Script started on Mon Aug 7 11:31:22 2006 localhost ~ # asterisk -r Asterisk SVN-trunk-r38826, 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 SVN-trunk-r38826 currently running on localhost (pid = 19436) localhost*CLI> Verbosity is at least 3 localhost*CLI> Asterisk Ready.  -- Remote UNIX connection localhost*CLI> se send set localhost*CLI> set verbose 20000000000000000 localhost*CLI> Verbosity was 3 and is now 2147483647 localhost*CLI> sip debug localhost*CLI> SIP Debugging enabled localhost*CLI> <-- SIP read from 192.168.254.128:5060: REGISTER sip:192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-d5faab59 From: 201 ;tag=46a6c0a418facbbbo0 To: 201 Call-ID: a21c7e9a-221c358d@192.168.254.12 CSeq: 21003 REGISTER Max-Forwards: 70 Authorization: Digest username="201",realm="asterisk",nonce="3732c91b",uri="sip:192.168.254.96",algorithm=MD5,response="61237beed466f7a8603c17f99b7229b5" Contact: 201 ;expires=3600 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.254.128 : 5060 (no NAT) Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-d5faab59;received=192.168.254.128 From: 201 ;tag=46a6c0a418facbbbo0 To: 201 Call-ID: a21c7e9a-221c358d@192.168.254.12 CSeq: 21003 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-d5faab59;received=192.168.254.128 From: 201 ;tag=46a6c0a418facbbbo0 To: 201 ;tag=as19295522 Call-ID: a21c7e9a-221c358d@192.168.254.12 CSeq: 21003 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="695c2532" Content-Length: 0 --- Scheduling destruction of SIP dialog 'a21c7e9a-221c358d@192.168.254.12' in 32000 ms (Method: REGISTER) localhost*CLI> <-- SIP read from 192.168.254.128:5060: REGISTER sip:192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-a9aeea8a From: 201 ;tag=46a6c0a418facbbbo0 To: 201 Call-ID: a21c7e9a-221c358d@192.168.254.12 CSeq: 21004 REGISTER Max-Forwards: 70 Authorization: Digest username="201",realm="asterisk",nonce="695c2532",uri="sip:192.168.254.96",algorithm=MD5,response="6b26e5c9f3dd66de52d6adcc6b27a21e" Contact: 201 ;expires=3600 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.254.128 : 5060 (no NAT) Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-a9aeea8a;received=192.168.254.128 From: 201 ;tag=46a6c0a418facbbbo0 To: 201 Call-ID: a21c7e9a-221c358d@192.168.254.12 CSeq: 21004 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- localhost*CLI> Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-a9aeea8a;received=192.168.254.128 From: 201 ;tag=46a6c0a418facbbbo0 To: 201 ;tag=as19295522 Call-ID: a21c7e9a-221c358d@192.168.254.12 CSeq: 21004 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 180 Contact: ;expires=180 Date: Mon, 07 Aug 2006 11:31:37 GMT Content-Length: 0 --- Scheduling destruction of SIP dialog 'a21c7e9a-221c358d@192.168.254.12' in 15000 ms (Method: REGISTER) localhost*CLI> Really destroying SIP dialog 'a21c7e9a-221c358d@192.168.254.12' Method: REGISTER localhost*CLI> <-- SIP read from 192.168.254.126:5060: INVITE sip:201@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-e870924a From: 200 ;tag=b12b7fe817b157bo0 To: Remote-Party-ID: 200 ;screen=yes;party=calling Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 101 INVITE Max-Forwards: 70 Contact: 200 Expires: 240 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 446 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 27362 27362 IN IP4 192.168.254.126 s=- c=IN IP4 192.168.254.126 t=0 0 m=audio 16386 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a localhost*CLI> =fmtp:101 0-15 a=ptime:30 a=sendrecv --- (15 headers 20 lines)--- Sending to 192.168.254.126 : 5060 (no NAT) Using INVITE request as basis request - 1c6107a-5bd372f@192.168.254.126 Reliably Transmitting (no NAT) to 192.168.254.126:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-e870924a;received=192.168.254.126 From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as14d7d04e Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d31d96e" Content-Length: 0 --- Scheduling destruction of SIP dialog '1c6107a-5bd372f@192.168.254.126' in 32000 ms (Method: INVITE) Found user '200' localhost*CLI> <-- SIP read from 192.168.254.126:5060: ACK sip:201@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-e870924a From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as14d7d04e Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 101 ACK Max-Forwards: 70 Contact: 200 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.254.126:5060: INVITE sip:201@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-5af30242 From: 200 ;tag=b12b7fe817b157bo0 To: Remote-Party-ID: 200 ;screen=yes;party=calling Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="6d31d96e",uri="sip:201@192.168.254.96",algorithm=MD5,response="8551495441dc7bc2be3df653cc488cdd" Contact: 200 Expires: 240 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 446 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 27362 27362 IN IP4 192.168.254.126 s=- c=IN IP4 192.168.254.126 t=0 0 m=audio 16386 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (16 headers 20 lines)--- Sending to 192.168.254.126 : 5060 (no NAT) Using INVITE request as basis request - 1c6107a-5bd372f@192.168.254.126 Found user '200' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.126:16386 Found description format PCMU for ID 0 Found description format G726-32 for ID 2 Found description format G723 for ID 4 Found description format PCMA for ID 8 Found description format G729a for ID 18 Found description format G726-40 for ID 96 Found description format G726-24 for ID 97 Found description format G726-16 for ID 98 Found description format NSE for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xd0d (g723|ulaw|alaw|g726|g729|ilbc) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.254.126:16386 Looking for 201 in default (domain 192.168.254.96) list_route: hop: Transmitting (no NAT) to 192.168.254.126:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-5af30242;received=192.168.254.126 From: 200 ;tag=b12b7fe817b157bo0 To: Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- localhost*CLI> -- Executing [201@default:1] Dial("SIP/200-081caf18", "SIP/201") in new stack localhost*CLI> Audio is at 192.168.254.96 port 12714 localhost*CLI> Adding codec 0x4 (ulaw) to SDP localhost*CLI> Adding codec 0x1 (g723) to SDP localhost*CLI> Adding codec 0x2 (gsm) to SDP localhost*CLI> Adding codec 0x8 (alaw) to SDP localhost*CLI> Adding codec 0x10 (g726aal2) to SDP localhost*CLI> Adding codec 0x20 (adpcm) to SDP localhost*CLI> Adding codec 0x40 (slin) to SDP localhost*CLI> Adding codec 0x80 (lpc10) to SDP localhost*CLI> Adding codec 0x100 (g729) to SDP localhost*CLI> Adding codec 0x200 (speex) to SDP localhost*CLI> Adding codec 0x400 (ilbc) to SDP localhost*CLI> Adding codec 0x800 (g726) to SDP localhost*CLI> Adding non-codec 0x1 (telephone-event) to SDP localhost*CLI> Reliably Transmitting (no NAT) to 192.168.254.128:5060: INVITE sip:201@192.168.254.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK772ba11c;rport From: "Reception" ;tag=as7989bca6 To: Contact: Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 07 Aug 2006 11:32:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 563 v=0 o=root 19499 19499 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 12714 RTP/AVP 0 4 3 8 112 5 10 7 18 110 97 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- localhost*CLI> -- Called 201 localhost*CLI> <-- SIP read from 192.168.254.128:5060: SIP/2.0 100 Trying To: From: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK772ba11c Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (8 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.254.128:5060: SIP/2.0 180 Ringing To: ;tag=4deb9e900064d6i0 From: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK772ba11c Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (8 headers 0 lines)--- localhost*CLI> -- SIP/201-081dcb68 is ringing Transmitting (no NAT) to 192.168.254.126:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-5af30242;received=192.168.254.126 From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as31211534 Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- localhost*CLI> <-- SIP read from 192.168.254.128:5060: SIP/2.0 200 OK To: ;tag=4deb9e900064d6i0 From: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK772ba11c Contact: 201 Server: Sipura/SPA2100-3.2.5(d) Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 22877 22877 IN IP4 192.168.254.128 s=- c=IN IP4 192.168.254.128 t=0 0 m=audio 16480 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (12 headers 13 lines)--- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.128:16480 Found description format PCMU for ID 0 Found description format NSE for ID 100 Found description format telephone-event for ID 101 Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.254.128:16480 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.128, port 5060 Transmitting (no NAT) to 192.168.254.128:5060: ACK sip:201@192.168.254.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK04bec874;rport From: "Reception" ;tag=as7989bca6 To: ;tag=4deb9e900064d6i0 Contact: Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> -- SIP/201-081dcb68 answered SIP/200-081caf18 Audio is at 192.168.254.96 port 14818 Adding codec 0x1 (g723) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.254.126:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-5af30242;received=192.168.254.126 From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as31211534 Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 399 v=0 o=root 19499 19499 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 14818 RTP/AVP 4 0 8 18 97 2 101 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- localhost*CLI> <-- SIP read from 192.168.254.126:5060: ACK sip:201@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-69bb295 From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as31211534 Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="6d31d96e",uri="sip:201@192.168.254.96",algorithm=MD5,response="3f253fafaa842146fb557e891150ac28" Contact: 200 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (11 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.254.128:5060: INVITE sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-fc2c7069 From: ;tag=4deb9e900064d6i0 To: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 101 INVITE Max-Forwards: 70 Contact: 201 Expires: 30 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 347 Content-Type: application/sdp v=0 o=- 23816 23816 IN IP4 192.168.254.128 s=- c=IN IP4 192.168.254.128 t=0 0 m=image 16480 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy --- (12 headers 15 lines)--- Sending to 192.168.254.128 : 5060 (no NAT) Got T.38 offer in SDP in dialog 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 Peer doesn't provide audio Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Aug 7 11:32:33] NOTICE[19447]: chan_sip.c:4859 process_sdp: No compatible codecs, not accepting this offer! Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-fc2c7069;received=192.168.254.128 From: ;tag=4deb9e900064d6i0 To: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- localhost*CLI> <-- SIP read from 192.168.254.128:5060: ACK sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-fc2c7069 From: ;tag=4deb9e900064d6i0 To: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 101 ACK Max-Forwards: 70 Contact: 201 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.254.128:5060: INVITE sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-79253233 From: ;tag=4deb9e900064d6i0 To: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 INVITE Max-Forwards: 70 Contact: 201 Expires: 30 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 154 Content-Type: application/sdp v=0 o=- 23817 23817 IN IP4 192.168.254.128 s=- c=IN IP4 192.168.254.128 t=0 0 m=audio 16480 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:30 a=sendrecv --- (12 headers 9 lines)--- Sending to 192.168.254.128 : 5060 (no NAT) Found RTP audio format 0 Peer audio RTP is at port 192.168.254.128:16480 Found description format PCMU for ID 0 Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.254.128:16480 Audio is at 192.168.254.96 port 12714 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-79253233;received=192.168.254.128 From: ;tag=4deb9e900064d6i0 To: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 176 v=0 o=root 19499 19500 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 12714 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=sendrecv --- localhost*CLI> <-- SIP read from 192.168.254.128:5060: ACK sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-a3814af2 From: ;tag=4deb9e900064d6i0 To: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 102 ACK Max-Forwards: 70 Contact: 201 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (10 headers 0 lines)--- localhost*CLI> <-- SIP read from 192.168.254.126:5060: BYE sip:201@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-b82d0e6a From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as31211534 Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="200",realm="asterisk",nonce="6d31d96e",uri="sip:201@192.168.254.96",algorithm=MD5,response="20c7d9cc35cc79d96f3195c0d9cafd02" User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.254.126 : 5060 (no NAT) Transmitting (no NAT) to 192.168.254.126:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.126:5060;branch=z9hG4bK-b82d0e6a;received=192.168.254.126 From: 200 ;tag=b12b7fe817b157bo0 To: ;tag=as31211534 Call-ID: 1c6107a-5bd372f@192.168.254.126 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.128, port 5060 Reliably Transmitting (no NAT) to 192.168.254.128:5060: BYE sip:201@192.168.254.128:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK5ef16c11;rport From: "Reception" ;tag=as7989bca6 To: ;tag=4deb9e900064d6i0 Contact: Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (default, 201, 1) exited non-zero on 'SIP/200-081caf18' localhost*CLI> <-- SIP read from 192.168.254.128:5060: SIP/2.0 200 OK To: ;tag=4deb9e900064d6i0 From: "Reception" ;tag=as7989bca6 Call-ID: 0de893ec1bd5f68e043d433414ec9445@192.168.254.96 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK5ef16c11 Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 --- (8 headers 0 lines)--- Really destroying SIP dialog '0de893ec1bd5f68e043d433414ec9445@192.168.254.96' Method: ACK Really destroying SIP dialog '1c6107a-5bd372f@192.168.254.126' Method: BYE localhost*CLI> quit localhost ~ # exit Script done on Mon Aug 7 11:33:24 2006