PP1 asterisk # asterisk -r Asterisk 1.2.14, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer f Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.14 currently running on PP1 (pid = 4971) -- Remote UNIX connection Verbosity is at least 3 PP1*CLI> set verbose 11 Verbosity was 3 and is now 11 PP1*CLI> sip debug SIP Debugging enabled 12 headers, 0 lines Reliably Transmitting (NAT) to 24.7.133.237:1204: OPTIONS sip:541@24.7.133.237:1204 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK5ff8956e;rport From: "Unknown" ;tag=as750f4b04 To: Contact: Call-ID: 1f13bfab700cf154150aed5040b81fcd@38.103.144.10 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Feb 2007 10:45:05 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 4.78.224.161:5060: REGISTER sip:4.78.224.161 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK78474ef7;rport From: ;tag=as71f849e3 To: Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1 CSeq: 108 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- PP1*CLI> <-- SIP read from 4.78.224.161:5060: SIP/2.0 200 OK From: ;tag=as71f849e3 To: ;tag=192.168.67.2+1+0+a67f3e5f Server: DC-SIP/2.0 Organization: Content-Length: 0 Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1 CSeq: 108 REGISTER Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK78474ef7 Contact: sip:s@4.78.224.161:5060;expires=3600 --- (10 headers 0 lines) --- Scheduling destruction of call '69555ff937282717319589672fb5bae1@127.0.0.1' in 32000 ms PP1*CLI> <-- SIP read from 24.7.133.237:1204: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK5ff8956e;rport=5060 From: "Unknown" ;tag=as750f4b04 To: ;tag=000c753543badb11abbe005056c00008 Call-ID: 1f13bfab700cf154150aed5040b81fcd@38.103.144.10 CSeq: 102 OPTIONS Contact: Max-Forwards: 70 Server: SIPPER for PhonerLite Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '1f13bfab700cf154150aed5040b81fcd@38.103.144.10' REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 4.78.224.161:5060: REGISTER sip:4.78.224.161 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK202c65f2;rport From: ;tag=as7a9b39c9 To: Call-ID: 766a1e4d3177ae566ecf85126063aef6@127.0.0.1 CSeq: 108 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- PP1*CLI> <-- SIP read from 4.78.224.161:5060: SIP/2.0 409 Conflict Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK202c65f2;rport From: ;tag=as7a9b39c9 To: Call-ID: 766a1e4d3177ae566ecf85126063aef6@127.0.0.1 CSeq: 108 REGISTER Content-Length: 0 --- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161 Destroying call '766a1e4d3177ae566ecf85126063aef6@127.0.0.1' 12 headers, 0 lines Reliably Transmitting (NAT) to 68.218.191.131:50684: OPTIONS sip:543@68.218.191.131:50684;rinstance=654b8b1488039a8f SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4b8aad6d;rport From: "Unknown" ;tag=as11f012aa To: Contact: Call-ID: 73a4757560e2aa046c7b0c7f5fb69f24@38.103.144.10 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Feb 2007 10:45:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 4.78.224.161:5060: REGISTER sip:4.78.224.161 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK6fa656c7;rport From: ;tag=as4f0b7dd1 To: Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1 CSeq: 108 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- PP1*CLI> <-- SIP read from 4.78.224.161:5060: SIP/2.0 409 Conflict Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK6fa656c7;rport From: ;tag=as4f0b7dd1 To: Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1 CSeq: 108 REGISTER Content-Length: 0 --- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161 Destroying call '54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1' PP1*CLI> <-- SIP read from 68.218.191.131:50684: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4b8aad6d;rport=5060 Contact: To: ;tag=072bd245 From: "Unknown";tag=as11f012aa Call-ID: 73a4757560e2aa046c7b0c7f5fb69f24@38.103.144.10 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 --- (12 headers 0 lines) --- Destroying call '73a4757560e2aa046c7b0c7f5fb69f24@38.103.144.10' PP1*CLI> <-- SIP read from 4.78.224.161:5060: INVITE sip:s@4.78.224.161:5060 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: From: Anonymous ;tag=192.168.67.2+1+13d854+7f64318d Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.com CSeq: 1 INVITE Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKd633cbde884b8ecb88b334ec00fceced Contact: sip:anonymous@4.78.224.161:5060 Allow-Events: line-seize Allow-Events: dialog Allow-Events: refer Allow-Events: message-summary Content-Type: application/sdp Content-Length: 144 v=0 o=dllsl3-ntsbc02 2147483647 2147483647 IN IP4 4.78.224.161 s=sip call c=IN IP4 4.78.224.161 t=0 0 m=audio 30064 RTP/AVP 0 a=ptime:20 --- (16 headers 7 lines) --- Using INVITE request as basis request - 415219-3380409986-455979@dllsl3-ntsbc02.une3.com Sending to 4.78.224.161 : 5060 (NAT) Found peer 'suntel-4245' Found RTP audio format 0 Peer audio RTP is at port 4.78.224.161:30064 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for s in from-trunk (domain 4.78.224.161) list_route: hop: Transmitting (NAT) to 4.78.224.161:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKd633cbde884b8ecb88b334ec00fceced;received=4.78.224.161 From: Anonymous ;tag=192.168.67.2+1+13d854+7f64318d To: Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/9048664245-b682d660", "FROM_DID=s") in new stack -- Executing Goto("SIP/9048664245-b682d660", "ivr-3|s|1") in new stack -- Goto (ivr-3,s,1) -- Executing Set("SIP/9048664245-b682d660", "LOOPCOUNT=0") in new stack -- Executing Set("SIP/9048664245-b682d660", "__DIR-CONTEXT=zonemessages") in new stack -- Executing Answer("SIP/9048664245-b682d660", "") in new stack We're at 38.103.144.10 port 12284 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 4.78.224.161:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKd633cbde884b8ecb88b334ec00fceced;received=4.78.224.161 From: Anonymous ;tag=192.168.67.2+1+13d854+7f64318d To: ;tag=as3b456056 Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 160 v=0 o=root 4971 4971 IN IP4 38.103.144.10 s=session c=IN IP4 38.103.144.10 t=0 0 m=audio 12284 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/9048664245-b682d660", "1") in new stack PP1*CLI> <-- SIP read from 4.78.224.161:5060: ACK sip:s@4.78.224.161:5060 SIP/2.0 Max-Forwards: 69 To: ;tag=as3b456056 From: Anonymous ;tag=192.168.67.2+1+13d854+7f64318d Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.com CSeq: 1 ACK Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKde9947c8913a6b421120479bf99d0b64 Contact: sip:anonymous@4.78.224.161:5060 Allow-Events: line-seize Allow-Events: dialog Allow-Events: refer Allow-Events: message-summary Content-Length: 0 --- (13 headers 0 lines) --- -- Executing Set("SIP/9048664245-b682d660", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing Set("SIP/9048664245-b682d660", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("SIP/9048664245-b682d660", "custom/SpanishEnterYourPinSunTel") in new stack -- Playing 'custom/SpanishEnterYourPinSunTel' (language 'en') PP1*CLI> <-- SIP read from 4.78.224.161:5060: BYE sip:s@4.78.224.161:5060 SIP/2.0 Max-Forwards: 69 To: ;tag=as3b456056 From: Anonymous ;tag=192.168.67.2+1+13d854+7f64318d Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.com CSeq: 2 BYE Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKad1d6fb96fe2a550b78211c41adfa17a Contact: sip:anonymous@4.78.224.161:5060 Allow-Events: line-seize Allow-Events: dialog Allow-Events: refer Allow-Events: message-summary Content-Length: 0 --- (13 headers 0 lines) --- Sending to 4.78.224.161 : 5060 (NAT) Transmitting (NAT) to 4.78.224.161:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKad1d6fb96fe2a550b78211c41adfa17a;received=4.78.224.161 From: Anonymous ;tag=192.168.67.2+1+13d854+7f64318d To: ;tag=as3b456056 Call-ID: 415219-3380409986-455979@dllsl3-ntsbc02.une3.com CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/9048664245-b682d660' -- Executing Hangup("SIP/9048664245-b682d660", "") in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9048664245-b682d660' Destroying call '415219-3380409986-455979@dllsl3-ntsbc02.une3.com' PP1*CLI> PP1 asterisk # asterisk -r Asterisk 1.2.14, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.14 currently running on PP1 (pid = 4971) -- Remote UNIX connection Verbosity is at least 11 Destroying call '766a1e4d3177ae566ecf85126063aef6@127.0.0.1' REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 4.78.224.161:5060: REGISTER sip:4.78.224.161 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK47b7ff4b;rport From: ;tag=as124705d0 To: Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1 CSeq: 116 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- PP1*CLI> <-- SIP read from 4.78.224.161:5060: SIP/2.0 409 Conflict Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK47b7ff4b;rport From: ;tag=as124705d0 To: Call-ID: 54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1 CSeq: 116 REGISTER Content-Length: 0 --- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161 Destroying call '54695ca94be9176b1ae0687b4bf0aa07@127.0.0.1' REGISTER 12 headers, 0 lines Reliably Transmitting (NAT) to 4.78.224.161:5060: REGISTER sip:4.78.224.161 SIP/2.0 Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK4920bff9;rport From: ;tag=as60d8295a To: Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1 CSeq: 112 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- PP1*CLI> <-- SIP read from 4.78.224.161:5060: SIP/2.0 409 Conflict Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK4920bff9;rport From: ;tag=as60d8295a To: Call-ID: 69555ff937282717319589672fb5bae1@127.0.0.1 CSeq: 112 REGISTER Content-Length: 0 --- (7 headers 0 lines) --- -- Got SIP response 409 "Conflict" back from 4.78.224.161 Destroying call '69555ff937282717319589672fb5bae1@127.0.0.1' PP1*CLI> <-- SIP read from 4.78.224.161:5060: INVITE sip:s@4.78.224.161:5060 SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;Refresher=uac Supported: timer To: From: Anonymous ;tag=192.168.67.2+1+14dc52+9b54aee Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.com CSeq: 1 INVITE Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK3b6736c57b4df713fc1ab064426e5dcc Contact: sip:anonymous@4.78.224.161:5060 Allow-Events: line-seize Allow-Events: dialog Allow-Events: refer Allow-Events: message-summary Content-Type: application/sdp Content-Length: 144 v=0 o=dllsl3-ntsbc02 2147483647 2147483647 IN IP4 4.78.224.161 s=sip call c=IN IP4 4.78.224.161 t=0 0 m=audio 30082 RTP/AVP 0 a=ptime:20 --- (16 headers 7 lines) --- Using INVITE request as basis request - 415225-3380410145-954166@dllsl3-ntsbc02.une3.com Sending to 4.78.224.161 : 5060 (NAT) Found peer 'suntel-4245' Found RTP audio format 0 Peer audio RTP is at port 4.78.224.161:30082 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for s in from-trunk (domain 4.78.224.161) list_route: hop: Transmitting (NAT) to 4.78.224.161:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK3b6736c57b4df713fc1ab064426e5dcc;received=4.78.224.161 From: Anonymous ;tag=192.168.67.2+1+14dc52+9b54aee To: Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/9048664245-b68419b8", "FROM_DID=s") in new stack -- Executing Goto("SIP/9048664245-b68419b8", "ivr-3|s|1") in new stack -- Goto (ivr-3,s,1) -- Executing Set("SIP/9048664245-b68419b8", "LOOPCOUNT=0") in new stack -- Executing Set("SIP/9048664245-b68419b8", "__DIR-CONTEXT=zonemessages") in new stack -- Executing Answer("SIP/9048664245-b68419b8", "") in new stack We're at 38.103.144.10 port 17826 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 4.78.224.161:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK3b6736c57b4df713fc1ab064426e5dcc;received=4.78.224.161 From: Anonymous ;tag=192.168.67.2+1+14dc52+9b54aee To: ;tag=as09c5756e Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.com CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 160 v=0 o=root 4971 4971 IN IP4 38.103.144.10 s=session c=IN IP4 38.103.144.10 t=0 0 m=audio 17826 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - --- -- Executing Wait("SIP/9048664245-b68419b8", "1") in new stack PP1*CLI> <-- SIP read from 4.78.224.161:5060: ACK sip:s@4.78.224.161:5060 SIP/2.0 Max-Forwards: 69 To: ;tag=as09c5756e From: Anonymous ;tag=192.168.67.2+1+14dc52+9b54aee Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.com CSeq: 1 ACK Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKe308536a98702bb4e9dfb62dd3668492 Contact: sip:anonymous@4.78.224.161:5060 Allow-Events: line-seize Allow-Events: dialog Allow-Events: refer Allow-Events: message-summary Content-Length: 0 --- (13 headers 0 lines) --- -- Executing Set("SIP/9048664245-b68419b8", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing Set("SIP/9048664245-b68419b8", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("SIP/9048664245-b68419b8", "custom/SpanishEnterYourPinSunTel") in new stack -- Playing 'custom/SpanishEnterYourPinSunTel' (language 'en') PP1*CLI> <-- SIP read from 4.78.224.161:5060: BYE sip:s@4.78.224.161:5060 SIP/2.0 Max-Forwards: 69 To: ;tag=as09c5756e From: Anonymous ;tag=192.168.67.2+1+14dc52+9b54aee Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.com CSeq: 2 BYE Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKfc587f1ef7ee6f68eb31e8ddddffc9e0 Contact: sip:anonymous@4.78.224.161:5060 Allow-Events: line-seize Allow-Events: dialog Allow-Events: refer Allow-Events: message-summary Content-Length: 0 --- (13 headers 0 lines) --- Sending to 4.78.224.161 : 5060 (NAT) Transmitting (NAT) to 4.78.224.161:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bKfc587f1ef7ee6f68eb31e8ddddffc9e0;received=4.78.224.161 From: Anonymous ;tag=192.168.67.2+1+14dc52+9b54aee To: ;tag=as09c5756e Call-ID: 415225-3380410145-954166@dllsl3-ntsbc02.une3.com CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/9048664245-b68419b8' -- Executing Hangup("SIP/9048664245-b68419b8", "") in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/9048664245-b68419b8' Destroying call '415225-3380410145-954166@dllsl3-ntsbc02.une3.com' PP1*CLI>