== Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.2.10, 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 1.2.10 currently running on emma (pid = 30593) Verbosity is at least 3 emma*CLI> sip debug SIP Debugging enabled Destroying call '5da954c0064578231984302b20e8f933@sip.broadvoice.com' emma*CLI> <-- SIP read from 147.135.20.128:5060: INVITE sip:202xxxxxxx@10.1.1.26:5060 SIP/2.0 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE From: "Washington DC";tag=ixy0 To: "User" Via: SIP/2.0/UDP 147.135.20.128:5060 Contact: Supported: 100rel Content-Length: 302 Content-Type: application/sdp v=0 o=2475103479 10 10 IN IP4 147.135.20.250 s=- c=IN IP4 147.135.20.250 t=0 0 m=audio 37378 RTP/AVP 0 8 2 18 96 97 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 G729AB/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 --- (10 headers 13 lines)--- Using INVITE request as basis request - 5350194-35@147.135.20.128 Sending to 147.135.20.128 : 5060 (non-NAT) Found peer 'sip.broadvoice.com' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 147.135.20.250:37378 Found description format PCMU Found description format PCMA Found description format G726-32 Found description format G729 Found description format G729AB Found description format iLBC Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 202xxxxxxx in broadvoice (domain 10.1.1.26) list_route: hop: Transmitting (no NAT) to 147.135.20.128:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User" Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Goto("SIP/202xxxxxxx-091b4380", "demo|s|1") in new stack -- Goto (demo,s,1) -- Executing Wait("SIP/202xxxxxxx-091b4380", "1") in new stack -- Executing Answer("SIP/202xxxxxxx-091b4380", "") in new stack We're at 69.xx.xx.xx port 10072 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 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 emma*CLI> v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing Set("SIP/202xxxxxxx-091b4380", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5 -- Executing Set("SIP/202xxxxxxx-091b4380", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing BackGround("SIP/202xxxxxxx-091b4380", "demo-congrats") in new stack -- Playing 'demo-congrats' (language 'en') Retransmitting #1 (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #2 (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- emma*CLI> <-- SIP read from 66.92.174.2:5060: --- (0 headers 0 lines) Nat keepalive --- Retransmitting #3 (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #4 (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #5 (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 12 11:53:04 NOTICE[25003]: chan_sip.c:5336 sip_reregister: -- Re-registration for 202xxxxxxx@sip.broadvoice.com@sip.broadvoice.com REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 147.135.20.128:5060: REGISTER sip:sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP 69.xx.xx.xx:5060;branch=z9hG4bK219f4d58;rport From: ;tag=as43247893 To: Call-ID: 5da954c0064578231984302b20e8f933@sip.broadvoice.com CSeq: 116 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- emma*CLI> <-- SIP read from 147.135.20.128:5060: SIP/2.0 200 OK Call-ID: 5da954c0064578231984302b20e8f933@sip.broadvoice.com CSeq: 116 REGISTER From: ;tag=as43247893 To: Via: SIP/2.0/UDP 10.1.1.26:5060;branch=z9hG4bK219f4d58 Contact: Expires: 30 Event: registration User-Agent: Asterisk PBX Content-Length: 0 --- (11 headers 0 lines)--- Scheduling destruction of call '5da954c0064578231984302b20e8f933@sip.broadvoice.com' in 32000 ms Sep 12 11:53:04 NOTICE[25003]: chan_sip.c:9828 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s) Retransmitting #6 (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 30593 30593 IN IP4 69.xx.xx.xx s=session c=IN IP4 69.xx.xx.xx t=0 0 m=audio 10072 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 12 11:53:10 WARNING[25003]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 5350194-35@147.135.20.128 for seqno 1 (Non-critical Response) Destroying call '5da954c0064578231984302b20e8f933@sip.broadvoice.com' -- Executing BackGround("SIP/202xxxxxxx-091b4380", "demo-instruct") in new stack -- Playing 'demo-instruct' (language 'en') emma*CLI> sip no debug <-- SIP read from 147.135.20.128:5060: CANCEL sip:9999@10.1.1.26:5060 SIP/2.0 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 CANCEL From: "Washington DC";tag=ixy0 To: "User" Via: SIP/2.0/UDP 147.135.20.128:5060 Content-Length: 0 --- (7 headers 0 lines)--- Sending to 147.135.20.128 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 147.135.20.128:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 147.135.20.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128 From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- == Spawn extension (demo, s, 6) exited non-zero on 'SIP/202xxxxxxx-091b4380' emma*CLI> sip no debug <-- SIP read from 147.135.20.128:5060: ACK sip:9999@10.1.1.26:5060 SIP/2.0 Call-ID: 5350194-35@147.135.20.128 CSeq: 1 ACK From: "Washington DC";tag=ixy0 To: "User";tag=as6ff063f6 Via: SIP/2.0/UDP 147.135.20.128:5060;received=10.1.1.26 Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '5350194-35@147.135.20.128' emma*CLI> sip no debug SIP Debugging Disabled emma*CLI>