Here is the scenario of the problem: Environment I am sending calls for external DIDs 204.16.49.121 via SIP This problem is reproducible every time. This configuration has worked in the past. We did not make any changes to this setup or the switch that we are aware of that would coincide when this stopped working. Problem A user dials 6366789921 which is a DID provisioned on my switch. On my switch I have this DID call a phone on my network and dial 3142674833 which is a cell phone outside my network. The switch rings the phone on my network and sends a call back out to provider at 204.16.49.121 to ring the cell phone. Both of these events happen as expected today. If I pick up the call on the phone on my network the call connects without problem. If I pick up the call on the forwarded call to the cell phone the call immediately terminates with fast busy. Here is the console output from my server which includes the error I get - (P1_100001 is the phone on my network) -- Called P1_100001 -- Called 3142674833@provider -- SIP/P1_100001-86d5 is ringing -- SIP/provider-aa90 is making progress passing it to SIP/204.16.49.121-0a057668 -- SIP/provider-aa90 answered SIP/204.16.49.121-0a057668 -- Attempting native bridge of SIP/204.16.49.121-0a057668 and SIP/provider-aa90 -- Got SIP response 480 "Temporarily Unavailable" back from 204.16.49.121 == Spawn extension (origination, 6366789921, 2) exited non-zero on 'SIP/204.16.49.121-0a057668' Other Observations This problem happens regardless of what number I use outside my network - not just the cell number. Here is the debug output of one of these calls: voice1*CLI> <-- SIP read from 204.16.49.121:5060: INVITE sip:6366789921@63.246.27.240;user=phone;npdi=yes SIP/2.0 Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1 From: ;tag=SDe9bu301-10000000-0-274604363 To: CSeq: 1 INVITE Min-SE: 1800 Session-Expires: 1800 Supported: timer Contact: Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 Remote-Party-ID: ;party=calling;id-type=subscriber;privacy=off;screen=yes Max-Forwards: 66 Content-Type: application/sdp Content-Length: 285 v=0 o=Sonus_UAC 31958 8114 IN IP4 204.16.49.10 s=SIP Media Capabilities c=IN IP4 204.16.49.10 t=0 0 m=audio 32848 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv a=maxptime:20 --- (14 headers 13 lines)--- Using INVITE request as basis request - SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 Sending to 204.16.49.121 : 5060 (non-NAT) voice1*CLI> Found peer 'provider' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 100 Peer audio RTP is at port 204.16.49.10:32848 Found description format G729 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 6366789921 in origination (domain 63.246.27.240) list_route: hop: Transmitting (no NAT) to 204.16.49.121:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121 From: ;tag=SDe9bu301-10000000-0-274604363 To: Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- voice1*CLI> -- Executing Macro("SIP/204.16.49.121-09fe5f18", "orig_global") -- Executing SetAccount("SIP/204.16.49.121-09fe5f18", "6366789921") in new stack -- Executing Set("SIP/204.16.49.121-09fe5f18", "foo=EVT00001") in new stack -- Executing SetCDRUserField("SIP/204.16.49.121-09fe5f18", "EVT00001") in new stack voice1*CLI> -- Executing Dial("SIP/204.16.49.121-09fe5f18", "SIP/P1_100001&SIP/3142674833@provider|30") -- Called P1_100001 voice1*CLI> We're at 63.246.27.240 port 19642 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (no NAT) to 204.16.49.121:5060: INVITE sip:3142674833@204.16.49.121 SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport From: "6368250852" ;tag=as1c2568a6 To: Contact: C voice1*CLI> all-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 13 Feb 2007 18:34:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 289 v=0 o=root 15800 15800 IN IP4 63.246.27.240 s=session c=IN IP4 63.246.27.240 t=0 0 m=audio 19642 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 3142674833@provider voice1*CLI> <-- SIP read from 204.16.49.121:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport=5060 From: "6368250852" ;tag=as1c2568a6 To: Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 102 INVITE --- (6 headers 0 lines)--- voice1*CLI> -- SIP/P1_100001-40aa is ringing Transmitting (no NAT) to 204.16.49.121:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121 From: ;tag=SDe9bu301-10000000-0-274604363 To: ;tag=as4c1153d2 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- voice1*CLI> <-- SIP read from 204.16.49.121:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport=5060 From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 102 INVITE Server: DC-SIP/1.2 Supported: timer Contact: Content-Type: application/sdp Content-Length: 264 v=0 o voice1*CLI> =- 3380380448 3380380469 IN IP4 204.16.49.10 s=- c=IN IP4 204.16.49.10 t=0 0 m=audio 50254 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- (11 headers 13 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 204.16.49.10:50254 Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- SIP/provider-08d8 is making progress passing it to SIP/204.16.49.121-09fe5f18 We're at 63.246.27.240 port 11876 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Transmitting (no NAT) to 204.16.49.121:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121 From: ;tag=SDe9bu301-10000000-0-274604363 To: ;tag=as4c1153d2 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 15800 15800 IN IP4 63.246.27.240 s=session c=IN IP4 63.246.27.240 t=0 0 m=audio 11876 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - --- voice1*CLI> <-- SIP read from 204.16.49.121:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1080f14c;rport=5060 From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 102 INVITE Server: DC-SIP/1.2 Supported: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=- 3380380448 3380380469 IN IP4 204.16.49.10 s=- c=IN IP4 204.16.49.10 t=0 0 m=audio 50254 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- (12 headers 13 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 204.16.49.10:50254 Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 204.16.49.121, port 5060 Transmitting (no NAT) to 204.16.49.121:5060: ACK sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK2b003b61;rport From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Contact: Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/provider-08d8 answered SIP/204.16.49.121-09fe5f18 We're at 63.246.27.240 port 11876 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 204.16.49.121:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKh6hpmk000gigjesa4480.1;received=204.16.49.121 From: ;tag=SDe9bu301-10000000-0-274604363 To: ;tag=as4c1153d2 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 15800 15801 IN IP4 63.246.27.240 s=session c=IN IP4 63.246.27.240 t=0 0 m=audio 11876 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/204.16.49.121-09fe5f18 and SIP/provider-08d8 set_destination: Parsing for address/port to send to set_destination: set destination to 204.16.49.121, port 5060 We're at 63.246.27.240 port 19642 Adding codec 0x100 (g729) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 13 lines Reliably Transmitting (no NAT) to 204.16.49.121:5060: INVITE sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK0446fd3c;rport From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Contact: Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 287 v=0 o=root 15800 15801 IN IP4 204.16.49.10 s=session c=IN IP4 204.16.49.10 t=0 0 m=audio 32848 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- voice1*CLI> <-- SIP read from 204.16.49.121:5060: ACK sip:6366789921@63.246.27.240 SIP/2.0 Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKu23a89206o4h9dk8g5c1.1 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 From: ;tag=SDe9bu301-10000000-0-274604363 To: ;tag=as4c1153d2 CSeq: 1 ACK Contact: Supported: timer Content-Length: 0 Max-Forwards: 66 --- (10 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 204.16.49.121, port 5060 We're at 63.246.27.240 port 11876 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 204.16.49.121:5060: INVITE sip:6368250852@204.16.49.121:5060 SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1bb72d82;rport From: ;tag=as4c1153d2 To: ;tag=SDe9bu301-10000000-0-274604363 Contact: Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 15800 15802 IN IP4 204.16.49.10 s=session c voice1*CLI> =IN IP4 204.16.49.10 t=0 0 m=audio 50254 RTP/AVP 18 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - --- voice1*CLI> <-- SIP read from 204.16.49.121:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK0446fd3c;rport=5060 From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 103 INVITE Server: DC-SIP/1.2 Supported: timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Content-Length: 264 v=0 o=- 3380380453 3380380527 IN IP4 204.16.49.10 s=- c=IN IP4 204.16.49.10 t=0 0 m=audio 50254 RTP/AVP 18 101 a=sendrecv a=ptime:20 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=silenceSupp:off - - - - --- (12 headers 13 lines)--- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 204.16.49.10:50254 Found description format G729 Found description format telephone-event Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to 204.16.49.121, port 5060 Transmitting (no NAT) to 204.16.49.121:5060: ACK sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK76c43541;rport From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Contact: Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voice1*CLI> <-- SIP read from 204.16.49.121:5060: SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1bb72d82;rport=5060 From: ;tag=as4c1153d2 To: ;tag=SDe9bu301-10000000-0-274604363 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 102 INVITE Server: DC-SIP/1.2 Supported: timer Contact: Content-Length: 0 --- (10 headers 0 lines)--- -- Got SIP response 480 "Temporarily Unavailable" back from 204.16.49.121 set_destination: Parsing for address/port to send to set_destination: set destination to 204.16.49.121, port 5060 Transmitting (no NAT) to 204.16.49.121:5060: ACK sip:6368250852@204.16.49.121:5060 SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK1bb72d82;rport From: ;tag=as4c1153d2 To: ;tag=SDe9bu301-10000000-0-274604363 Contact: Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voice1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 204.16.49.121, port 5060 Reliably Transmitting (no NAT) to 204.16.49.121:5060: BYE sip:3142674833@204.16.49.121:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK7d5b1c1c;rport From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Contact: Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (origination, 6366789921, 2) exited non-zero on 'SIP/204.16.49.121-09fe5f18' voice1*CLI> <-- SIP read from 204.16.49.121:5060: BYE sip:6366789921@63.246.27.240 SIP/2.0 Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKu23a89206o4h9dk8g5c1cd0000010.1 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 From: ;tag=SDe9bu301-10000000-0-274604363 To: ;tag=as4c1153d2 CSeq: 2 BYE Supported: timer Content-Length: 0 Max-Forwards: 69 --- (9 headers 0 lines)--- Sending to 204.16.49.121 : 5060 (non-NAT) Transmitting (no NAT) to 204.16.49.121:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 204.16.49.121:5060;branch=z9hG4bKu23a89206o4h9dk8g5c1cd0000010.1;received=204.16.49.121 From: ;tag=SDe9bu301-10000000-0-274604363 To: ;tag=as4c1153d2 Call-ID: SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- voice1*CLI> <-- SIP read from 204.16.49.121:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 63.246.27.240:5060;branch=z9hG4bK7d5b1c1c;rport=5060 From: "6368250852" ;tag=as1c2568a6 To: ;tag=SDchsb499-31e00005-0-480760441 Call-ID: 7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240 CSeq: 104 BYE Server: DC-SIP/1.2 Supported: timer Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '7081ffbe3abfdcb402c6dc4a66abebc7@63.246.27.240' Destroying call 'SDe9bu301-d9955b296fcb71c337118c9978849e95-v3000i1' voice1*CLI> voice1*CLI> voice1*CLI>