sr-users June 2013

sr-users@lists.kamailio.org

94 participants
170 discussions

memcache problem - reporting protocol error
by Dragos Oancea 06 Jun '13

06 Jun '13

Hello We are having issues with the memcached module (Kamailio 4.0.0) . All of a sudden we are getting this kind of messages in the logs, and the values for the requested keys are not retrieved. ALERT: memcached [memcached.c:189]: ALERT: memcached: mcm_fetch_cmd():1305: memcache(4) protocol error: STORED#015#012VALUE 29515_HA1 0 32#015#012b52299247f16c6adfa4c8cc53b34e74a#015#012END#015#012STORED#015#012 [...] ALERT: memcached [memcached.c:189]: ALERT: memcached: mcm_fetch_cmd():… [View More]

2 2

Re: [SR-Users] Problem with ACK
by phillman25 06 Jun '13

06 Jun '13

Hello Daniel/Stoyan Thanks for your reply, here is a full sip trace from the first INVITE message to the last acknowledge which is sent to PGW. 192.168.10.189 ==> 81.21.38.34 INVITE sip:94294294@81.21.38.34 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: … [View More]696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 265 v=0 o=root 1692903116 1692903116 IN IP4 192.168.10.189 s=Asterisk PBX 1.8.21.0 c=IN IP4 192.168.10.189 t=0 0 m=audio 12584 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 81.21.38.34 ==> 192.168.10.189 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 81.21.38.34 ==> 81.21.38.5 INVITE sip:3005A94294294@81.21.38.5 SIP/2.0 Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 16 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.5 ==> 81.21.38.34 SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Content-Length: 0 81.21.38.5 ==> 81.21.38.34 INVITE sip:7000A94294294@81.21.38.34:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 15 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.34 ==> 81.21.38.5 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Server: kamailio (4.0.1 (x86_64/linux)) Content-Length: 0 81.21.38.34==>81.21.38.55 INVITE sip:94294294@81.21.38.55 SIP/2.0 Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0 Via: SIP/2.0/UDP 81.21.38.5:5060;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK7ddbce3f;rport=5060 Max-Forwards: 14 From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34> Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE User-Agent: FPBX-2.8.1(1.8.21.0) Date: Thu, 06 Jun 2013 09:40:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 277 P-hint: outbound P-hint: outbound v=0 o=root 1692903116 1692903116 IN IP4 81.21.38.34 s=Asterisk PBX 1.8.21.0 c=IN IP4 81.21.38.34 t=0 0 m=audio 38796 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes 81.21.38.55 ==> 81.21.38.34 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.ef179a29b1792073be0e0618cf49ac25.0 Via: SIP/2.0/UDP 81.21.38.5:5060 ;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 242 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.55 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.55 t=0 0 m=audio 49156 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 81.21.38.34 ==> 81.21.38.5 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.5:5060 ;branch=z9hG4bKterm-1105c-22498045-94294294-67117 Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes 81.21.38.5 ==> 81.21.38.34 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.4dbc7d09c7e34a53fab5087a11e6f210.0 Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes 81.21.38.34 ==> 192.168.10.189 SIP/2.0 200 OK From: "22498045"<sip:22498045@192.168.10.189>;tag=as181922af To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096 Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.10.189:5060;rport=5060;branch=z9hG4bK7ddbce3f Supported: replaces Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af> Record-Route: <sip:94294294@81.21.38.5;pgw-call=call-2aa6> Record-Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641> Contact: <sip:94294294@81.21.38.55> Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO, REFER, NOTIFY, UPDATE Content-Type: application/sdp Content-Length: 260 v=0 o=Dialogic_SIP_CCLIB 64680888 64680889 IN IP4 81.21.38.34 s=Dialogic_SIP_CCLIB c=IN IP4 81.21.38.34 t=0 0 m=audio 47332 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=nortpproxy:yes # U 192.168.10.189:5060 -> 81.21.38.34:5060 ACK sip:94294294@81.21.38.55 SIP/2.0* Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport* Route: <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,< sip:94294294@81.21.38.5 ;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>* Max-Forwards: 70* From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af* To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096* Contact: <sip:22498045@192.168.10.189:5060>* Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* CSeq: 102 ACK* User-Agent: FPBX-2.8.1(1.8.21.0)* Content-Length: 0* 81.21.38.34:5060 -> 81.21.38.5:5060 ACK sip:94294294@81.21.38.55 SIP/2.0* Via: SIP/2.0/UDP 81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0* Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060* Route: <sip:94294294@81.21.38.5 ;pgw-call=call-2aa6>,<sip:81.21.38.34;lr=on;ftag=as181922af>* Max-Forwards: 16* From: "22498045" <sip:22498045@192.168.10.189>;tag=as181922af* To: <sip:94294294@81.21.38.34 >;tag=3d95248-37261551-13c4-50022-1c3096-5cc2673d-1c3096* Contact: <sip:22498045@192.168.10.189:5060>* Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* CSeq: 102 ACK* User-Agent: FPBX-2.8.1(1.8.21.0)* Content-Length: 0* Message: 3 Date: Thu, 06 Jun 2013 16:51:42 +0200 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Problem with ACK Message-ID: <51B0A1FE.6040406(a)gmail.com> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" On 6/6/13 4:34 PM, Stoyan Mihaylov wrote: > We had some similar problems. But what was the actual problem? At least in the two ACKs provided below, loose routing handling with looks correct. Is something that Asterisk doesn't like? Cheers, Daniel > Our configuration is: > SIP client 1 <-> Kamailio <-> Asterisk <->Kamailio<->SIP client 2 > My solution was to check $td and $si and if they are same as Kamailio, > to forward call to Asterisk. > Because I planed to use more then 1 Asterisk, I keep in variable which > one to use. > > > > On Thu, Jun 6, 2013 at 5:28 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > the incoming ACK has the top Route with lr parameter, meaning is > loose routing. By that, the proxy removes the top route header, > preserves the R-URI and sends to the URI in the next Route header. > > From what I can see in the Route stack, it seems a spiral back to > the proxy because ip 81.21.38.34 is two times there. > > If you can't sort it out, send the full SIP trace taken on the > proxy from the initial INVITE to the ACK. Then we can see how > Record-Route headers are set and the signaling flow. > > Cheers, > Daniel > > On 6/6/13 3:30 PM, phillman25 wrote: >> Dear list further to the above problem i observed the following: >> >> ACK message coming from PABX1: >> >> U +0.001877 192.168.10.189:5060 <http://192.168.10.189:5060> -> >> 81.21.38.34:5060 <http://81.21.38.34:5060> >> ACK sip:94294294@81.21.38.55 <mailto:sip%3A94294294@81.21.38.55> >> SIP/2.0* >> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport* >> Route: >> <sip:81.21.38.34;lr=on;ftag=as181922af;did=c83.f641>,< sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6>,< sip:81.21.38.34;lr=on;ftag=as181922af>* >> Max-Forwards: 70* >> From: "22498045" <sip:22498045@192.168.10.189 >> <mailto:sip%3A22498045@192.168.10.189>>;tag=as181922af* >> To: <sip:94294294@81.21.38.34 >> <mailto:sip%3A94294294@81.21.38.34>>;tag=3d95248-37261551- 13c4-50022-1c3096-5cc2673d-1c3096* >> Contact: <sip:22498045@192.168.10.189:5060 >> <http://sip:22498045@192.168.10.189:5060>>* >> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* >> <mailto:696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*> >> CSeq: 102 ACK* >> User-Agent: FPBX-2.8.1(1.8.21.0)* >> Content-Length: 0* >> >> >> >> ACK message sent to PGW from Kamailio1 >> >> U +0.001254 81.21.38.34:5060 <http://81.21.38.34:5060> -> >> 81.21.38.5:5060 <http://81.21.38.5:5060> >> ACK sip:94294294@81.21.38.55 <mailto:sip%3A94294294@81.21.38.55> >> SIP/2.0* >> Via: SIP/2.0/UDP >> 81.21.38.34;branch=z9hG4bKc526.9402c2edbc3ef96d9e405408364506a9.0* >> Via: SIP/2.0/UDP >> 192.168.10.189:5060;branch=z9hG4bK76f8f103;rport=5060* >> Route: <sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6>,< sip:81.21.38.34;lr=on;ftag=as181922af>* >> Max-Forwards: 16* >> From: "22498045" <sip:22498045@192.168.10.189 >> <mailto:sip%3A22498045@192.168.10.189>>;tag=as181922af* >> To: <sip:94294294@81.21.38.34 >> <mailto:sip%3A94294294@81.21.38.34>>;tag=3d95248-37261551- 13c4-50022-1c3096-5cc2673d-1c3096* >> Contact: <sip:22498045@192.168.10.189:5060 >> <http://sip:22498045@192.168.10.189:5060>>* >> Call-ID: 696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060* >> <mailto:696cfa577f42395767bc812e7e8a38a5@192.168.10.189:5060*> >> CSeq: 102 ACK* >> User-Agent: FPBX-2.8.1(1.8.21.0)* >> Content-Length: 0* >> >> >> >> >> Shouldn't the ACK message to the PGW have the header ACK >> sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-2aa6 and the >> Route: <sip:81.21.38.34;lr=on;ftag=as181922af>* ??? >> >> >> >> >> Your help is much appreciated!! >> >> Phillip >> >> >> >> On Thu, Jun 6, 2013 at 12:26 PM, phillman25 <phillman25(a)gmail.com >> <mailto:phillman25@gmail.com>> wrote: >> >> Dear List >> >> I upgraded from Kamailio v 3.3 to 4.0.1 and am now facing an >> issue for the below scenario: >> >> PABX1 ==> Kamailio1 ==> Cisco PGW ==> Kamailio1 ==> PABX2 >> >> >> I understand that this is a hairpin scenario but was working >> normally on v 3.3. >> >> Checking in the syslog i see: >> ERROR: <core> [receive.c:230]: ERROR: receive_msg: no via >> found in reply >> >> Checking the sip trace i see that when calling from PABX1 to >> PABX2. After PABX2 answers and the the 200 OK is eventually >> sent to PABX1 , PABX1 answers with ACK but seems like its >> not sent back to PABX2 as a result PABX resends a 200 OK and >> the cycle continues until PABX2 sends a BYE message. Please >> see below the ACK received from PABX1: >> >> ACK sip:94294294@81.21.38.55 >> <mailto:sip%3A94294294@81.21.38.55> SIP/2.0 >> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK6bffe37c;rport >> Route: >> <sip:81.21.38.34;lr=on;ftag=as1cd4f8f1;did=e36.c471>,< sip:94294294@81.21.38.5 >> <mailto:sip%3A94294294@81.21.38.5>;pgw-call=call-26eb>,< sip:81.21.38.34;lr=on;ftag=as1cd4f8f1> >> Max-Forwards: 70 >> From: "22498045" <sip:22498045@192.168.10.189 >> <mailto:sip%3A22498045@192.168.10.189>>;tag=as1cd4f8f1 >> To: <sip:94294294@81.21.38.34 >> <mailto:sip%3A94294294@81.21.38.34>>;tag=3d94f08-37261551- 13c4-50022-1c1e67-87fe958-1c1e67 >> Contact: <sip:22498045@192.168.10.189:5060 >> <http://sip:22498045@192.168.10.189:5060>> >> Call-ID: 03042a717e27a87e759f7f4879e70377@192.168.10.189:5060 >> <http://03042a717e27a87e759f7f4879e70377@192.168.10.189:5060> >> CSeq: 102 ACK >> User-Agent: FPBX-2.8.1(1.8.21.0) >> Content-Length: 0 >> >> >> Is there an issue with the above ACK message? Is there any >> way to solve this issue quickly perhaps by disabling loose route? >> I have observed that this issue occurs only when hairpinned. >> >> >> Thanking you in advance! >> >> Phillip >> >> >> [View Less]

1 0

Problem with ACK
by phillman25 06 Jun '13

06 Jun '13

Dear List I upgraded from Kamailio v 3.3 to 4.0.1 and am now facing an issue for the below scenario: PABX1 ==> Kamailio1 ==> Cisco PGW ==> Kamailio1 ==> PABX2 I understand that this is a hairpin scenario but was working normally on v 3.3. Checking in the syslog i see: ERROR: <core> [receive.c:230]: ERROR: receive_msg: no via found in reply Checking the sip trace i see that when calling from PABX1 to PABX2. After PABX2 answers and the the 200 OK is eventually sent to … [View More]

3 5

No ACK from Cisco
by Victor V. Kustov 06 Jun '13

06 Jun '13

Hi all! I've got a problem with Kamailio<->Cisco<->PSTN. Called from PSTN: 16:20:14.328786 IP (tos 0x80, ttl 255, id 0, offset 0, flags [none], proto UDP (17), length 1166) 172.16.16.3.58446 > 172.16.17.8.sip: SIP, length: 1138 INVITE sip:599674@172.16.17.8:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Remote-Party-ID: <sip:595311@172.16.16.3>;party=calling;screen=no;privacy=off From: <sip:595311@172.16.16.3>;tag=144D20C-8A7 To: <… [View More]sip:599674@172.16.17.8> Date: Thu, 06 Jun 2013 04:25:44 GMT Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 4251252826-3449229794-2149122083-881571867 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: 70 Timestamp: 1370492744 Contact: <sip:595311@172.16.16.3:5060> Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 279 v=0 o=CiscoSystemsSIP-GW-UserAgent 6723 8551 IN IP4 172.16.16.3 s=SIP Call c=IN IP4 172.16.16.3 t=0 0 m=audio 18550 RTP/AVP 8 18 101 c=IN IP4 172.16.16.3 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 16:20:14.329130 IP (tos 0x10, ttl 64, id 17361, offset 0, flags [none], proto UDP (17), length 327, bad cksum 0 (->bc99)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 299 SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 From: <sip:595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:599674@172.16.17.8> Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Server: kamailio (4.1.0-dev6 (x86_64/freebsd)) Content-Length: 0 16:20:14.335619 IP (tos 0x10, ttl 64, id 17363, offset 0, flags [none], proto UDP (17), length 359, bad cksum 0 (->bc77)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 331 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 From: <sip:595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:599674@172.16.17.8> Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Server: kamailio (4.1.0-dev6 (x86_64/freebsd)) Content-Length: 0 16:20:14.362576 IP (tos 0x10, ttl 64, id 17365, offset 0, flags [none], proto UDP (17), length 623, bad cksum 0 (->bb6d)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 595 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Record-Route: <sip:172.16.17.8;lr=on;did=558.b04> From: <sip:0074832595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:0074832599674@172.16.17.8>;tag=1054623052 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Contact: <sip:0074832599674@10.120.0.18:32225;user=phone> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3140 1.0.7.76 Allow-Events: talk, hold Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 16:20:23.621104 IP (tos 0x10, ttl 64, id 17380, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba50)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.16.3:5060;branch=z9hG4bK2824D9 Record-Route: <sip:172.16.17.8;lr=on;did=558.b04> From: <sip:0074832595311@172.16.16.3>;tag=144D20C-8A7 To: <sip:0074832599674@172.16.17.8>;tag=1054623052 Call-ID: FD659482-CD9711E2-80DC8E65-5DBD56D4(a)172.16.16.3 CSeq: 101 INVITE Contact: <sip:0074832599674@10.120.0.18:32225;user=phone> Supported: replaces, path, timer, eventlist User-Agent: Grandstream GXV3140 1.0.7.76 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 266 v=0 o=0074832599674 8002 8000 IN IP4 10.120.0.18 s=SIP Call c=IN IP4 10.120.0.18 t=0 0 m=audio 39206 RTP/AVP 8 18 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 16:20:24.126589 IP (tos 0x10, ttl 64, id 17383, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba4d)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK 16:20:25.135006 IP (tos 0x10, ttl 64, id 17389, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba47)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK 16:20:27.144412 IP (tos 0x10, ttl 64, id 17395, offset 0, flags [none], proto UDP (17), length 893, bad cksum 0 (->ba41)!) 172.16.17.8.sip > 172.16.16.3.sip: SIP, length: 865 SIP/2.0 200 OK And no ACK from Cisco. Is it Cisco config problem? P.S. no sip-ua configuration, dial-peer voice 10002 voip description ** xxx ** preference 1 destination-pattern 5T voice-class codec 1 session protocol sipv2 session target ipv4:xxx session transport udp I had no idea about no ACK. Maybe OK from Kamailio incorrect? -- WBR, Victor JID: coyote(a)bks.tv JID: coyote(a)bryansktel.ru I use FREE operation system: 3.9.4-calculate GNU/Linux [View Less]

2 5

New developer: Camille Oudot
by Daniel-Constantin Mierla 06 Jun '13

06 Jun '13

Hello, I want to announce that a new person got developer GIT write access to repository: Camille Oudot - from Orange, France. Currently, he is developing mainly on Kamailio IMS modules, several patches being submitted lately by him. His git commit id is: coudot My warm welcome and looking forward to future work within the project! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World … [View More]

2 1

New developer: Torrey Searle
by Daniel-Constantin Mierla 06 Jun '13

06 Jun '13

1 1

Re: [SR-Users] cnxcc developer module pulse concept
by Carlos Ruiz Díaz 06 Jun '13

06 Jun '13

Hi Eduardo, please take into account that there might be telephony concepts mistranslated from Spanish to English. In case you find something else that seems confusing, please let me know. Regards, Carlos On Thu, Jun 6, 2013 at 7:56 AM, Eduardo Lejarreta <lejarreta.e(a)sarenet.es>wrote: > Good evening Carlos.**** > > ** ** > > I’ve found an example on git repository where you have additional > information.**** > > ** ** > > For a credit of 50, a … [View More]

1 0

cnxcc developer module pulse concept
by Eduardo Lejarreta 06 Jun '13

06 Jun '13

Good evening Carlos. I've found an example on git repository where you have additional information. For a credit of 50, a discount per second of 0,5, a initial_pulse of 30 and final_pulse of 6 .. # if only one call is established, that call should last 1m, 36s Now It makes sense to me. Thanks and regards. -- Eduardo Lejarreta. De: Eduardo Lejarreta [mailto:lejarreta.e@sarenet.es] Enviado el: jueves, 06 de junio de 2013 11:40 Para: 'sr-users(a)lists.sip-router.org' … [View More]

1 0

Microseconds not milliseconds in cfgutils usleep
by David K 06 Jun '13

06 Jun '13

Hello, According to this page : http://kamailio.org/docs/modules/stable/modules/cfgutils.html#idp15252400 usleep will sleep for the number of milli-seconds specified. I am familiar with Linux's usleep which accepts milliseconds and therefore had a look at the code in Kamailio, it looks to me that your usleep accepts micro-seconds ( 1/1000000 of a second ) and not milli seconds. Can anyone confirm this? If so, maybe update the doc. Thanks, David

2 1

cnxcc developer module pulse concept
by Eduardo Lejarreta 06 Jun '13

06 Jun '13

Good morning Carlos. On cnxcc_set_max_credit() function I don't understand the concept "pulse" (sure it's something easy but I'm a not aware about this matter) Could you please explain it in an example? Thanks and best regards. -- Eduardo Lejarreta.

1 0

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sr-users June 2013