On Wed, Feb 26, 2014 at 8:10 AM, Haley,Scott A <scott.ha...@edwardjones.com> wrote: > 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:51...@edj.devjones.com SIP/2.0 > > From: "Haley, Scott" > <sip:3145152...@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00 > > To: <sip:51...@edj.devjones.com> > > 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:3145152244@172.17.184.46;transport=tcp> > > Route: <sip:192.168.122.51;transport=tcp;lr;phase=terminating> > > Accept-Language: en;q=1 > > Alert-Info: <cid:internal@edj.devjones.com>;avaya-cm-alert-type=internal > > History-Info: <sip:51...@edj.devjones.com>;index=1 > > History-Info: "51104" <sip:51...@edj.devjones.com>;index=1.1 > > Min-SE: 1200 > > P-Asserted-Identity: "Haley, Scott" <sip:3145152...@edwardjones.com> > > 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:3145152...@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00 > > To: <sip:51...@edj.devjones.com> > > 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:51104@192.168.122.51:5060;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:3145152...@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00 > > To: <sip:51...@edj.devjones.com>;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:51...@edj.devjones.com SIP/2.0 > > From: "Haley, Scott" > <sip:3145152...@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00 > > To: <sip:51...@edj.devjones.com>;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) --- >
You'll want to talk to the FreePBX guys, as you are just hanging up the outbound call. -- 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 -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users