Below is the output of CLI with sip debug on while I call one extension to another and do NOT answer and wait for the NOANSWER voice mail to startup. The last 20 lines are probably the most significant. The others are there incase they are useful for your analysis. Rich ---------------------- bx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> pbx*CLI> sip debug SIP Debugging enabled 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.101.151:5061: OPTIONS sip:192.168.101.151:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK07aa426b;rport From: "asterisk" ;tag=as0147bf2d To: Contact: Call-ID: 6aa0277b3fa713391ba8f7b01f64f874@192.168.101.1 CSeq: 102 OPTIONS User-Agent: figero 2006 Max-Forwards: 70 Date: Fri, 16 Jun 2006 17:23:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pbx*CLI> <-- SIP read from 192.168.101.151:5061: SIP/2.0 200 OK To: ;tag=9413b93b2fe8d347i1 From: "asterisk" ;tag=as0147bf2d Call-ID: 6aa0277b3fa713391ba8f7b01f64f874@192.168.101.1 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK07aa426b Server: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (10 headers 0 lines)--- Destroying call '6aa0277b3fa713391ba8f7b01f64f874@192.168.101.1' Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:5237 sip_reregister: -- Re-registration for 3145630068@sunrocket.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 67.133.234.125:5060: REGISTER sip:sunrocket.com SIP/2.0 Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK114f5304;rport From: ;tag=as110fdf4d To: Call-ID: 2ed766e550b99fec451eb7d26ace0af1@192.168.101.1 CSeq: 110 REGISTER User-Agent: figero 2006 Max-Forwards: 70 xpires: 120 Contact: Event: registration Content-Length: 0 --- pbx*CLI> <-- SIP read from 67.133.234.125:5060: SIP/2.0 200 OK v:SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK114f5304;rport f:;tag=as110fdf4d t: i:2ed766e550b99fec451eb7d26ace0af1@192.168.101.1 CSeq:110 REGISTER User-Agent: figero 2006 m:;expires=90 Event: registration l:0 --- (10 headers 0 lines)--- Scheduling destruction of call '2ed766e550b99fec451eb7d26ace0af1@192.168.101.1' in 32000 ms Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sunrocket.com is 90 sec (Scheduling reregistration in 75 s) Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:5237 sip_reregister: -- Re-registration for 2679661066@sunrocket.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 67.133.234.125:5060: REGISTER sip:sunrocket.com SIP/2.0 Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK59721511;rport From: ;tag=as52781b46 To: Call-ID: 1eef83ea76f0938d5131eadf22d6db7d@192.168.101.1 CSeq: 110 REGISTER User-Agent: figero 2006 Max-Forwards: 70 xpires: 120 Contact: Event: registration Content-Length: 0 --- pbx*CLI> <-- SIP read from 67.133.234.125:5060: SIP/2.0 200 OK v:SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK59721511;rport f:;tag=as52781b46 t: i:1eef83ea76f0938d5131eadf22d6db7d@192.168.101.1 CSeq:110 REGISTER User-Agent: figero 2006 m:;expires=90 Event: registration l:0 --- (10 headers 0 lines)--- Scheduling destruction of call '1eef83ea76f0938d5131eadf22d6db7d@192.168.101.1' in 32000 ms Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sunrocket.com is 90 sec (Scheduling reregistration in 75 s) Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:5237 sip_reregister: -- Re-registration for 08122966@sip.stanaphone.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 204.147.183.18:5060: REGISTER sip:sip.stanaphone.com SIP/2.0 Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK73c68cb0;rport From: ;tag=as62b04a51 To: Call-ID: 28b9be0b53b29f866dd04fc34d3a85a4@192.168.101.1 CSeq: 109 REGISTER User-Agent: figero 2006 Max-Forwards: 70 Authorization: Digest username="08122966", realm="sip.stanaphone.com", algorithm=MD5, uri="sip:sip.stanaphone.com", nonce="4492e7f9cf435d9929fb83d3d36a54520c5b635f", response="b847801cedbda809df8d3f80b456b515", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- pbx*CLI> <-- SIP read from 204.147.183.18:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 68.38.105.1:5060;branch=z9hG4bK73c68cb0;rport=5060 From: ;tag=as62b04a51 To: ;tag=4e487ad2f2fae333606267674b3a79b1.c48e Call-ID: 28b9be0b53b29f866dd04fc34d3a85a4@192.168.101.1 CSeq: 109 REGISTER PortaBilling: available-funds:0.00 currency:USD Contact: ;expires=115 Server: Sip EXpress router (0.9.5-pre1 (i386/linux)) Content-Length: 0 Warning: 392 204.147.183.18:5060 "Noisy feedback tells: pid=21601 req_src_ip=68.38.105.1 req_src_port=5060 in_uri=sip:sip.stanaphone.com out_uri=sip:sip.stanaphone.com via_cnt==1" --- (11 headers 0 lines)--- Scheduling destruction of call '28b9be0b53b29f866dd04fc34d3a85a4@192.168.101.1' in 32000 ms Jun 16 17:23:49 NOTICE[1468]: chan_sip.c:9661 handle_response_register: Outbound Registration: Expiry for sip.stanaphone.com is 115 sec (Scheduling reregistration in 100 s) 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.101.135:5061: NOTIFY sip:1025@192.168.101.135:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK2a764dc8;rport From: "asterisk" ;tag=as21af82df To: Contact: Call-ID: 2990a2ac35689a3219a06ccd2cc64825@192.168.101.1 CSeq: 102 NOTIFY User-Agent: figero 2006 Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:asterisk@ Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '2990a2ac35689a3219a06ccd2cc64825@192.168.101.1' in 15000 ms 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.101.135:5060: NOTIFY sip:1024@192.168.101.135:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK4817e49f;rport From: "asterisk" ;tag=as0b78daca To: Contact: Call-ID: 5c1a9e52528c63761d32530c680b7b54@192.168.101.1 CSeq: 102 NOTIFY User-Agent: figero 2006 Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:asterisk@ Voice-Message: 0/0 (0/0) --- Scheduling destruction of call '5c1a9e52528c63761d32530c680b7b54@192.168.101.1' in 15000 ms pbx*CLI> <-- SIP read from 192.168.101.135:5060: SIP/2.0 200 OK Call-ID: 5c1a9e52528c63761d32530c680b7b54@192.168.101.1 From: asterisk;tag=as0b78daca To: CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK4817e49f Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '5c1a9e52528c63761d32530c680b7b54@192.168.101.1' pbx*CLI> <-- SIP read from 192.168.101.135:5061: SIP/2.0 200 OK Call-ID: 2990a2ac35689a3219a06ccd2cc64825@192.168.101.1 From: asterisk;tag=as21af82df To: CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK2a764dc8 Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '2990a2ac35689a3219a06ccd2cc64825@192.168.101.1' pbx*CLI> <-- SIP read from 192.168.101.135:5060: INVITE sip:1023@asterisk SIP/2.0 Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk From: 1024;tag=36afc-29eeb4 To: CSeq: 100 INVITE Via: SIP/2.0/UDP 192.168.101.135:5060 Contact: 1024 Max-Forwards: 70 User-Agent: Telco Systems AC-211 v4.57.41 Supported: timer,replaces Allow: REFER, UPDATE, INFO, OPTIONS, ACK, INVITE, BYE, CANCEL, NOTIFY Expires: 9999 Content-Type: application/sdp Content-Length: 131 v=0 o=1024 2748083 2748083 IN IP4 192.168.101.135 s=- c=IN IP4 192.168.101.135 t=0 0 m=audio 16384 RTP/AVP 2 0 8 a=ptime:30 --- (14 headers 7 lines)--- Using INVITE request as basis request - abc4dad49d-11902221-62be4-c0a8794b@asterisk Sending to 192.168.101.135 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.101.135:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135 From: 1024;tag=36afc-29eeb4 To: ;tag=as5fa2abd4 Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk CSeq: 100 INVITE User-Agent: figero 2006 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="75cd5a0e" Content-Length: 0 --- Scheduling destruction of call 'abc4dad49d-11902221-62be4-c0a8794b@asterisk' in 15000 ms Found user '1024' pbx*CLI> <-- SIP read from 192.168.101.135:5060: ACK sip:1023@asterisk SIP/2.0 Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk From: 1024;tag=36afc-29eeb4 To: ;tag=as5fa2abd4 CSeq: 100 ACK Via: SIP/2.0/UDP 192.168.101.135:5060 Max-Forwards: 70 Content-Length: 0 --- (8 headers 0 lines)--- pbx*CLI> <-- SIP read from 192.168.101.135:5060: INVITE sip:1023@asterisk SIP/2.0 Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk From: 1024;tag=36afc-29eeb4 To: CSeq: 101 INVITE Via: SIP/2.0/UDP 192.168.101.135:5060 Contact: 1024 Max-Forwards: 70 User-Agent: Telco Systems AC-211 v4.57.41 Supported: timer,replaces Allow: REFER, UPDATE, INFO, OPTIONS, ACK, INVITE, BYE, CANCEL, NOTIFY Expires: 9999 Proxy-Authorization: Digest username="1024",realm="asterisk",uri="sip:1023@asterisk",response="e6856c469c373086e52156f4a06ede5b",nonce="75cd5a0e" Content-Type: application/sdp Content-Length: 131 v=0 o=1024 2748083 2748083 IN IP4 192.168.101.135 s=- c=IN IP4 192.168.101.135 t=0 0 m=audio 16384 RTP/AVP 2 0 8 a=ptime:30 --- (15 headers 7 lines)--- Using INVITE request as basis request - abc4dad49d-11902221-62be4-c0a8794b@asterisk Sending to 192.168.101.135 : 5060 (non-NAT) Found user '1024' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 8 Peer audio RTP is at port 192.168.101.135:16384 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 1023 in home (domain asterisk) list_route: hop: Transmitting (no NAT) to 192.168.101.135:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135 From: 1024;tag=36afc-29eeb4 To: Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk CSeq: 101 INVITE User-Agent: figero 2006 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Macro("SIP/1024-e33e", "voicemail|SIP/1023") in new stack -- Executing Dial("SIP/1024-e33e", "SIP/1023|20") in new stack We're at 192.168.101.1 port 10680 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 12 lines Reliably Transmitting (no NAT) to 192.168.101.151:5060: INVITE sip:1023@192.168.101.151:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8;rport From: "AC211-1" ;tag=as324fb104 To: Contact: Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1 CSeq: 102 INVITE User-Agent: figero 2006 Max-Forwards: 70 Date: Fri, 16 Jun 2006 17:23:52 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 263 v=0 o=root 2594 2594 IN IP4 192.168.101.1 s=session c=IN IP4 192.168.101.1 t=0 0 m=audio 10680 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 1023 pbx*CLI> <-- SIP read from 192.168.101.151:5060: SIP/2.0 100 Trying To: From: "AC211-1" ;tag=as324fb104 Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8 Server: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 --- (8 headers 0 lines)--- pbx*CLI> <-- SIP read from 192.168.101.151:5060: SIP/2.0 180 Ringing To: ;tag=2c35d737b3d0dfd9i0 From: "AC211-1" ;tag=as324fb104 Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8 Server: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/1023-c56f is ringing Transmitting (no NAT) to 192.168.101.135:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135 From: 1024;tag=36afc-29eeb4 To: ;tag=as11fd6592 Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk CSeq: 101 INVITE User-Agent: figero 2006 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- Destroying call '518b57c99-2e09bd50-70310-c0a8794b@asterisk' Destroying call '53258c5ec-86c96e4-8163c-c0a8794c@asterisk' -- Nobody picked up in 20000 ms Reliably Transmitting (no NAT) to 192.168.101.151:5060: CANCEL sip:1023@192.168.101.151:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.101.1:5060;branch=z9hG4bK6a21bdb8;rport From: "AC211-1" ;tag=as324fb104 To: Contact: Call-ID: 4c2b857814eef94546818b9c2d4f50c1@192.168.101.1 CSeq: 102 CANCEL User-Agent: figero 2006 Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '4c2b857814eef94546818b9c2d4f50c1@192.168.101.1' in 15000 ms -- Executing Goto("SIP/1024-e33e", "s-NOANSWER|1") in new stack -- Goto (macro-voicemail,s-NOANSWER,1) -- Executing VoiceMail("SIP/1024-e33e", "u1023") in new stack We're at 192.168.101.1 port 12222 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Reliably Transmitting (no NAT) to 192.168.101.135:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.101.135:5060;received=192.168.101.135 From: 1024;tag=36afc-29eeb4 To: ;tag=as11fd6592 Call-ID: abc4dad49d-11902221-62be4-c0a8794b@asterisk CSeq: 101 INVITE User-Agent: figero 2006 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 207 v=0 o=root 2594 2594 IN IP4 192.168.101.1 s=session c=IN IP4 192.168.101.1 t=0 0 m=audio 12222 RTP/AVP 0 8 3 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=silenceSupp:off - - - - --- pbx*CLI> Disconnected from Asterisk server Executing last minute cleanups pbx root # ============= machine is dead now =============