Apology for not posting too much details. I'm trying to figure it out how the ATA adapter knows which context (from sip.conf) send the call to?
I'm puzzled as I have never encounter this problem before. I have for example two ATA adapters (Linksys and Audiocodes) both register with asterisk per-port and both have FXS/FXO interfaces. In sip.conf [pstn-4444] ... context=incoming ... [pstn-9998] ... context=fax-incoming ... They both register with asterisk just fine. The cheaper one has only one FXO interface and send the call correctly to the interface it is registered to via sip.conf. The higher end ATA Audiocodes has two FXO interfaces and forwards the calls only to ONE context regardless of which interface the all come IN. I captured the traffic via "tcpdump" but I'm not sure how to recognized why and how to call is being forwarded incorrectly from Audiocodes gateway. From Audiocodes: .......a..INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac445090997 Max-Forwards: 70 From: "KMIEC Z" <sip:[email protected]>;tag=1c445087336 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]:5060> Supported: em,100rel,timer,replaces,path,early-session,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-114 FXS_FXO/v.5.60A.030.001 Content-Type: application/sdp Content-Disposition: session Content-Length: 249 v=0 o=AudiocodesGW 445081214 445081091 IN IP4 10.10.0.8 s=Phone-Call c=IN IP4 10.10.0.8 t=0 0 m=audio 6020 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv 14:02:34.305631 IP (tos 0x0, ttl 64, id 37384, offset 0, flags [none], proto UDP (17), length 426) 10.10.0.2.5060 > 10.10.0.8.5060: [udp sum ok] UDP, length 398 e.......@... .. ..........SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.8;branch=z9hG4bKac445090997;received=10.10.0.8 From: "KMIEC Z" <sip:[email protected]>;tag=1c445087336 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 14:02:34.305950 IP (tos 0x0, ttl 64, id 37385, offset 0, flags [none], proto UDP (17), length 804) 10.10.0.2.5060 > 10.10.0.8.5060: [udp sum ok] UDP, length 776 E..$. ....@... .. .........UINVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.0.2:5060;branch=z9hG4bK50dcf744;rport From: "KMIEC Z" <sip:[email protected]>;tag=as6f0a71bb To: <sip:[email protected]:5060> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 Feb 2010 21:02:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 258 ... From Linksys: ..........INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK-a6fea026 From: KMIEC Z <sip:[email protected]>;tag=3da21e945d4dbff6o1 To: <sip:[email protected]> Remote-Party-ID: KMIEC Z <sip:[email protected]>;screen=yes;party=calling Call-ID: [email protected] CSeq: 101 INVITE Max-Forwards: 70 Contact: <sip:[email protected]:5060> Expires: 240 User-Agent: Linksys/SPA3102-5.1.7(GW) Content-Length: 434 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 15099 15099 IN IP4 10.10.0.6 s=- c=IN IP4 10.10.0.6 t=0 0 m=audio 16410 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv 16:00:12.666784 IP (tos 0x0, ttl 64, id 31864, offset 0, flags [none], proto UDP (17), length 432) 10.10.0.2.5060 > 10.10.0.6.5060: [udp sum ok] UDP, length 404 E...|x...@... .. ..........SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK-a6fea026;received=10.10.0.6 From: KMIEC Z <sip:[email protected]>;tag=3da21e945d4dbff6o1 To: <sip:[email protected]> Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 16:00:12.667389 IP (tos 0x0, ttl 64, id 31865, offset 0, flags [none], proto UDP (17), length 732) 10.10.0.2.5060 > 10.10.0.6.5060: [udp sum ok] UDP, length 704 E...|y...@..| .. ..........SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.0.6:5060;branch=z9hG4bK-a6fea026;received=10.10.0.6 From: KMIEC Z <sip:[email protected]>;tag=3da21e945d4dbff6o1 To: <sip:[email protected]>;tag=as2fdf6ea0 Call-ID: [email protected] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 256 Any ideas, how asterisk sip.conf knows how to interpret this incoming data? Which context to select? -- Joseph On 02/17/10 21:18, John Timms wrote: >Your question is a little vague. I assume that you would be looking for the >"GoTo" application. The syntax is explained here: >http://www.voip-info.org/wiki/view/Asterisk+cmd+goto > ><http://www.voip-info.org/wiki/view/Asterisk+cmd+goto>Also, you can look on >page 426 of the Asterisk book, which is really helpful if you're new to >Asterisk. Download it for free from the publisher here: >http://downloads.oreilly.com/books/9780596510480.pdf > ><http://downloads.oreilly.com/books/9780596510480.pdf>John Timms > >-- >John Timms >(864) 416-1809 >johngtimms (at) gmail (dot) com >-- >IT Department - Gnoso Inc. >john (at) gnoso (dot) com >-- >Grapedial- Affordable group phone messaging >www.grapedial.com >john (at) grapedial (dot) com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
