rendered paste body == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': FoundAsterisk 1.2.10, Copyright (C) 1999 - 2006 Digium, Inc. and others.Created by Mark Spencer <markster@digium.com>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.This is free software, with components licensed under the GNU General PublicLicense version 2 and other licenses; you are welcome to redistribute it undercertain conditions. Type 'show license' for details.=========================================================================Connected to Asterisk 1.2.10 currently running on emma (pid = 30593)Verbosity is at least 3emma*CLI> sip debugSIP Debugging enabledDestroying call '5da954c0064578231984302b20e8f933@sip.broadvoice.com'emma*CLI> <-- SIP read from 147.135.20.128:5060: INVITE sip:202xxxxxxx@10.1.1.26:5060 SIP/2.0Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEFrom: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>Via: SIP/2.0/UDP 147.135.20.128:5060Contact: <sip:202xxxxxxx@147.135.20.128:5060>Supported: 100relContent-Length: 302Content-Type: application/sdpv=0o=2475103479 10 10 IN IP4 147.135.20.250s=-c=IN IP4 147.135.20.250t=0 0m=audio 37378 RTP/AVP 0 8 2 18 96 97 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:2 G726-32/8000a=rtpmap:18 G729/8000a=rtpmap:96 G729AB/8000a=rtpmap:97 iLBC/8000a=rtpmap:101 telephone-event/8000--- (10 headers 13 lines)---Using INVITE request as basis request - 5350194-35@147.135.20.128Sending to 147.135.20.128 : 5060 (non-NAT)Found peer 'sip.broadvoice.com'Found RTP audio format 0Found RTP audio format 8Found RTP audio format 2Found RTP audio format 18Found RTP audio format 96Found RTP audio format 97Found RTP audio format 101Peer audio RTP is at port 147.135.20.250:37378Found description format PCMUFound description format PCMAFound description format G726-32Found description format G729Found description format G729ABFound description format iLBCFound description format telephone-eventCapabilities: 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: <sip:202xxxxxxx@147.135.20.128:5060>Transmitting (no NAT) to 147.135.20.128:5060:SIP/2.0 100 TryingVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>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 stackWe're at 69.xx.xx.xx port 10072Adding codec 0x4 (ulaw) to SDPAdding codec 0x8 (alaw) to SDPAdding non-codec 0x1 (telephone-event) to SDPReliably Transmitting (no NAT) to 147.135.20.128:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240emma*CLI> v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=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 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ----Retransmitting #2 (no NAT) to 147.135.20.128:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=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 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ----Retransmitting #4 (no NAT) to 147.135.20.128:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ----Retransmitting #5 (no NAT) to 147.135.20.128:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - ----Sep 12 11:53:04 NOTICE[25003]: chan_sip.c:5336 sip_reregister: -- Re-registration for 202xxxxxxx@sip.broadvoice.com@sip.broadvoice.comREGISTER 12 headers, 0 linesReliably Transmitting (no NAT) to 147.135.20.128:5060:REGISTER sip:sip.broadvoice.com SIP/2.0Via: SIP/2.0/UDP 69.xx.xx.xx:5060;branch=z9hG4bK219f4d58;rportFrom: <sip:202xxxxxxx@sip.broadvoice.com>;tag=as43247893To: <sip:202xxxxxxx@sip.broadvoice.com>Call-ID: 5da954c0064578231984302b20e8f933@sip.broadvoice.comCSeq: 116 REGISTERUser-Agent: Asterisk PBXMax-Forwards: 70Expires: 120Contact: <sip:9999@69.xx.xx.xx>Event: registrationContent-Length: 0---emma*CLI> <-- SIP read from 147.135.20.128:5060: SIP/2.0 200 OKCall-ID: 5da954c0064578231984302b20e8f933@sip.broadvoice.comCSeq: 116 REGISTERFrom: <sip:202xxxxxxx@sip.broadvoice.com>;tag=as43247893To: <sip:202xxxxxxx@sip.broadvoice.com>Via: SIP/2.0/UDP 10.1.1.26:5060;branch=z9hG4bK219f4d58Contact: <sip:9999@10.1.1.26>Expires: 30Event: registrationUser-Agent: Asterisk PBXContent-Length: 0--- (11 headers 0 lines)---Scheduling destruction of call '5da954c0064578231984302b20e8f933@sip.broadvoice.com' in 32000 msSep 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 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Type: application/sdpContent-Length: 240v=0o=root 30593 30593 IN IP4 69.xx.xx.xxs=sessionc=IN IP4 69.xx.xx.xxt=0 0m=audio 10072 RTP/AVP 0 8 101a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=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/2.0Call-ID: 5350194-35@147.135.20.128CSeq: 1 CANCELFrom: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>Via: SIP/2.0/UDP 147.135.20.128:5060Content-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 TerminatedVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 INVITEUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>Content-Length: 0---Transmitting (no NAT) to 147.135.20.128:5060:SIP/2.0 200 OKVia: SIP/2.0/UDP 147.135.20.128:5060;received=147.135.20.128From: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Call-ID: 5350194-35@147.135.20.128CSeq: 1 CANCELUser-Agent: Asterisk PBXAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContact: <sip:202xxxxxxx@69.xx.xx.xx>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/2.0Call-ID: 5350194-35@147.135.20.128CSeq: 1 ACKFrom: "Washington DC"<sip:202xxxxxxx@147.135.20.128;user=phone>;tag=ixy0To: "User"<sip:9999@10.1.1.26;user=phone>;tag=as6ff063f6Via: SIP/2.0/UDP 147.135.20.128:5060;received=10.1.1.26Content-Length: 0--- (7 headers 0 lines)---Destroying call '5350194-35@147.135.20.128'emma*CLI> sip no debugSIP Debugging Disabledemma*CLI>