I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a "SIP/2.0 603 Declined" message. Here is the output.
Any reason that this might be happening? It has been working up until now this week. I rebooted the machine on Tuesday. <--- SIP read from TCP:172.17.184.46:31285 ---> INVITE sip:[email protected] SIP/2.0 From: "Haley, Scott" <sip:[email protected]>;tag=8066eb6f589ce3124b652973b4b00 To: <sip:[email protected]> Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Max-Forwards: 71 Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Supported: 100rel,histinfo,join,replaces,sdp-anat,timer Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE User-Agent: Avaya CM/R016x.02.0.823.0 Contact: "Haley, Scott" <sip:[email protected];transport=tcp> Route: <sip:192.168.122.51;transport=tcp;lr;phase=terminating> Accept-Language: en;q=1 Alert-Info: <cid:[email protected]>;avaya-cm-alert-type=internal History-Info: <sip:[email protected]>;index=1 History-Info: "51104" <sip:[email protected]>;index=1.1 Min-SE: 1200 P-Asserted-Identity: "Haley, Scott" <sip:[email protected]> Record-Route: <sip:172.17.184.46;transport=tcp;lr> Session-Expires: 1200;refresher=uac Content-Type: application/sdp Content-Length: 257 v=0 o=- 1393419743 1 IN IP4 172.17.184.46 s=- c=IN IP4 172.17.184.93 b=AS:64 t=0 0 a=avf:avc=n prio=n a=csup:avf-v0 m=audio 28196 RTP/AVP 0 18 127 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 <-------------> --- (23 headers 13 lines) --- Sending to 172.17.184.46:31285 (NAT) Using INVITE request as basis request - 8066eb6f589ce3125b652973b4b00 Found peer 'trunk503in' for '3145152244' from 172.17.184.46:31285 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 127 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 127 Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 172.17.184.93:28196 Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com) list_route: hop: <sip:172.17.184.46;transport=tcp;lr> <--- Transmitting (NAT) to 172.17.184.46:31285 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 Record-Route: <sip:172.17.184.46;transport=tcp;lr> From: "Haley, Scott" <sip:[email protected]>;tag=8066eb6f589ce3124b652973b4b00 To: <sip:[email protected]> Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.13.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1200;refresher=uac Contact: <sip:[email protected]:5060;transport=TCP> Content-Length: 0 <------------> -- Executing [51104@from-trunk-sip-trunk503out:1] Set("SIP/trunk503in-0000010b", "GROUP()=OUT_1") in new stack -- Executing [51104@from-trunk-sip-trunk503out:2] Goto("SIP/trunk503in-0000010b", "from-trunk,51104,1") in new stack -- Goto (from-trunk,51104,1) -- Executing [51104@from-trunk:1] Set("SIP/trunk503in-0000010b", "__FROM_DID=51104") in new stack -- Executing [51104@from-trunk:2] Gosub("SIP/trunk503in-0000010b", "app-blacklist-check,s,1") in new stack -- Executing [s@app-blacklist-check:1] GotoIf("SIP/trunk503in-0000010b", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:2] Set("SIP/trunk503in-0000010b", "CALLED_BLACKLIST=1") in new stack -- Executing [s@app-blacklist-check:3] Return("SIP/trunk503in-0000010b", "") in new stack -- Executing [51104@from-trunk:3] Gosub("SIP/trunk503in-0000010b", "cidlookup,cidlookup_1,1") in new stack -- Executing [cidlookup_1@cidlookup:1] GotoIf("SIP/trunk503in-0000010b", "1?cidlookup,cidlookup_return,1") in new stack -- Goto (cidlookup,cidlookup_return,1) -- Executing [cidlookup_return@cidlookup:1] ExecIf("SIP/trunk503in-0000010b", "0?Set(CALLERID(name)=)") in new stack -- Executing [cidlookup_return@cidlookup:2] Return("SIP/trunk503in-0000010b", "") in new stack -- Executing [51104@from-trunk:4] ExecIf("SIP/trunk503in-0000010b", "0 ?Set(CALLERID(name)=3145152244)") in new stack -- Executing [51104@from-trunk:5] Set("SIP/trunk503in-0000010b", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [51104@from-trunk:6] Set("SIP/trunk503in-0000010b", "CALLERPRES()=allowed_not_screened") in new stack -- Executing [51104@from-trunk:7] Goto("SIP/trunk503in-0000010b", "app-blackhole,hangup,1") in new stack -- Goto (app-blackhole,hangup,1) -- Executing [hangup@app-blackhole:1] NoOp("SIP/trunk503in-0000010b", "Blackhole Dest: Hangup") in new stack -- Executing [hangup@app-blackhole:2] Hangup("SIP/trunk503in-0000010b", "") in new stack == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'SIP/trunk503in-0000010b' Scheduling destruction of SIP dialog '8066eb6f589ce3125b652973b4b00' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (NAT) to 172.17.184.46:31285 ---> SIP/2.0 603 Declined Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00 From: "Haley, Scott" <sip:[email protected]>;tag=8066eb6f589ce3124b652973b4b00 To: <sip:[email protected]>;tag=as06e2e068 Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 INVITE Server: FPBX-2.8.1(1.8.13.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from TCP:172.17.184.46:31285 ---> ACK sip:[email protected] SIP/2.0 From: "Haley, Scott" <sip:[email protected]>;tag=8066eb6f589ce3124b652973b4b00 To: <sip:[email protected]>;tag=as06e2e068 Call-ID: 8066eb6f589ce3125b652973b4b00 CSeq: 1 ACK Max-Forwards: 70 Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285 User-Agent: Avaya CM/R016x.02.0.823.0 Route: <sip:192.168.122.51;transport=tcp;lr;phase=terminating> Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Thanks, Scott Haley Edward Jones Investments If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments. 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