated_sdp and kamailio 4.4.4
by Anthony Messina 28 Nov '16

28 Nov '16

Prior to the upgrade to Kamailio 4.4.4, I was using 4.4.3 with updates through 81df84b from the 4.4 branch and the following fix_nated_sdp call worked through the initial INVITE and when a voicemail branch was appended from the failure_route. route[NATMANAGE] { ... # RTPEngine is not needed but if STUN addresses are being used on our # internal network, rewrite the addresses to the internal source address if(!isbflagset(FLB_RTPENGINE) && has_body("application/sdp")) { if(dst_ip==10.1.1.2 && compare_pure_ips($sel(contact.uri.host), "<EXTERNAL_IP>")) { fix_nated_sdp("10"); } } ... } Since the upgrade to 4.4.4, when the original INVITE fails over to voicemail (branch appended), fix_nated_sdp creates the SDP below, doubling the replaced IP addresses. Asterisk replies with 488, as the media address is completely invalid. I need something like the above (which worked for the request and the reply) but can't figure out what the problem is. v=0 o=Zoiper 0 0 IN IP4 10.1.1.18210.1.1.182 s=Zoiper c=IN IP4 10.1.1.18210.1.1.182 t=0 0 m=audio 63732 RTP/SAVP 9 3 0 97 101 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 4

kamailio core dumps once every few days
by davy van de moere 28 Nov '16

28 Nov '16

Gents, On a not really loaded kamailio (less than 50 users at the moment), I'm getting a core dump every few days. I've tried multiple verisons of Kamailio, 4.4.x , 4.3.x and also on 5.0.0-dev7 . I can't reproduce it adhoc, but every few days it happens... Is that something someone wants to jump in? grtz, davy Version: version: kamailio 5.0.0-dev7 (x86_64/linux) 7360b0 [New LWP 29728] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -f /usr/local/etc/kamailio//k'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f92c5a0be98 in __dialog_sendpublish (dlg=0x7f92c0e3a6c0, type=64, _params=0x7f92c0423500 <params>) at pua_dialoginfo.c:264 264 uri = dlginfo->to_uri; (gdb) backtrace #0 0x00007f92c5a0be98 in __dialog_sendpublish (dlg=0x7f92c0e3a6c0, type=64, _params=0x7f92c0423500 <params>) at pua_dialoginfo.c:264 #1 0x00007f92c01e2563 in run_dlg_callbacks (type=64, dlg=0x7f92c0e3a6c0, req=0x7f92cc6d5db8, rpl=0x0, dir=1, dlg_data=0x0) at dlg_cb.c:256 #2 0x00007f92c01a92b1 in dlg_terminated (req=0x7f92cc6d5db8, dlg=0x7f92c0e3a6c0, dir=1) at dlg_handlers.c:366 #3 0x00007f92c01b610f in dlg_onroute (req=0x7f92cc6d5db8, route_params=0x7ffc7de27ba0, param=0x0) at dlg_handlers.c:1409 #4 0x00007f92c9292171 in run_rr_callbacks (req=0x7f92cc6d5db8, rr_param=0x7ffc7de27c20) at rr_cb.c:96 #5 0x00007f92c9287ac2 in after_loose (_m=0x7f92cc6d5db8, preloaded=0) at loose.c:928 #6 0x00007f92c9288168 in loose_route (_m=0x7f92cc6d5db8) at loose.c:962 #7 0x00007f92c928b7ee in w_loose_route (msg=0x7f92cc6d5db8, p1=0x0, p2=0x0) at rr_mod.c:266 #8 0x000000000041f5d9 in do_action (h=0x7ffc7de285b0, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1054 #9 0x000000000042bcde in run_actions (h=0x7ffc7de285b0, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1552 #10 0x000000000042c383 in run_actions_safe (h=0x7ffc7de29c60, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1620 #11 0x000000000061fe86 in rval_get_int (h=0x7ffc7de29c60, msg=0x7f92cc6d5db8, i=0x7ffc7de28cc0, rv=0x7f92cc5f7bc8, cache=0x0) at rvalue.c:912 #12 0x00000000006243c3 in rval_expr_eval_int (h=0x7ffc7de29c60, msg=0x7f92cc6d5db8, res=0x7ffc7de28cc0, rve=0x7f92cc5f7bc0) at rvalue.c:1910 #13 0x000000000041f0d1 in do_action (h=0x7ffc7de29c60, a=0x7f92cc600430, msg=0x7f92cc6d5db8) at action.c:1030 #14 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc600430, msg=0x7f92cc6d5db8) at action.c:1552 #15 0x000000000041f548 in do_action (h=0x7ffc7de29c60, a=0x7f92cc6006d0, msg=0x7f92cc6d5db8) at action.c:1045 #16 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc6006d0, msg=0x7f92cc6d5db8) at action.c:1552 #17 0x000000000041c0fb in do_action (h=0x7ffc7de29c60, a=0x7f92cc545730, msg=0x7f92cc6d5db8) at action.c:678 #18 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc537760, msg=0x7f92cc6d5db8) at action.c:1552 #19 0x000000000042c44b in run_top_route (a=0x7f92cc537760, msg=0x7f92cc6d5db8, c=0x0) at action.c:1641 #20 0x00000000005726a2 in receive_msg ( buf=0xa1c200 <buf> "BYE sip:echo@123.48.11.215:5070;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 0.0.0.0:5060;rport;branch=z9hG4bKPj26b505ba-7ede-4198-a917-bde966b6d8b6\r\nMax-Forwards: 69\r\nFrom: sip:192.168.60.1;tag=9a10d8ab-"..., len=462, rcv_info=0x7ffc7de29fb0) at receive.c:264 #21 0x000000000047bcdc in udp_rcv_loop () at udp_server.c:507 #22 0x0000000000509de1 in main_loop () at main.c:1618 #23 0x00000000005112de in main (argc=5, argv=0x7ffc7de2a478) at main.c:2639 (gdb)

2 2

Balancer with Dispacther Module
by Rodrigo Moreira 28 Nov '16

28 Nov '16

I need help. I'm building a load balancer using the dispatcher. It does NAT with too many SIP Servers. But when the balancer receives an ACK from UAC, it immediately returns the message "404 Not Here" to UAC. Help me solve this, since this message (ACK) has to be forwarded to SIP Servers and finally to UAS to start an RTP session. I thank you. Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Daniel, I am trying more simple example of $dbr in REGISTER first. Insert into location temporary data. The problem is not looping though all entries in database only last one. Which compare same entry 100 times. if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query("cb", "SELECT * FROM location", "result"); xlog("rows: $dbr(result=>rows) cols: $dbr(result=>cols)\n"); $var(result) = 0; while($var(i)<$dbr(result=>rows)) { sql_pvquery("cb", "SELECT ruid FROM location WHERE callid = '$ci'", "$var(dbruid)"); if(fnmatch("$var(dbruid)", "$var(ruid)")) { xlog("L_INFO", "AOR: Found match --> [$var(dbruid)] = [$var(ruid)]\n"); } else { sql_query_async("cb","DELETE FROM location WHERE ruid = '$var(dbruid)'"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', NOW(), '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } } } From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:58:29 Subject: Re: [SR-Users] msilo Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

tm:local-request does not intercept locally generated CANCEL
by Andy 25 Nov '16

25 Nov '16

should tm:local-request intercept CANCELs, if not then where can the locally generated CANCELs from t_cancel.c be intercepted -- View this message in context: http://sip-router.1086192.n5.nabble.com/tm-local-request-does-not-intercept… Sent from the Users mailing list archive at Nabble.com.

2 2

protocol/port mismatch
by Yuriy Gorlichenko 25 Nov '16

25 Nov '16

Hi. Have strange issue Calling from kamailio through WS invite and resposes goes ok but with ACK to WS client has this issue get_send_socket2(): protocol/port mismatch (forced tls:kamailioIP:4443, to udp:MyWSClientIP:65451) Guess kamailio thinks that it is UDP client because $ru is sip:10002@MyWSClientIP:65451 But can not understand how to fix this...

2 2

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 24 Nov '16

24 Nov '16

Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Wednesday, 23 November, 2016 21:10:53 Subject: Re: [SR-Users] msilo Hello Everyone, Need another set of eyes, can't find how to To: got << in uri. Nov 23 19:50:55 cavprx00 kamailio: 3(9625) DEBUG: <core> [parser/parse_uri.c:1280]: parse_uri(): parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:4200@client_public_ip:49418>;messagetype=SMS> (47) Nov 23 19:50:55 cavprx00 kamailio: 3(9625) ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: [<sip:4200@ client_public_ip :49418>;messagetype=SMS] Nov 23 19:50:55 cavprx00 kamailio: 3(9625) ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches route[MESSAGE_FORWARD] { xlog("L_INFO", "Incoming new SMS [$rm] from $tU --> $td\n"); if(fnmatch("$rs", "202") && fnmatch("$rr", "Accepted") && allow_trusted("$si", "$proto")) { $sht(b=>from) = $fu; $sht(b=>oldto) = $tu; $avp(new_to) = "<" + $sht(b=>oldto) + ">" + ";messagetype=SMS"; xlog("L_INFO", "MESSAGE: B2BUA send --> [$rs] with [$rr] from [$sht(b=>from)] to [$sht(b=>oldto)] -->[$td]\n"); } if(compare_ips("$td", "10.18.130.27")) { xlog("L_INFO", "Message from B2BUA contain domain --> $td user --> $(sht(b=>oldto){uri.user}) updating...\n"); sql_pvquery("cb", "select contact from location where username = '$(sht(b=>oldto){uri.user})'","$avp(dst)"); #$avp(new_dst) = $(hdr(Route){param.value,received}); xlog("L_INFO", "New destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } if(fnmatch("$fU", "offline")) { xlog("L_INFO", "OFFLINE SMS: from [$fU] user --> $fU\n"); $avp(from) = $(sht(b=>oldto){uri.user}); xlog("L_INFO", "OFFLINE SMS: Searching destination user $avp(oexten)\n"); sql_pvquery("cb", "select contact from location where username = '$avp(from)'","$avp(dst)"); xlog("L_INFO", "OFFLINE SMS: set new destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } xlog("L_INFO", "OFFLINE SMS: new AVP DST $avp(new_to)\n"); $sht(a=>to) = $avp(new_to); remove_hf("To"); insert_hf("To: $sht(a=>to)\r\n", "To"); $du = $sht(a=>to); xlog("SMS from --> $fU domain $fd\n"); xlog("-------------------------------------\n"); xlog("L_INFO", "FROM --> $avp(from)\n"); xlog("L_INFO", "OLD_TO --> $sht(b=>oldto)\n"); xlog("L_INFO", "NEW_TO --> $avp(new_to)\n"); } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 17 November, 2016 05:44:22 Subject: Re: [SR-Users] msilo Hello, if you want to fetch them for config usage is ok, otherwise m_dump() is still better, because it takes care of removing delivered messages, etc... Cheers, Daniel On 16/11/16 16:08, Slava Bendersky wrote: Hello Daniel, Yes that good alternative. I will use sql to fetch or insert staff. Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 16 November, 2016 04:37:24 Subject: Re: [SR-Users] msilo Hello, as an alternative, I think it should work by using sql_query(...) to insert whatever values one would like to have in msilo table. This might be a quick workaround until someone will code the extension to get the body via a function parameter. Cheers, Daniel On 15/11/16 16:07, Federico Cabiddu wrote: BQ_BEGIN Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: BQ_BEGIN Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 2

Howto send rtp stats infos to homer
by Oliver Roth 23 Nov '16

23 Nov '16

Hi there We are using homer to analyce and monitor traffic. I did not get a solution to send x-rtp or p-rtp stats to homer. How can I find infos about jitter, sent/received packages, lost packages, etc.? Our current infos are based on Rtpstat provided by kamailio. At the moment I do $var(xrtpstat) = $(rtpstat{s.striptail,1}); # Work the stats $var(octetsent) = $(var(xrtpstat){s.select,1, }); #octetsent / OS $var(packetsent) = $(var(xrtpstat){s.select,3, }); #PS $var(octetrcv) = $(var(xrtpstat){s.select,8, }); #OR $var(packetrcv) = $(var(xrtpstat){s.select,10, }); #PR $var(errorsent) = $(var(xrtpstat){s.select,5, }); $var(errorrcv) = $(var(xrtpstat){s.select,12, }); if ($var(octetsent) != "" || $var(packetsent) != "") { xlog(,"L_INFO", "WITHINDLG PS=$var(rtp0),PR=$var(rtp1),PL=$var(rtp3) \n"); xlog(,"L_INFO", "WITHINDLG xrtpstats=$var(xrtpstat) \r\n"); xlog(,"L_INFO", "WITHINDLG rtpstats=$rtpstat \r\n"); xlog(,"L_INFO", "WITHINDLG X-RTP-Stat: PS=$var(packetsent),OS=$var(octetsent),PR=$var(packetrcv),OR=$var(octetrcv),PL=$var(errorsent) \r\n"); #append_hf("P-RTP-Stat: $rtpstat\r\n"); append_hf("X-RTP-Stat: PS=$var(packetsent);OS=$var(octetsent);PR=$var(packetrcv);OR=$var(octetrcv);PL=$var(errorsent);JI=600;LA=40;\r\n"); } Other question: Where has this to be placed? Only on bye messages? KR, Oli

1 0

Change from in a failure route
by Nelson Migliaro 23 Nov '16

23 Nov '16

Hello, My sip vendor sends me a 403 error when the caller id (CLI) sent is not valid. I would like to set up a failure route to capture that 403 and set up a default CLI. I tried to use a uac_replace_from() in a failure route but I get: "Command cannot be used in the block" Does anyone know how can I change the CLI in a failure route how how can achieve the default CLI? Thank you

2 1

Force OpenSER to send a Stop radius accounting record in failure route.
by Jan Hazenberg 23 Nov '16

23 Nov '16

Hi all, I have a issue here on a old OpenSER 1.3 system. We have some clients that respond to a BYE message with a 481 resulting in a missing radius accounting record. I have added some code to the failure route to force accounting even if a 481 response in received: if (t_check_status("481") && $rm == "BYE") { xlog("L_INFO", "INFO: [FAILURE] 481 response detected!! Forcing accounting! - SRC=$si:$sp R=$ru ID=$ci FU=$fu M=$rm S=$rs\n"); acc_rad_request("BYE with 481 response"); } Now OpenSER sends a accounting record, but this has the attribute Acct-Status-Type set to Failed. This gives a problem with our accounting service and i would like to change it to a normal Stop record. Is there a way to change this and force OpenSER/Kamailio to send a Stop record instead of a Failed record? Thanks, Jan

2 3

SDP Codec not removed with RTPengine - but with rtpproxy it worked
by Oliver Roth 23 Nov '16

23 Nov '16

Hi there I have the following problem - I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage(). Originating INVITE with the "clearmode" m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 I do the following in the config route[SDP] { xlog(, "L_INFO", "SDP: request method $rm"); # remove CLEARMODE if Colt if($avp(s:todirection) =~"^MyCarrier+") { if (is_method("INVITE")) { xlog(, "L_INFO", "SDP remove: request method $rm"); msg_apply_changes(); sdp_remove_codecs_by_name("CLEARMODE"); #sdp_remove_codecs_by_id("125"); } } } This block gets hit as I can see with the xlog entry. In the sent INVITE the "Clearmode" is still in the sdp header a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 So with rtpproxy - it worked With rtpengine not any more ... Any idea? KR, Oli

3 10

IMS related : ims_usrloc_pcscf : "lookup_check_received" setting is useless now
by Dragos Oancea 23 Nov '16

23 Nov '16

Hi The setting "lookup_check_received" is documented here : http://www.kamailio.org/docs/modules/4.4.x/modules/ims_usrloc_pcscf.html#id… But looking through the code I saw it's not used anymore, although it can be passed as a parameter. I was wondering if this is an accident or something intentional. I found some more details here: http://lists.sip-router.org/pipermail/sr-dev/2014-June/024371.html Cheers, Dragos

2 1

async_workers vs. async workers modparam in case of async_route()
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, If I am using async_route() with delay, what is the relationship between the core async_workers parameter value and the async module 'workers' modparam? This doc change was intended to clarify it, I think: https://www.mail-archive.com/sr-dev@lists.sip-router.org/msg25002.html But it does not for me: > Number of worker processes to be started to handle the > asynchronous tasks for async_route() and async_sleep(). So, does that mean the module starts its own processes for these cases? If so, does that mean I don't need core async_workers at all? Or do I? If I do, what happens if I have fewer core async_workers than modparam workers? What if I have more? Should I aim to keep them in sync? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

log_prefix ignored in resumed routes
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have this log_prefix: log_prefix="[$ci $mt] " Which means xlog("L_INFO", "xyz") normally results in log entries like this: Nov 22 00:29:50 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27625]: [1-27702(a)10.150.20.6 1] INFO: xyz But if xlog("L_INFO", "xyz") happens to occur in a route[] block that is resumed after async_route()—I have not tested other async functions—there is no prefix: Nov 22 00:29:52 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27678]: INFO: xyz -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

flags documentation
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have this set up in my configuration boilerplate: -- flags DUMMY : 1, PDD_DAMPEN : 2, PROXY_MEDIA_SET : 3, #!ifdef WITH_NAT_TRAVERSAL SIPPING : 4, #!endif DUMMY2 : 12 -- I then use these aliases in transaction flags, e.g. setflag(PDD_DAMPEN). But I can honestly say I don't remember how I arrived at this approach. I wrote it somewhere in early 3.x days. However, when I look at the core documentation for the 'flags' declaration, all I find is: http://www.kamailio.org/wiki/cookbooks/4.4.x/core#flags Or namely: -- flags Alias name: bool -- And there is no entry for "bool". So, I couldn't have possibly arrived at this usage from the documentation. Can anyone help me reverse-engineer this historical phenomenon? And, given that the documentation doesn't support it particularly, is it still an encouraged practice? Or should one use #!defined constants instead, as in the stock config? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Message flags preservation for replies in onsend_route
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have -- flags BLAH : 1, DUMMY : 12 onsend_route_reply=yes route { ... setflag(BLAH); if(!t_relay()) sl_reply_error(); } onsend_route { if(is_reply() && isflagset(BLAH)) do_things(); } -- My discovery has been that isflagset(BLAH) always == FALSE _unless_ an onreply_route has been armed for the request, regardless of whether it is necessary. In other words, the following causes isflagset(BLAH) to == TRUE (if set, when appropriate) in the onsend_route for the reply: -- flags BLAH : 1, DUMMY : 12 onsend_route_reply=yes onreply_route[BOGUS] { exit; } route { ... setflag(BLAH); t_on_reply("BOGUS"); if(!t_relay()) sl_reply_error(); } onsend_route { if(is_reply() && isflagset(BLAH)) do_things(); } -- But without the onreply_route, it doesn't work. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

Registering Trunk help please
by Andrei Mahalean 22 Nov '16

22 Nov '16

Hello, I am having a bit of trouble finding information on how to setup a registering trunk with Kamailio. By registering trunk, I mean a trunk between a PBX and Kamailio, of say 15 DID's, where the main DID (Pilot) registers with Kamailio and all the other 14 DID are able to make/receive calls without having to register all DID's individually. If the pilot DID registration goes away, the trunk is then considered to be unreachable/down. I have seen this in the past with a Broadsoft setup, but I can't figure out a way to do it with Kamailio. I have my pilot DID setup and registered with Kamailio ok and I can make calls in and out through my PSTN gateway, but I'm unsure how to group all the other DID's so they are matched against the pilot DID. Is this possible? Any help is appreciated. Regards, Andrei. -- Regards, Andrei.

2 1

Roadmap to Kamailio v5.0
by Daniel-Constantin Mierla 21 Nov '16

21 Nov '16

Hello, following the last IRC devel meeting, I am writing the summary of the main points being discussed there and what's planned to get to Kamailio v5.0. 1) Freezing should be done either before Chrismas holidays or at beginning of January 2017 to aim meeting the deadline for next Debian stable distro 2) Restructuring the source tree To be done in the weeks before freezing, so eventual fixes won't have hard time to be backported to 4.4. This means it will be done in December. Open issues: a) anyone aware of a tool that can update include directives upon relocation of the file? b) who is interested to join an eventual online session to perform the restructuring? C coding is not a must here, but more knowledge of sed/awk/perl/python/... and Makefiles ... 3) Defining a template for generating main file of a module The scope is to have a descriptive file for the module where one lists exported symbols (config params, functions, kemi functions, rpc/mi commands, etc.) which along with a template will generate the c code. The effective implementation of the functions/commands will be done separately, in a different file (e.g, the main file that is generated can have references to parameter variables with extern ... and will also generate the .h which will have only the prototypes for the functions). The main goal here is to get a way of easily generated an index of exported symbols that can be used to discover missing documentation. Right now we sort of do 'gentlemen agreement' on documenting functions and parameters, but many rpc commands are not documented (they are extracted with some tool scanning compiled objects). Also, same concept as with c code can be used to generate some stubs for documentation that can be included in docbook xml files. Few proposals of templating engines were mentioned, coming in my mind now being jinja2 and mustache. Anyone else with more suggestions? Or even propose alternatives of what can be used to solve this relation between code and docs? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

1 0

usage of app_mono : how to communicate results back to config file
by Pranathi Venkatayogi 21 Nov '16

21 Nov '16

Hi, I am using app_mono module and building my logic in c# app. I am unable to understand how to send the decision back to kamalio.cfg. More precisely: I would like my routing logic to allow specific user names though they are not registered as subscribers. Like: translate(a)myip.cyracom.com<mailto:translate@myip.cyracom.com> I have the logic in c# app which takes $ru and validates such that: Host is myip.cyracom.com User is translate How do I communicate the decision if it is valid or not back to my config script. Do I need to introduce new pseudo variable for this purpose? Thanks, Pranathi

2 1

Is there a way to detect inactive MSRP chat sessions without using SIP session timers
by Pranathi Venkatayogi 18 Nov '16

18 Nov '16

Hi, I am using MSRP feature of Blink client. I want to track when the chat starts and ends so I can free up server resources. Apparently Blink client does not send BYE when MSRP chat window closes. It does not support SIP session timers even. So potential use of SIP session timers (sst and dialog modules) is ruled out. Only possible way is to track every message that goes through the server and keep the session alive. If no message comes within “expiry time period”, server can send “bye” to all parties and tear up the session. Can you please comment if there are other approaches to solve this problem? If not, are they any modules that will help me implement the above approach easily? Thanks Pranathi

2 2

tcp_set_connection_lifetime - does not close the connection immediately
by Andy 18 Nov '16

18 Nov '16

I am calling tcp_set_connection_lifetime(conid, 1), which I understand is one sec after now, but I see it taking upto 60 secs to kill the connection, is there any other timer which has a larger interval which might affect this. -- View this message in context: http://sip-router.1086192.n5.nabble.com/tcp-set-connection-lifetime-does-no… Sent from the Users mailing list archive at Nabble.com.

2 2

Htable sht_rm_name_re usage
by Grant Bagdasarian 18 Nov '16

18 Nov '16

Hello Community, We were facing a weird issue in one of our Kamailio servers where custom htable entries were removed which did not belong to a particular call. Background Information: We use htable to store the call-id and a uuid when an INVITE message is received and remove these entries when the call is completed/failed (so on CANCEL, BYE, 4XX, 5XX, 6XX, etc). In this case the function sht_rm_name_re was causing incremented call-id's belonging to different calls to be removed from the htable, since the removal is based on regex and not the exact key. On INVITE we do this: $var(reference) = $uuid(g); $sht(calls=>$ci) = $ci; $sht(calls=>$ci::reference) = $var(reference); On BYE/4XX,5XX,6XX, etc we did this: sht_rm_name_re("calls=>$ci"); sht_rm_name_re("calls=>$ci::reference"); SIPP generates incremented call-id's, like this: 1-14875(a)1.0.0.1<mailto:1-14875@1.0.0.1>. When you generate enough calls, for instance 30 calls, the 21st call will have the call-id 21-14875(a)1.0.0.1<mailto:21-14875@1.0.0.1>. If for instance the BYE for the call-id 1-14875(a)1.0.0.1<mailto:1-14875@1.0.0.1> arrives before 21-14875(a)1.0.0.1<mailto:21-14875@1.0.0.1> , both htable entries will be removed by the sht_rm_name_re function, since it matches based on regex. To fix this we made sure the function matches on the exact keys by doing this: $var(callid) = '^' + $ci + '$'; $var(callid_reference) = '^' + $ci + '::reference' + '$'; sht_rm_name_re("calls=>$var(callid)"); sht_rm_name_re("calls=>$var(callid_reference)"); Hope this may help someone in the future facing the same issue. Regards, Grant Bagdasarian CM

2 1

Kamailio for SIP Capture With Registration
by Ali Dakroub~Vanrise Technical Support 18 Nov '16

18 Nov '16

Hello, Kindly note that I am configuring Kamailio for SIP capture with "Registration" module (registrar.so). Before loading the "Sip Capture" module (sipcapture.so) into Kamailio.cfg file, the registration of the SIP Phone (X-lite) is successful. Yet, when we add the "Sip Capture" module (sipcapture.so) into Kamailio.cfg file with the below parameters: modparam("sipcapture", "db_url", "postgres://postgres:postgres@localhost/kamailio") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_moni_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.10.1:5060") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth0") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* Linux only */ /* Promiscious mode RAW socket. Mirroring port. */ modparam("sipcapture", "promiscious_on", 1) /* activate BPF */ modparam("sipcapture", "raw_moni_bpf_on", 1) With the following routing logic: ####### Routing Logic ######## request_route { #sip_capture(); # CANCEL processing # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service } route[REGISTRAR] { if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } } The registration of the SIP Phone fails and we are getting the following error in Kamailio.log file: 2016-11-16T09:49:28.809247-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f60c7593910,481,0,192.168.26.4:14661,16): Bad file descriptor(9) 2016-11-16T09:49:28.856080-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: sl [../../forward.h:192]: msg_send(): udp_send failed 2016-11-16T09:49:28.860251-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: registrar [reply.c:734]: reg_send_reply(): failed to send 200 OK It would be highly appreciated to check what went wrong here. Thanks in advance. Ali Dakroub

2 1

Kamctl stats
by Alexandru Covalschi 18 Nov '16

18 Nov '16

Hello list, Where can I found any information to completely understand what do values returned by 'kamctl stats' represent? Cheers, Alex

2 1

Re: [SR-Users] msilo
by Slava Bendersky 17 Nov '16

17 Nov '16

Hello Daniel, Yes that good alternative. I will use sql to fetch or insert staff. Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 16 November, 2016 04:37:24 Subject: Re: [SR-Users] msilo Hello, as an alternative, I think it should work by using sql_query(...) to insert whatever values one would like to have in msilo table. This might be a quick workaround until someone will code the extension to get the body via a function parameter. Cheers, Daniel On 15/11/16 16:07, Federico Cabiddu wrote: Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: BQ_BEGIN Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Possible memory leak in mysql driver
by Alexandru Covalschi 17 Nov '16

17 Nov '16

Hello list, We’re using dev version of Kamailio: version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: ff63e5 compiled on 15:46:49 May 31 2016 with gcc 4.9.2 Sometimes we encounter such issue: Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_res.c:188]: db_mysql_convert_rows(): error while converting row #16 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_res.c:217]: db_mysql_convert_result(): error while converting rows Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_dbase.c:261]: db_mysql_store_result(): error while converting result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_query.c:139]: db_do_query_internal(): error while storing result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: permissions [trusted.c:91]: reload_trusted_table(): failed to query database Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_dbase.c:444]: db_mysql_fetch_result(): error while converting row #15 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: htable [ht_db.c:234]: ht_db_load_table(): Error while fetching result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f5ebda8ae18), called from db_mysql: km_dbase.c: db_mysql_free_result(305), first free db_mysql: km_dbase.c: db_mysql_free_result(305) - aborting Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12281]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 16 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT: <core> [main.c:739]: handle_sigs(): child process 12276 exited by a signal 6 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT: <core> [main.c:742]: handle_sigs(): core was not generated Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD The thing is we heavily use mysql module, but only to update the in-memory tables by kamcmd. Each N minutes a special script updates the trusted,address and htable executing kamcmd. Kamailio (and kamcmd as well) talks only with localhost mysql server. What I saw when encountered that issue on a live machine is that issue happens only with one of child processes, any other are ok. Interesting thing is that happens at the same time with machines on the same «set», I mean that issue happened simultaneously with two our test machines which actually didn’t have any load on them. The common thing between those machines is that they are in same subnet and local mysql databases are filled by scripts which query same external db. I can’t confirm if there were or there weren’t any networking issues at that time with those machines, but as soon as kamcmd queries localhost that shouldn’t be the source of the issue. So my questions are: 1. Has anyone encountered such thing? 2. Maybe the issue is already localized so it has sense to update? We actually use that on production (pls don’t throw too much rocks at me), so maintenance should be properly planned and I must be sure update won’t break anything. 3. If update is proposed - how to do it? I mean - follow the guide https://www.kamailio.org/wiki/install/devel/git or there are some other tips? I suppose in ideal world I don’t even stop the binary, only restart after make all && make install are done, as everything is in-memory. Am I correct? 4. When can we expect stable 5.0 version? (at least tell if it’s months/years) Thanks in advance!

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Error message not seen before
by Kevin Greene 17 Nov '16

17 Nov '16

Hi fellow Kamailio people: I was using the Siremis (v2.0) to manage my Kamailio application. I received a message labeled "System Internal Error" It seems, assuming I understand it correctly, that a table called "kamailio.uacreg" does not exist in the database. I receive this error when I click on the following: Main System > SIP Admin Menu > Server Services > UACReg List function. Any help in pointing me in the right direction would be great. Thanks. Kevin 2016-11-16 22:29:23 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.l_uuid, T0.l_username, T0.l_domain, T0.r_username, T0.r_domain, T0.realm, T0.auth_username, T0.auth_password, T0.auth_proxy, T0.expires FROM uacreg T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'kamailio.uacreg' doesn't exist Script name and line number of error: /var/www/siremis/openbiz/bin/data/BizDataObj_Lite.php:447 *function:* _run_search ( Array(2) ) @ /var/www/siremis/openbiz/bin/data/BizDataObj_Lite.php 243 *function:* fetch ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 690 *function:* fetchDataSet ( ) @ /var/www/siremis/openbiz/bin/easy/FormRenderer.php 69 *function:* renderSmarty ( Object(EasyForm), "/var/www/siremis/siremis/modules/ser/template/grid.tpl" ) @ /var/www/siremis/openbiz/bin/easy/FormRenderer.php 45 *function:* render ( Object(EasyForm) ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1648 *function:* renderHTML ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1538 *function:* render ( ) @ /var/www/siremis/openbiz/bin/easy/ViewRenderer.php 83 *function:* renderSmarty ( Object(EasyView), "/var/www/siremis/siremis/modules/ser/template/view.tpl" ) @ /var/www/siremis/openbiz/bin/easy/ViewRenderer.php 40 *function:* render ( Object(EasyView) ) @ /var/www/siremis/openbiz/bin/easy/EasyView.php 274 *function:* _render ( ) @ /var/www/siremis/openbiz/bin/easy/EasyView.php 237 *function:* render ( ) @ /var/www/siremis/openbiz/bin/BizController.php 221 *function:* renderView ( "ser.view.UacregListView", "", "", Null, "" ) @ /var/www/siremis/openbiz/bin/BizController.php 107 *function:* dispatchRequest ( ) @ /var/www/siremis/openbiz/bin/BizController.php 32 *function:* include_once ( "/var/www/siremis/openbiz/bin/BizController.php" ) @

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[CFP] reminder! FOSDEM RTC dev-room talks: deadline tomorrow
by FOSDEM RTC Team 16 Nov '16

16 Nov '16

Reminder: speaker's deadline tomorrow, 17 November at 23:59 UTC The Free RTC dev-room has already received some really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the original call for participation: https://danielpocock.com/fosdem-2017-rtc-cfp We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

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Re: [SR-Users] msilo
by Federico Cabiddu 16 Nov '16

16 Nov '16

Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" <volga629(a)skillsearch.ca> wrote: Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. ------------------------------ *From: *"Daniel-Constantin Mierla" <miconda(a)gmail.com> *To: *"sr-users" <sr-users(a)lists.sip-router.org> *Sent: *Tuesday, 15 November, 2016 03:00:19 *Subject: *Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 16 Nov '16

16 Nov '16

Hello Everyone, I am trying just forward MESSAGE through kamailio and I reported issue with mod_sms in freeswitch https://freeswitch.org/jira/browse/FS-9701. So I was need add work around on kamailio side. The issue that my setup MSILO should work in intercept mode meaning check when user online of offline and act as m_store() or m_dump() without rely on lookup("location");, because registration on B2BUA side. If m_store() will have additional parameter to store body this will allow check user state and send when need it. On reply I put something like this to fix To, as work around and forward MESSAGE to client, but when client send MESSAGE kamailio receive on public interface and push out of public interface instead RELAY through local interface and then to B2BUA. route[MESSAGE_FORWARD] { xlog("L_INFO", "Incoming new SMS [$rm] from $tU --> $td\n"); if(fnmatch("$rs", "202") && fnmatch("$rr", "Accepted") && allow_trusted("$si", "$proto")) { $avp(from) = $fu; $avp(to) = $tu; $avp(new_to) = "<" + $avp(to) + ">" + ";messagetype=SMS"; xlog("L_INFO", "Message Accepted by B2BUA --> [$rs] with [$rr] from [$avp(from)] to [$avp(to)]\n"); if(fnmatch("$fU", "offline")) { xlog("L_INFO", "OFFLINE SMS: from [$fU] user --> $fU\n"); $avp(from) = $(avp(to){uri.user}); xlog("L_INFO", "OFFLINE SMS: Searching destination user $avp(oexten)\n"); sql_pvquery("cb", "select contact from location where username = '$avp(from)'","$avp(dst)"); xlog("L_INFO", "OFFLINE SMS: set new destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } $sht(a=>to) = $avp(new_to); xlog("SMS from --> $fU domain $fd\n"); xlog("-------------------------------------\n"); xlog("L_INFO", "FROM --> $avp(from)\n"); xlog("L_INFO", "OLD_TO --> $avp(to)\n"); xlog("L_INFO", "NEW_TO --> $avp(new_to)\n"); } if(compare_ips("$td", "10.18.130.27")) { xlog("L_INFO", "Message from B2BUA contain domain --> $td . Updating...\n"); $avp(new_dst) = $(hdr(Route){param.value,received}); xlog("L_INFO", "New destination --> $avp(new_dst)\n"); $avp(new_to) = "<" + $avp(new_dst) + ">" + ";messagetype=SMS"; $sht(a=>to) = $avp(new_to); remove_hf("To"); insert_hf("To: $sht(a=>to)\r\n", "To"); $du = $sht(a=>to); } } From: "Federico Cabiddu" <federico.cabiddu(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, 15 November, 2016 11:07:44 Subject: Re: [SR-Users] msilo Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

Hello I have tried $Rp before sending this to mailing list, but with $Rp I have different values for example 188, 6847, 3163, 9175 for every call ?! Don't know where are they come from, because in URI I see :6060, and using $rp give my 6060 for examples with $Rp. Only have problem with UAC Panasonic-MPR12-V004.41009/VSIPGW-V3.0000 with doesn't send $rp in URI. Greetings

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RTPProxy benchmark
by Gholamreza Sabery 15 Nov '16

15 Nov '16

I managed to create about 1900 concurrent calls using a single Kamailio and RTPProxy server. But after this number RTPProxy returns 0 and the following error is shown in the Kamailio log files: incorrect port 0 in reply from rtp proxy What is the problem here? Also number of file descriptors that RTPProxy can use are set to a million.

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How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

Hello My point is to log INVITEs in my Network. My SIP Network work on port 6060. Im am using HEP to do that, and hep clients that are listening on port range 5060-6066 to detect some SIP attack to port 5060 and others. But when I have attack to port 5060 I don't want to insert that INVITE to my "good traffic" table, but place it to "fraud" table. Some devices like Panasonic PBX send INVITE to port 6060 so it is "good traffic" but there is no port 6060 in URI, so I can't detect it in right way, because $dp or $rp are 5060 by default. This kamailio work as capture server as "promiscuous_on", so I can't use any "force_rport" etc. because I am only listening. Tcpdump for that SIP session show me that client send traffic to 6060, but I can't get that information from INVITE header. Greetings

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[CFP] FOSDEM 2017, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 15 Nov '16

15 Nov '16

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2017 takes place 4-5 February 2017 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 4 February 2017. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 17 November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM17/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2017/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 31 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 4 February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists See the mailing list discussion for more details about volunteering: https://lists.fsfe.org/pipermail/free-rtc/2016-October/000285.html Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 2 and 3 February 2017 http://wiki.xmpp.org/web/Summit_21 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially in collaboration with the XMPP summit. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 3 February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

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INVITE timeout timer
by Diogenes Aviles 15 Nov '16

15 Nov '16

I have changed the following tm parameters in order to have an timeout INVITE equal to 20 sec (20000 ms). # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 0) # default retransmission timeout: 20sec modparam("tm", "fr_timer", 20000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) However, the timeout is ever 3 min (180 sec.) In accord to tm module the timeout is defined only by "r_timer" Then, I don't know where is the problem. Thanks a lot for your help. -- Diogenes

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msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava.

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 14 Nov '16

14 Nov '16

Hello Daniel, I resolved all configuration issues and right now call goes both directions include rtp. Again huge thank you for help. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 12:24:57 Subject: Re: [SR-Users] BYE dispatcher Hello, this logic is definitely wrong -- FreeSwitch can send also a request, it means that you send it back to it. Only the initial request of a dialog should be routed with rules like dispatcher/load balancer/least cost routing/... The requests within dialog should be routed based on loose routing. Of course, one can think of exceptions, but then you should be fully aware of what kind of routing you do. Cheers, Daniel On 10/11/16 16:25, Slava Bendersky wrote: Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 1

Homer's kamailio config + enabling siptrace module
by Ján Füri 11 Nov '16

11 Nov '16

Hi guys, I'm using captagent on my all kamalios and freeswitches to capture and send sip to homer server. That works very well. But I would like to enable, on homer's kamailio, the module SipTrace (http://www.kamailio.org/docs/modules/4.4.x/modules/siptrace.html) and save captured data into mysql as well. I already set it up using this mysql structure : http://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp1279328 Almost everything works except the column "DIRECTION". Each row is saved as "IN" direction. Do you guys think that can be solved ? If so, can you pleas direct me HOW ... Thx Jan F.

2 3

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 11 Nov '16

11 Nov '16

Hello I have some UAC like Panasonic PBX, that send traffic to port 6060 that I am listening on, but that port isn't included to R-URI. I can see that only at tcpdump, or sip_trace from sip_capture module. Kamailio variables like $dp or $rp have default value of 5060. Thank You,

3 2

Execution Error while getting profile size with jsonrpc-s
by Aqs Younas 11 Nov '16

11 Nov '16

Greetings list, I am trying to get profile size with jsonrpc-s module. Below is jsonrpc-s configuration and a curl command to get the profile size. listen=tcp:0.0.0.0:5060 loadmodule "xhttp" loadmodule "jsonrpc-s" modparam("xhttp", "url_match", "^/rpc_path/") modparam("jsonrpc-s", "pretty_format", 1) event_route[xhttp:request] { if($hu =~ "^/rpc_path") jsonrpc_dispatch(); else xhttp_reply("404", "Not Found", "text/html", ""); } curl -v -H 'Content-Type: application/json' -H 'Call-ID: abc12' -d '{"jsonrpc": "2.0", "method": "dlg.profile_get_size","params":{"accountno": "1234567"}, "id": "0deadb33f"}' http://127.0.0.1:5060/rpc_path/ I am getting this in respose. * Hostname was NOT found in DNS cache * Trying 127.0.0.1... * Connected to 127.0.0.1 (127.0.0.1) port 5060 (#0) > POST /rpc_path/ HTTP/1.1 > User-Agent: curl/7.38.0 > Host: 127.0.0.1:5060 > Accept: */* > Content-Type: application/json > Call-ID: abc12 > Content-Length: 106 > * upload completely sent off: 106 out of 106 bytes *< HTTP/1.1 404 Profile not found: 1234567* < Sia: SIP/2.0/TCP 127.0.0.1:49374 < Call-ID: abc12 < Content-Type: application/json * Server kamailio (4.4.1 (x86_64/linux)) is not blacklisted < Server: kamailio (4.4.1 (x86_64/linux)) < Content-Length: 106 < { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": "0deadb33f" * Connection #0 to host 127.0.0.1 left intact Whereas fifo command gives me this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo profile_get_size accountno 1234567 profile:: name=accountno value=1234567 count=2 Why jsonrpc-s is search for profile 1234567 whereas profile should be accountno. Anyhelp what i am doing wrong here is much appreciated. Best Regards.

2 4

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 11 Nov '16

11 Nov '16

Hello Daniel, I really ask for help, here are configuration file https://paste.fedoraproject.org/477652/88413891/ I spent quite a lot of time trying understand loose_route() /record_route() mix. I can get signalling working, call is not disconnects, but no RTP. Or I can get rtp and signalling BYE is not routed properly. My setup is just proxy all requests to freesiwtch boxes base on dispatcher selection where kamailio setup with 2 interfaces public and private. I really appreciate on you time and help. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Cc: miconda(a)gmail.com Sent: Thursday, 10 November, 2016 23:54:40 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, What From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 11:25:19 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio 4.4 VoiceMail
by Gonzalo Gasca Meza 10 Nov '16

10 Nov '16

I have the following call flow: INVITE -> sip:gonzalo@sip.parzee.io -- TLS/TCP/UDP -> *KAMAILIO* - DB Lookup -> INVITE sip:gonzalo58@test.external.com;transport=tls (Phone1) If Phone1 is Busy or No answer, I want call to go to VM. Phone1, is not registered to Kamailio, nor I'm using Realtime Integration. This Phone1 is registered to an external PBX. Currently in sample configuration script, seems to be that value: $avp(oexten) is used to redirect to VM, but in my case this value is null. I didnt find any documentation for this. *Questions:* a) What is $avp(oexten) ? b) What is the best way to pass a Redirect number in SIP INVITE to VoiceMail system (Asterisk or Freeswitch) c) Is there a way to configure CFNA timer per alias/uri ? *Example*: route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { xdbg("SIP Request: method [$rm] from [$fu] to [$tu]\n"); xlog("VoiceMail routing enabled $avp(oexten)\n"); if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("SCRIPT: VoiceMail to $tu\n"); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; }

2 2

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 1

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 10 Nov '16

10 Nov '16

Hello Thank You. In using Kamailio only as: route { if (is_method("INVITE")) { sql_query("cb", "insert into invites (from_user,to_user,callid,src_ip,dst_ip,src_port,dst_port) values ('$fU','$rU','$ci','$si','$rd','$sp','$rp')"); } } I know that sip_capture(); In some way get real DST port even if there is no port in r-URI. Greetings

2 1

Change CLID
by Nelson Migliaro 10 Nov '16

10 Nov '16

Hello, I have several SIP vendors and I must present a different CLID for each of them. I am planning to use failure routers in Kamailio to route calls to a different vendor in case of failure. The problem I face is changing clid in each route. I am planning to use a mysql database to select the clid based on a prefix and sip vendor and in case of failure, select new clid base on new sip vendor. Is mysql the right way to do that, is there a module for this? In order to change the clid I am thinking of using: uac_replace_to(display,uri). Is this correct way to do that? And finally, in case of using uac_replace_to(display,uri), do I have to use uac_restore_from() to restone the correct from to avoid further issues? Thank you in advance Nelson.-

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

usrloc module expires column issues
by Grant Bagdasarian 10 Nov '16

10 Nov '16

Hello, When saving in the location table, the expires and last_modified columns contain the value 000-00-00 00:00:00 which causes issues when Kamailio is restarted and the preload modparam is set to load the location table from db in memory at startup. I've already upgraded Kamailio to 4.4.3, and used the dbtables script from version 4.4 to recreate the location table, but without success. Anyone else having this issue? Any resolution for it? Regards, Grant Bagdasarian CM

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Here more clear trace https://www.dropbox.com/s/2z6ck71ulidqelh/kamailio-BYE-flow.png?dl=0 BYE send by kamailio to wrong freeswitch box. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 20:09:53 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Kamailio v4.4.4 Released
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, Kamailio SIP Server v4.4.4 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.4. For more details about version 4.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/11/kamailio-v4-4-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. for ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info againts a gateway.? Or any way to insert early dialog into database with callee_contact not empty? Best Regards, Aqs Younas

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Planning the release of Kamailio v4.4.4
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, I am considering to package a new release out of latest stable branch 4.4, respectively v4.4.4. The target date is one of the days between Nov 8-10, a matter of work load that needs to be done for getting it out. As usual, if you are aware of any issue not yet reported on github bug tracker, do it as soon as possible to give it a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

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Re: [SR-Users] fnmatch
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Daniel, Yes, reply was 202 not 200. Right now it work as expected. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, 7 November, 2016 04:51:11 Subject: Re: [SR-Users] fnmatch Hello, for clarification -- what's the actual problem here? fnmatch("$rs", "200") is matching the 202? Cheers, Daniel On 07/11/16 08:52, Serge S. Yuriev wrote: Hello, In debug you can see 202 Accepted but match configured on 200. Not matched -- Wbr, Serge via mobile 07.11.2016, 01:08, "Slava Bendersky" <volga629(a)skillsearch.ca> : BQ_BEGIN Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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BYE dispatcher
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). client_public_ip.28972 > proxy_public_ip.llrp: [udp sum ok] UDP, length 660 E....2..5...c....E.\q,.....CBYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Via: SIP/2.0/UDP client_public_ip:28972;branch=z9hG4bK1248337638;rport Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x10, ttl 64, id 10813, offset 0, flags [none], proto UDP (17), length 930) 10.18.130.27.sip > 10.18.130.24.5160: [bad udp cksum 0x1bf7 -> 0xd85c!] UDP, length 902 E...*=..@.4. ... ......(....BYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Record-Route: <sip:10.18.130.27;r2=on;lr=on;ftag=487304085> Record-Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=487304085> Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 69 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x0, ttl 64, id 38202, offset 0, flags [none], proto UDP (17), length 587) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 559 E..K.:..@... ... ....(...7i.SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x10, ttl 64, id 782, offset 0, flags [none], proto UDP (17), length 503) proxy_public_ip.llrp > client_public_ip.28972: [bad udp cksum 0x4c50 -> 0xd018!] UDP, length 475 E.......@.+}.E.\c.....q,..LPSIP/2.0 200 OK Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x0, ttl 64, id 10584, offset 0, flags [none], proto UDP (17), length 764) 10.18.130.26.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 736 E...)X..@.6@ ... ....(....Y^BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974;alias=client_public_ip~48147~1 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.26:5160;rport;branch=z9hG4bK33785247SrX3H Route: <sip:10.18.130.27;r2=on;lr=on;ftag=d2e84f4a> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=d2e84f4a> Max-Forwards: 70 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 802, offset 0, flags [none], proto UDP (17), length 741) proxy_public_ip.llrp > client_public_ip.48147: [bad udp cksum 0x4d3e -> 0x4f2d!] UDP, length 713 E...."..@.*{.E.\c.........M>BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H Max-Forwards: 69 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 810, offset 0, flags [DF], proto UDP (17), length 504) client_public_ip.48147 > proxy_public_ip.llrp: [udp sum ok] UDP, length 476 E....*@.4..oc....E.\........SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 IP (tos 0x10, ttl 64, id 63040, offset 0, flags [none], proto UDP (17), length 413) 10.18.130.27.sip > 10.18.130.26.5160: [bad udp cksum 0x19f4 -> 0x6386!] UDP, length 385 E....@..@.j. ... ......(....SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 Slava.

2 1

Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. For ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info against each gateway or every dialog stat insertion in database dialog table. Best Regards.

1 0

using set_body_multipart and append_body_part on on_reply routes
by david 08 Nov '16

08 Nov '16

hello all i'm using kam version 4.4.1 and i would like to generate an ISUP part on the 1xx,2xx responses to a remote carrier since the fsw i send the calls to cannot handle ISUP SDP. despite seeing in the doc that those functions cannot be used there and only in REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE, i tried and it works.... partially, because it seems it doesnt create the last boundary and it creates a next boundary with an empty "Encapsulated multipart part: " this the the route i call from onreply_route route[SIPISUP] { set_body_multipart(); msg_apply_changes(); $var(isup) = "\x01\x12\x49\x01\x0a\x03\x02\x0a\x08\x84\x90\x33\x41 \x72\x17\x01\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x01 \x00"; append_body_part("$var(isup)","application/isup;version=itu-t92 +","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); } am i doing something wrong? how can i set this to end the SDP with last boundary? thanks a lot and regards david

2 3

Re: [SR-Users] kamailio + farm of Elastix + rtp direct
by Kato 08 Nov '16

08 Nov '16

Any comments on this please? Thanks! De: Kato [mailto:elkato@gmail.com] Enviado el: domingo, 6 de noviembre de 2016 11:29 a. m. Para: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Asunto: kamailio + farm of Elastix + rtp direct Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Next Kamailio IRC Devel Meeting, Nov 09, 2016
by Daniel-Constantin Mierla 08 Nov '16

08 Nov '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with alternatives for previous or next day. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016b Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 4

Contacts replacing each other when registering
by Sebastian Damm 08 Nov '16

08 Nov '16

Hi, we are running a Registrar server with 4.4.3, the registrar module is configured with the following parameters: modparam("registrar", "max_contacts", 50) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "received_avp", "$(avp(i:801))") modparam("registrar", "default_expires", 600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 600) modparam("registrar", "gruu_enabled", 0) A customer has configured 4 phones with the same username. Normally, all 4 phones would be visible in the location database, and incoming calls would fork out to all 4 phones. However, only one of the devices is always online, the others always replace each other on each registration. Client details: * All phones are snom 715 devices with the same firmware version. * All phones sit behind the same router. Registrations: client 1: * Via: SIP/2.0/UDP 192.168.208.91:35812;branch=z9hG4bK-4q0mmxgu6gm7;rport * Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 2: * Via: SIP/2.0/UDP 192.168.208.90:41331;branch=z9hG4bK-3cbxdvbmk0di;rport * Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 3: * Via: SIP/2.0/UDP 192.168.208.92:5360;branch=z9hG4bK-4drsogakez94;rport * Contact: <sip:1794388e3@192.168.208.92:5360;line=0n7jfzih>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 4: * Via: SIP/2.0/UDP 192.168.208.93:5260;branch=z9hG4bK-1ci0qia3yxyl;rport * Contact: <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;reg-id=1;q=1.0 Only client 4 is reachable all the time. The other 3 clients replace other registrations. So after client 1 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;q=1;expires=600;received="sip:1.2.3.4:6411";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=303;received="sip:1.2.3.4:8923" And after client 2 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;q=1;expires=600;received="sip:1.2.3.4:49061";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=301;received="sip:1.2.3.4:8923" So you see, one contact stays the same, the other one gets replaced. Does anybody know what could cause this kind of behavior? I think, it could be GRUU. I see that those three clients all send the same GRUU parameter. From my understanding, this UUID should be different from client to client. I don't know whether this is a bug in the snom firmware always sending the same GRUU UUID or whether som "intelligent" SIP aware router is putting this in the Contact header. However, since we have disabled gruu for the registrar module (see above), I think, Kamailio should ignore it. Ideas? Thanks in advance. Best Regards, Sebastian

2 4

TCP Keepalive for NAT'd Clients
by Colin Morelli 08 Nov '16

08 Nov '16

Hey all, Looking to figure out the best way to allow TCP connections to stay alive for NAT'd clients, however, to protect against people just opening TCP connections to the server, I'm hoping to only keepalive TCP connections for connections that have sent an INVITE and received a 200 OK. Does this type of per-socket connection lifetime setting exist? For a bit more background: this is specifically for clients that are using the TCP connection for an INVITE. They don't register with the server, and as a result there's no active flow timer for their connection. Is there a way to enable this type of behavior without registering? Best, Colin

2 2

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 08 Nov '16

08 Nov '16

Hello I am using Kamailio to collect data about active calls. I write INVITEs to database, and delete it on BYE or any REL. I would like to write source and destination port too, but don't know how to get destination port. I have tried variables: $dp, $rp, $Rp, $op But when I don't have port in URI like INVITE IP:PORT, $rp have default value of 5060. I would like to get real UDP port, that INVITE does to. Is it possible using PSEUDO VARIABLES? Greetings Andrzej

2 1

Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB
by Jurijs Ivolga 08 Nov '16

08 Nov '16

Hi, Our Kamailio server is crashing once per week, with following error: Oct 1 06:25:06 kamailio kernel: [26982632.803789] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB Core dump was never created, probably it is because of my environment, but I will try to get it. Server constantly eats memory, maybe some kind of memory leak? Any help is highly appreciated! Jurijs

3 25

NER and ASR
by Gholamreza Sabery 07 Nov '16

07 Nov '16

Are there any real data about number of concurrent calls, number of on-line users, NER (Network Effectiveness Ratio) and ASR (Answer Seizure Ratio) of a real VoIP system? I want this data for making comparisons. Can anyone provide a reference, article, etc..

3 4

cdr on remote POSTGRES server
by beer Ll 07 Nov '16

07 Nov '16

Hi all I have a kamailio server configured with accounting on a separate Postgres SQL server by an internal network with acc module and db_postgres module Everything works good but what can I do if I have problem on the internal network or on the postgres server ? Is it possible cache or save the data if the insert data is failed ?

2 1

fnmatch
by Slava Bendersky 07 Nov '16

07 Nov '16

Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava.

3 2

kamailio + farm of Elastix + rtp direct
by Kato 06 Nov '16

06 Nov '16

Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Kamailio DB lookup
by Gonzalo Gasca Meza 06 Nov '16

06 Nov '16

Hi Kamailions, Kamailio 4.3 will receive calls (SIP TLS) for users in this domain: *sip.parzee.io <http://sip.parzee.io>. *Example sip:gonzalo@sip.parzee.io When call is received by Kamailio I need to overwrite R-URI and transport. I should keep a mapping per user. ( gonzalo -> gonzalo58) Example. sip:gonzalo58@test.external.com;transport=tls I started looking into *dbaliases* table in LOCATION. #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif In dbaliases table I see 4 columns. Is the following configuration correct? *alias_username*: gonzalo58 *alias_domain*: test.external.com;transport=tls *username*: gonzalo *domain*: sip.parzee.io Thanks -Gonzalo

4 5

Malformed To header, or?
by Tobias 04 Nov '16

04 Nov '16

Hi! Having some issues with Kamailio 4.3 reporting a malformed header, but I cannot seem to figure out what's wrong. The problem seen is: Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: tm [t_lookup.c:1050]: t_check_msg(): ERROR: reply cannot be parsed Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:1457]: adjust_clen(): error parsing content-length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:2214]: generate_res_buf_from_sip_res(): error while adjusting Content-Length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [forward.c:760]: do_forward_reply(): building failed And the To header seen at this point is: To: +123456789 <sip:123456789@127.0.0.1>;tag=as24ed5606. version: kamailio 4.3.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 I obviously understand Kamailio views this To header as malformed, but what I cannot understand is why? Is it as simple as the number +123456789 is not enclosed with ", i.e. it should be sent as "+123456789"? A similar issue seen, which for some reason doesn't create an ERROR, is when Kamailio receives a reINVITE looking like this which it's supposed to proxy: To: +123456789 <sip:123456789@127.0.0.1>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. which is rewritten into: To: +123456789 <sip:123456789@127*0-)'.:/)&7'...0;user=phone>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. and sent out.. Thanks for your help, /Tobias

3 13

Configuring/Integrating Kamailio Open IMS with FHoSS
by Olvera, Victor 04 Nov '16

04 Nov '16

How does one configure Kamailio Open IMS and FHoSS? In other words, how does one configure IMS and FHoSS so that when soft calls are made they must be processed by the FHoSS before they go through the Open IMS server? Great regards, Victor Olvera

8 13

nathelper not changes IP
by Yuriy Gorlichenko 03 Nov '16

03 Nov '16

I trying to use fix_nated_sdp with 0x02 and 0x08 flags for changing ip addres and the SDP body part like fix_nated_sdp(10,"1.2.3.4") But for now it beaks SDP At the output i see next o=- 7300689428214760503 2 IN IP4 1.1.1.1 c=IN IP4 1.1.1.1 1.2.3.41.2.3.4 <- is just a line that added after fix_nated_sdp atthe end of SDP body kamailio -v version: kamailio 4.4.3 (x86_64/linux) e91aec flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: e91aec compiled on 07:26:14 Oct 18 2016 with gcc 4.9.2

3 4

Cannot send request to OpenSIPS
by Laura 02 Nov '16

02 Nov '16

Dear support, on my Kamailio server (version: kamailio 4.3.6 (x86_64/linux) ) i see lot of these messages.. Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: cannot write request to `/var/run/opensips/socket': No such file or directory Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: failed to end dialog: Cannot send request to OpenSIPS I have only configured Disptachet (mediaproxy) for write on db.. Any idea ? Thanks Laura

2 4

diff between location and kamctl ul show
by Никитенко Виталий 02 Nov '16

02 Nov '16

Hi all! I have kamailio 4.2.3 with options modparam ( "usrloc", "db_url", DBURL) modparam ( "usrloc", "db_mode", 2) But not all registred users displayed in table "location". When viewed "kamctl ul show" displays Domain :: location table = 1024 records = 1172 max_slot = 5 But "location" table has 651 rows. Why not all users saved to mysql table? Thank!

3 2

Kamailio TLS client config selection
by Sergey Basov 02 Nov '16

02 Nov '16

Hi Daniel. After some try to configure kamailio 4.4.3 to act as SPI TLS client for Cisco SIP TLS gateways I have found one issue. If I do client configuration for tls [client:10.1.23.19:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca1.pem [client:10.1.23.29:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca2.pem [client:default] verify_certificate = no require_certificate = no Kamailo always do default profile selection (I do configuration without server_name or server_id, with it kamailio works fine but there are some troubles to make selection of this parameters from config script, I need additional checks and queries) after some research in tls module source code I have added some debug information in file tls_server.c: if (c->flags & F_CONN_PASSIVE) { state=S_TLS_ACCEPTING; dom = tls_lookup_cfg(cfg, TLS_DOMAIN_SRV, &c->rcv.dst_ip, c->rcv.dst_port, 0, 0); } else { state=S_TLS_CONNECTING; sname = tls_get_connect_server_name(); srvid = tls_get_connect_server_id(); // ------------------------------------------------------------- DBG("Entered client config loockup (c->rcv.dst_port %d)\n", c->rcv.dst_port); DBG("Entered client config loockup (&c->rcv.dst_ip %s)\n", ip_addr2a(&c->rcv.dst_ip)); DBG("Entered client config loockup (c->rcv.src_port %d)\n", c->rcv.src_port); DBG("Entered client config loockup (&c->rcv.src_ip %s)\n", ip_addr2a(&c->rcv.src_ip)); // ------------------------------------------------------------- dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); } After making Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <INVITE> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:9098@10.1.23.19:5061;transport=TLS> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> I see Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 10.1.23.19 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:985]: tcpconn_new(): on port 5061, type 3 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:1295]: tcpconn_add(): hashes: 1394:0:0, 1 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:160]: tls_get_connect_server_name(): xavp with outbound server name not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:140]: tls_get_connect_server_id(): xavp with outbound server id not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:219]: tls_complete_init(): Entered client config loockup (c->rcv.dst_port 40123) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:220]: tls_complete_init(): Entered client config loockup (&c->rcv.dst_ip 10.1.23.23) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:221]: tls_complete_init(): Entered client config loockup (c->rcv.src_port 5061) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:222]: tls_complete_init(): Entered client config loockup (&c->rcv.src_ip 10.1.23.19) Where: &c->rcv.dst_ip 10.1.23.23 - it is my local kamailio tls socket ip address to make tls connect from c->rcv.dst_port 40123 - it is my local kamailio tls socket port &c->rcv.src_ip 10.1.23.19 - ip of my TLS device to make tls connection to c->rcv.src_port 5061 - port of my TLS device to make tls connection to so if I change line dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); to dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.src_ip, c->rcv.src_port, sname, srvid); I got correct client domain selection Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:233]: tls_complete_init(): Using initial TLS domain TLSc<10.1.23.19:5061> (dom 0x7fd2eefa3d68 ctx 0x7fd2ef7e70a8 sn []) Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started Can you look at this code? Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com tel: (+38067) 403-62-54

2 3

Kamailio + IMS + sipml5 call to softphone
by Serhat Guler 01 Nov '16

01 Nov '16

Hi, I have a setup as follows: IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for webrtc calls. Calls(both audio and video) between to sipml5 clients using firefox web browser is possible. The session is setup for the calls from sipml5 to Mercuro, but then there isn't audio flow as the codecs are not compatible. Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS codecs as firefox but this time the session isn't being setup. Boghe replies with "Reason: SIP; cause=488; text="Bad content" " I have seen a similar issue has been mentioned here: https://github.com/c00lz3r0/boghe/issues/157 but the initial invite request from sipml5 does have the SDP with media attributes.

3 8

ACC module and BYE attacks
by Gholamreza Sabery 01 Nov '16

01 Nov '16

If you enable acc module to log CDR data into a database a user can send an infinite number of BYE requests to the server and all of them will be logged into the database as "481 Call/transaction does not exist"! How can we prevent this?

3 5

/usr/sbin/kamailio - unresolvable destination error
by Ginti Saurabh 01 Nov '16

01 Nov '16

Hi, Please help to sort the problem. Getting "Unresolvable destination" error in /var/log/messages.

2 1

Terminating PCSCF unresolvable request URI (Kamailio IMS Websockets)
by Serhat Guler 01 Nov '16

01 Nov '16

Dear all, I am trying to make a call between 2 webrtc clients(sipml5) using my kamailio IMS setup. The invite request fails at the terminating PCSCF with error code "Status: 478 Unresolvable destination (478/TM)". I have traced the packages and saw that the package that arrives at the terminating PCSCF has an unresolvable request URI due to the websocket connection. The package being sent from terminating ICSCF to SCSCF has the "Request-URI: sip:bob@net1.test" The package from terminating SCSCF to terminating PCSCF has the "Request-URI: sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws" I guess the terminating SCSCF replaces the request URI with the one it stored during the registration process, but the terminating PCSCF cannot resolve the host name&port number; therefore, it reports back "Unresolvable destination". Is that really the case happening in here ? How can I get over this problem ? Any ideas are welcome. Cheers, Serhat

2 1

Server error for SIP client
by Hermann Norpois 01 Nov '16

01 Nov '16

Hello, if my SIP client tries to connect ... a server error occured (see below). Could you please help me to figure out the reason? Kamailio is installed on an amazon instance with an elastic IP (52.208.249.214, hostname: synth.ddns.net). Thanks Hermann ubuntu@ip-172-31-32-10:~sudo /etc/init.d/kamailio restart * Stopping Kamailio SIP Server: kamailio [ OK ] * Starting Kamailio SIP Server: kamailio loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 172.31.32.10:5060 tcp: 127.0.0.1:5060 tcp: 172.31.32.10:5060 Aliases: tcp: ip-172-31-32-10.eu-west-1.compute.internal:5060 tcp: localhost:5060 udp: ip-172-31-32-10.eu-west-1.compute.internal:5060 udp: localhost:5060 *: synth.ddns.net:* ubuntu@ip-172-31-32-10:~$ kamctl db show subscriber +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | Johanna | synth.ddns.net | 12345 | | 4fa26cedd456d7105c01a48742e71e42 | 1e249ac341d95701f855fce18bec0fba | NULL | ubuntu@ip-172-31-32-10:~$ sudo ngrep -d any -qt -W byline . port 5060 Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. . U 2016/10/20 13:44:33.727161 52.208.249.214:5060 -> 172.31.32.10:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2605d82fd11bd8f788cfad582884eb6d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.e425ad98e0436abc4661d1452a49670d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2d047656fb9df0edfccd4a40d20237b7.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.f54ab9c35b22814b15acbe16bcae387f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.db5960988332c2f4899a5e1c2b57e3c6.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.62f1191bd5af8480698ce118984536c0.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. 2) My logfile /var/log/kamailio/kamailio.log is empty though /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log And in /etc/kamailio/kamailio.conf: log_facility=LOG_LOCAL0 Why is the logfile empty? Thanks Herman

2 1

Store and add custom variables for sip registrations
by Markus Bönke 01 Nov '16

01 Nov '16

Hallo all, we want to use kamailio as registrar server in front of freeswitch. When a user registers we need to lookup and store some user variables. To every INVITE that is received from a registered user we want to add the variables as x sip headers before we relay them to freeswitch. The question is what is the best way to implement this? We tried to store the variables in a htable on register and then lookup and attach the variables on every INVITE from there. Is this the correct approach, one issue we see is that htable is not writing new inserts directly to the database. I’ve also seen that there is a table location_atts in the kamailio db which looks like that it is intended to store custom key value pairs for registrations, but the seems to be no module which writes to that table, or is there one? Thanks for any help or suggestions Markus

2 1

TSILO module use failure
by Igor Olhovskiy 31 Oct '16

31 Oct '16

Hi! I’m trying to implement scheme much like Federico Cabiddu is described here https://www.youtube.com/watch?v=4XIrR9bwUkM Done a bit modification to his script, cause it’s not fully correct on 4.4 version So, parts of config looks like …. route[REGISTRAR] { … if (!save("location")) { sl_reply_error(); } else { route(PUSHJOIN); } exit; …. } route[LOCATION] { if (!lookup("location")) { …… if (is_method("INVITE")) { send_reply("100", "Trying"); route(SUSPEND); } } else { if (is_method("INVITE")) { if (t_newtran()) { ts_store("sip:$tU@$fd"); } $sht(vtp=>stored::$rU) = 1; } } route(RELAY); } ….. route[SUSPEND] { if(!t_suspend()) { send_reply("501", "Unknown destination"); exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); } # append branches or resume the transaction route[PUSHJOIN] { … #if was suspended - recover, if not - ts_append ….. ts_append(«location", "sip:$tU@$fd"); return; … t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } # lookup and relay after resuming transaction route[INVRESUME] { lookup("location"); t_relay(); ts_store("sip:$tU@$fd"); $sht(vtp=>stored::$rU) = 1; } Branches on ts_store are saved in format (sip:username@domain) and also in this format they are added on ts_append. All regarding to routes working ok, except on ts_append I got ERROR: tm [t_append_branches.c:168]: t_append_branches(): ERROR: t_append_branch: failure to add branches (-1) Also, it’s a mixed environment, like wss + udp + tcp. Any idea why? Thanks. -- Best regards, Igor

2 3

Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 30 Oct '16

30 Oct '16

Hi all, I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great. Thanks, Serhat On 24 October 2016 at 21:00, Serhat Guler <srtguler(a)gmail.com> wrote: > Hi Daniel, > > I am using only record_route() without any parameters. I do not have a > proper computer atm to draw the network diagram, but I can tell you shortly > about the network setup. > > I have only enabled websockets for the pcscf to allow ws and wss > connections. In that case there is a ws connection that uses UDP protocol. > This is the ACK to complete the session setup. > > the sipml5 client is configured as follows: > WebSocket Server URL: ws://192.168.0.11:880 > SIP outbound Proxy URL: udp://192.168.0.11:4060 > > Mercuro IMS client: uses UDP port as well: 4060 > > The call is made from sipml5 client. The Mercuro phone rings, and when I > reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from > sipml5 doesn't pass the PCSCF as I explained in the previous message. > > A part of PCSCF cfg file: > > # Check for Subsequent requests: > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if ($route_uri =~ "sip:mo@.*") { > setflag(FLT_MO); > } > if(!isdsturiset()) { > handle_ruri_alias(); > } > # RTP-Relay, if necessary > route(RTPPROXY); > t_relay(); > } else { > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > xlog("L_INFO", "ACK without matching transaction ... > ignore and discard!!!!!\n"); > # ACK without matching transaction ... ignore and > discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > > Cheers, > Serhat > > > > On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I haven't noticed the log files, it's ok. >> >> From the Route header, I see that there is a proxy that uses WS: >> >> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >> 1nCMEI1mR0RztrB;did=e82.0c3> >> That is the address of the next hop and typically a proxy doesn't use >> websocket connection to another proxy. Can you show a diagram with the sip >> server nodes in your network and what protocols are used between them? >> >> Are you simply use record_route() function, or some other function or >> different parameters to it? >> >> Cheers, >> Daniel >> >> >> On 24/10/16 12:18, Serhat Guler wrote: >> >> Hi Daniel, >> >> Thanks for your reply. I actually attached a log file with debug level 3, >> consisting ACK related messages. If you would like to see more logs, I'll >> send a new log file in the evening. >> >> Cheers, >> Serhat >> >> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> can you get all the log messages for ACK but with debug=3 in the >>> kamailio.cfg? >>> >>> Cheers, >>> Daniel >>> >>> On 23/10/16 22:04, Serhat Guler wrote: >>> >>> Hello, >>> >>> I finally managed to place a call from sipml5 webrtc client to Mercuro >>> IMS client. The phone rings, and when I answer it sends 200 OK to the >>> sipml5 where as sipml5 send back an ACK message which never passes the >>> originating PCSCF. The PCSCF says: >>> >>> 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS >>> connection (id: 0) for WebSocket could not be found >>> 8(3640) ERROR: <core> [msg_translator.c:1947]: >>> build_req_buf_from_sip_req(): could not create Via header >>> 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building >>> failed >>> >>> I doubt that the WebSocket connection is closed, cause when I terminate >>> the call from Mercuro client a bye request is being sent to the sipml5. >>> >>> The ACK package: >>> >>> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. >>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 >>> hG4bKvuly7bmxnN4aqM4zZTIS;rport >>> From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB >>> To: <sip:alice@net1.test>;tag=18823 >>> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c >>> all=no;transport=ws>;+g.oma.sip-im;language="en,fr" >>> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 >>> CSeq: 3887 ACK >>> Content-Length: >>> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> >>> Max-Forwards: 69 >>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >>> 1nCMEI1mR0RztrB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz >>> trB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.1c3> >>> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 >>> Organization: Doubango Telecom >>> >>> Have been thinking for quite a while, but couldn't really find a reason >>> why it wouldn't add the v,a header. A debug 3 level log file is also >>> attached. >>> >>> Thanks in advance, >>> Serhat >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >>> >>> _______________________________________________ SIP Express Router >>> (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cg >>> i-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >> >> >

2 3

digest authentication based on two password columns
by Jayesh Nambiar 28 Oct '16

28 Oct '16

Hello, I was wondering if it is possible to do a failover in digest authentication. For eg: I get two possible valid password values. Can I do a pv_proxy_authenticate in such a way where if authentication fails, I do a pv_proxy_autenticate again on the different value to check if that succeeds? Is it even allowed in digest authentication mechanism. Just getting opinions. Thanks for any help. - Jayesh

3 3

Re: [SR-Users] fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Daniel, No that exactly the one, Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 28 October, 2016 07:37:31 Subject: Re: [SR-Users] fetch contact Hello, the contact address is one for attributes in $ulc(...), iirc, it is the =>addr -- see the wiki for pv cookbook for more details. Or are you looking for something else? Cheers, Daniel On 27/10/16 21:43, Slava Bendersky wrote: Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Errors when parsing Diversion header with multiple comma-separated diversions
by Björn Bylander 28 Oct '16

28 Oct '16

Hello! Given a Diversion header like the following: Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Kamailio 4.4.3 (and probably earlier versions) emit error messages complaining about the comma character. According to RFC 6044 multiple comma-separated diversions in a single Diversion-header is allowed. Full INVITE as an example: INVITE sip:+1234567890@127.0.0.100:5060;user=phone;transport=tcp SIP/2.0 Via:SIP/2.0/TCP 127.0.0.101;branch=z9hG4bKBroadWorks.-1od4eg8-127.0.0.100V5060-0-82844591-13 54681939-1477634439005- From:<sip:+9876543210@127.0.0.101;user=phone>;tag=1354681939-1477634439005- To:<sip:+1234567890@127.0.0.100:5060;user=phone> Call-ID:BW0800390052810161562691307@127.0.0.101 CSeq:82844591 INVITE Contact:<sip:127.0.0.101:5060;transport=tcp> P-Asserted-Identity:<sip:+9876543210@127.0.0.101;user=phone> Privacy:none Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE X-BroadWorks-DNC:network-address="sip:+9876543210@127.0.0.101;user=phone" Accept:application/dtmf-relay,application/media_control+xml,application/sdp, application/x-broadworks-call-center+xml,multipart/mixed Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:267 v=0 o=BroadWorks 100442303 1 IN IP4 127.0.0.101 s=- c=IN IP4 127.0.0.102 t=0 0 m=audio 29778 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 Example log messages: Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:488]: parse_to_param(): ERROR parse_to_param : invalid character ',' in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered>> Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:254]: parse_to_param(): ERROR: parse_to_param :unexpected char ["] in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered,>> . Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:906]: parse_addr_spec(): ERROR: parse_to: invalid To - unexpected end of header in state 21 Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_diversion.c:69]: parse_diversion_header(): ERROR:parse_diversion_header: bad diversion header Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (562,21-562,35) The code on line 562 is $avp(div_user) = $(di{uri.user});. Can this be handled in any way by Kamailio without changing the parser? Best regards, Björn Bylander

2 1

fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks

2 1

logfile is empty
by Hermann Norpois 27 Oct '16

27 Oct '16

my logfile /var/log/kamailio/kamailio.log is empty. 1) /etc/rsyslog.d/kamailio.cfg exists with the entry: log_facility=LOG_LOCAL0 2) /etc/rsyslog.con has the entry: # # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log # local0.* -/var/log/kamailio/kamailio/kamailio.log Alternatively I edited /etc/rsyslog.d/kamailio.conf that should be used by rsylog.conf 3) /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log I am working with ubuntu on an amazon instance. Why is the logfile empty? Thanks Herman

3 2

How do I see the full SDP?
by Martin Taschen 27 Oct '16

27 Oct '16

Hello, kamailio seems to work fine. After registration I would like to see the complete SDP of an user. . How is this done? Thanks Martin Taschen

2 1

Re: [SR-Users] sr-users Digest, Vol 137, Issue 25
by surendra tiwari 27 Oct '16

27 Oct '16

i used the cnxcc module its successfully terminated the call. but if you want i can help you. On Thu, Oct 27, 2016 at 12:58 AM, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: cnxcc doesn't terminate calls (Igor Potjevlesch) > 2. logfile is empty (Hermann Norpois) > 3. Re: Out of memory in UB 210: OOM killed process 12261 > (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB > (Jurijs Ivolga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2016 12:48:29 +0200 > From: "Igor Potjevlesch" <igor.potjevlesch(a)gmail.com> > To: "'Kamailio $SER$ - Users Mailing List'" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] cnxcc doesn't terminate calls > Message-ID: <009c01d22f76$7d0c9a30$7725ce90$(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > > > Yep, the dialog is created. I put the dialog into DB and see it. > > Nothing appears in the logs when the timer is reached. > > > > I succeed to do the same behaviour with dialog module only and > dlg_set_timeout. Looks to do why I want. > > > > I'm a bit frustrating about cnxcc because I really don't understand why > nothing happen after max_time, but, anyway, it works with dialog only. > > > > Regards, > > > > Igor. > > > > De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part > de Arsen > Envoyé : mercredi 26 octobre 2016 11:15 > À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> > Objet : Re: [SR-Users] cnxcc doesn't terminate calls > > > > Hi Igor, > > > > Make sure that you create a dialog for this call and $dlg_var(key) is > available (This pseudo-variable will be available only for subsequential > requests after doing loose_route().) > > > > Also check kamailio logs maybe you try to terminate call for wrong > customer id. > > > > On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch < > igor.potjevlesch(a)gmail.com <mailto:igor.potjevlesch@gmail.com> > wrote: > > Hi, > > > > I'm testing cnxcc module in order to make a simple call duration > limitation. > > My understanding is to use the function like this : > > > > $dlg_var(customer_id) = "customer-" + $fU; > > $var(max_time) = 5; > > > > if (!cnxcc_set_max_time("$dlg_var(customer_id)", > "$var(max_time)")) > > > > Unfortunately, after 5 seconds, nothing happen. I don't even see an error > from dialog module. > > > > Anyone has a better understanding of the way the module runs? Thank you! > > > > Regards, > > > > Igor. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > Regards, > Arsen. > >

1 0

cnxcc doesn't terminate calls
by Igor Potjevlesch 26 Oct '16

26 Oct '16

Hi, I'm testing cnxcc module in order to make a simple call duration limitation. My understanding is to use the function like this : $dlg_var(customer_id) = "customer-" + $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$dlg_var(customer_id)", "$var(max_time)")) Unfortunately, after 5 seconds, nothing happen. I don't even see an error from dialog module. Anyone has a better understanding of the way the module runs? Thank you! Regards, Igor.

2 2

app_perl modifies Via header?
by Igor Potjevlesch 25 Oct '16

25 Oct '16

Hello, I try to use app_perl and cnxcc module. As soon as I load the module like this: # --------- app_perl params ---- modparam("app_perl", "filename", "/usr/local/etc/kamailio/AAA/aaa-script.pl") modparam("app_perl", "modpath", "/usr/local/lib64/kamailio/perl/") The Via header is modified for each calls: /usr/local/sbin/kamailio[21064]: INFO: <core> [parser/msg_parser.c:355]: parse_headers(): ERROR: bad header field [Via:ASIP/2.0/UDP 91.] /usr/local/sbin/kamailio[21064]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <A> on state 100 What could explain this behaviour? Regards, Igor.

2 4

Kamailio SIP TLS issue
by Sergey Basov 25 Oct '16

25 Oct '16

Hi All I have some strange behavior of kamailio with TLS. I have configured second IP addres on server, added it to tls listener, and tls.cfg file. But when I try to connect using openssl s_client -showcerts -connect 10.1.23.33:5061 -tls1 -state and openssl s_client -showcerts -connect 10.1.23.23:5061 -tls1 -state I see same certificates (sip2 my config samples are bellow) if I make changes in port number (for ip 10.1.23.33 set port 5091 in both config parts) - I see correct certificates. Does anyone have this problem? Thanks in advance. ----- listen section ---- listen=tls:10.1.23.23:5061 listen=tls:10.1.23.33:5061 ----- tls.cfg ------ [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.33:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.23:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip2.key certificate = /etc/kamailio/keys/sip2.crt -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 3

Sending INVITE’s on REGISTER
by Igor Olhovskiy 25 Oct '16

25 Oct '16

Hi! I have this kind of scenario Receive an INVITE -> send it to user, but if possible, if I’ll receive REGISTER with same credentials I’m calling to - drop an INVITE there also, if call was not answered on first location. Means create other INVITE branch async on the fly, if possible. Where at least to look at? Thanks. -- Best regards, Igor

2 2

How compile Kamailio in freebsd : add carrierroute module
by AFRICALLSHOP 25 Oct '16

25 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I do : 1) cd /usr/ports/net/kamailio 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

3 2

How compile Kamailio in freebsd : add carrierroute module
by Abdoul Osséni 24 Oct '16

24 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I did : 1) cd /usr/ports/net/kamailio/ 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

1 0

Rewriting To on BYE
by Charles Phillips 24 Oct '16

24 Oct '16

I need a sanity check if you don’t mind… I am setting up a stateless proxy for several (separate, not load balanced) Asterisk servers behind Kamailio 4.2 (with domain and dispatcher). All works as expected except when called party hangs up, as Asterisk seems to address the BYE “To” (that is sent to the calling party) to the originating domain, and not the IP of the UA (as it does in other cases). When this happens, stateless Kamailio does not seem to know how to route them, which makes sense. I assume I would need to rewrite the “To” field on BYE message so that it contains the UA’s IP (to match the BYE’s that are routed correctly). The RURI header contains the IP/port of the UA, so I am attempting to use it and here is where I am stuck… Here is the subst from onreply_route: if (is_method("BYE")) { xlog("L_NOTICE", "BYE Detected \n"); subst('/^To:(.*)<sip:(.*)@(.*)>(.*)$/To: <sip:\2@$rd:$rp>\4/ig'); } Here is the “To”: To: "Test User 2" <sip:102@zxcv.asdf.qwerty.net>;tag=8AB53900B9646F1DF05330E7FBF846C8 Any obvious glaring mistakes here? Would I then need additional logic to forward the BYE on? My apologies if this is wordy, or nonsensical...

2 1

PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 24 Oct '16

24 Oct '16

Hello, I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says: 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5. The ACK package: ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB To: <sip:alice@net1.test>;tag=18823 Contact: "Bob"<sip:bob@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> Max-Forwards: 69 Route: <sip:mo@192.168.0.11:880 ;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:4060 ;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:4060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached. Thanks in advance, Serhat

2 3

configuration kamailio 4.4.2 with Asterisk 13.10.0
by Serge Gabriel Ahnoux 24 Oct '16

24 Oct '16

Hello I am trying to configure kamailio 4.4.2 with asterisk 13.10.0. My Os is Debian 7.4 I followed the tutorial : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… The two softwares are installed on my server. So they have the same IP adress but the port is different like they said in the tutorial. I use too Mysql to stock users informations I have 2 databases Kamailio and Asterisk. I use ODBC for communicate beetwen asterisk server and his database. However when my sip client (linephone) connects to kamalio, kamailio doesn't forward information to asterisk. I don't kow the reason. So i can send message but not call If someone knows how to configure, please help me. I join my kamailio configuration file. Regards. ahnouxs(a)gmail.com

2 1

Dropping non-provisional replies in onreply_route
by Alex Balashov 21 Oct '16

21 Oct '16

The core documentation says that in a named onreply_route[], only provisional replies can be drop()'d. To drop any reply, it is necessary to use a global onreply_route. Is there any workaround for this, i.e. so I can drop a 2xx reply from a specific TM transaction? The reason is, to know whether to drop it, I need access to either AVPs or, ideally, dialog variables. Since the global onreply_route is executed by the core, I presume I won't have access to anything that persists through TM there. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 5

Edge proxy / Registrar Script
by Infinicalls Infinicalls 21 Oct '16

21 Oct '16

Hi, I am trying to get a simple Edge Proxy - Registrar setup working behind NAT. Initially I tried various scripts - including the one mentioned in the Oubound page and Peter Dunkley' example - but got some errors related to PATH. Can somebody give me a working script of both proxy and registrar? Any help would be gladly appreciated. regards Ganesh Kumar -- --- http://www.infinicalls.com

2 2

Basic TM suspend question
by Alex Balashov 21 Oct '16

21 Oct '16

Hi, If a transaction is currently t_suspended() and has not been t_continued(), will t_check_trans() still find that transaction for retransmission-dampening purposes, e.g. if a retransmission of the same request as the original one is received? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Postgres: Exclusive Transaction Locks vs. Vacuumer
by Stefan Kohlhauser 21 Oct '16

21 Oct '16

Hey everyone! We noticed that under high load of registrations and using the pua module we can reach a state where the PUA table bloats our Postgres to several GB. However, a SELECT on pua only shows a few hundred entries. The other half million or so entries are obsoleted but not cleaned up. The reason for this seems to be that the autovacuumer of the Postgres can't get to work because the PUA table is almost permanently locked. Our Postgres logs are full with errors about the autovacuumer being canceled because it couldn't get to work. Looking at modules/db_postgres/km_dbase.c::db_postgres_start_transaction() I noticed that there are two locking modes used: str lock_write_end_str = str_init(" IN EXCLUSIVE MODE"); str lock_full_end_str = str_init(" IN ACCESS EXCLUSIVE MODE"); According to https://www.postgresql.org/docs/9.2/static/explicit-locking.html these modes mean forbidding everything else except reads and forbidding everything else. However, according to this documentation ROW EXCLUSIVE is the standard lock for entry altering operations (INSERT, UPDATE, DELETE). And this would still allow the autovacuumer to do its job. Is there a specific reason why these more restricting locking modes are used instead of ROW EXCLUSIVE? Thanks. Best regards, Stefan

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TCP FIN after 10 minutes
by gmele 20 Oct '16

20 Oct '16

Hello, we have deployed a Kamailio acting as SIP proxy on a RHEL 7.2 machine. Clients (mainly mobile phones) connects to the proxy using a TLS protected TCP connection. In the kamailio config, we've set : #!ifdef WITH_TLS enable_tls=yes tcp_async=yes tcp_connection_lifetime=3605 tcp_accept_no_cl=yes tcp_crlf_ping = yes #!endif Our problem is that, even if we set the tcp_connection_lifetime to > 1 hour, the tcp connection is closed after 10 minutes: on tcp dumps, we see clearly the TCP FIN sent by machine hosting the kamailio proxy... Setting parameters tcp_keepidle/keepintlv/keepcnt in the kamailio config didn't change the behavior. We also set TCP keepalived at system level, but without result... This TCP closure is causing us lot of problems when calls between 2 UAs last more than 10 minutes because the REINVITE or BYE messages are lost. Also, closing the TCP connection will wake up the mobile app and make it resend a REGISTER, thing we absolutely want to avoid. Is there a parameter we can use to avoid this closure? I had a look in previous posts, found people with the same problem as mine, but no real solution... Thx for your help. Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TCP-FIN-after-10-minutes-tp152646.h… Sent from the Users mailing list archive at Nabble.com.

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