Script started on Tue Aug 8 08:54:53 2006 localhost ~ # exitasterisk -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 = 21133) localhost*CLI> Verbosity is at least 4 localhost*CLI> -- Remote UNIX connection localhost*CLI> quitsip debuget verbose 4debug 4 localhost*CLI> Core debug is at least 4 localhost*CLI> set debug 4quitsip debugquitsip debuget verbose 4 localhost*CLI> Verbosity is at least 4 localhost*CLI> set verbose 4debug 4quitsip debug localhost*CLI> SIP Debugging re-enabled localhost*CLI> sip debug localhost*CLI> SIP Debugging re-enabled localhost*CLI> localhost*CLI> 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-11e73c62 From: 201 ;tag=79ba15362ebd6dfbo0 To: Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 101 INVITE Max-Forwards: 70 Contact: 201 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=- 81722 81722 IN IP4 192.168.254.128 s=- c=IN IP4 192.168.254.128 t=0 0 m=audio 16436 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 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: INVITE sip:200@192.168.254.96 SIP/2.0 (37) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-11e73c62 (61) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: 201 ;tag=79ba15362ebd6dfbo0 (57) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: To: (28) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: Call-ID: 98eef612-c584a2cf@192.168.254.128 (42) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: CSeq: 101 INVITE (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Max-Forwards: 70 (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Contact: 201 (43) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: Expires: 240 (12) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: User-Agent: Sipura/SPA2100-3.2.5(d) (35) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: Content-Length: 446 (19) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 11: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 12: Supported: x-sipura (19) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 13: Content-Type: application/sdp (29) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 14: (0) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: v=0 (3) localhost*CLI> [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: o=- 81722 81722 IN IP4 192.168.254.128 (38) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: s=- (3) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: c=IN IP4 192.168.254.128 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: t=0 0 (5) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101 (49) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:18 G729a/8000 (22) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:100 NSE/8000 (21) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=fmtp:100 192-193 (18) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=fmtp:101 0-15 (15) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=ptime:30 (10) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=sendrecv (10) --- (14 headers 20 lines)--- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4106 sip_alloc: Allocating new SIP dialog for 98eef612-c584a2cf@192.168.254.128 - INVITE (With RTP) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:13979 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1620 parse_sip_options: Begin: parsing SIP "Supported: x-sipura" localhost*CLI> [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1628 parse_sip_options: Found SIP option: -x-sipura- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1640 parse_sip_options: Found private SIP option, not supported: x-sipura Sending to 192.168.254.128 : 5060 (no NAT) Using INVITE request as basis request - 98eef612-c584a2cf@192.168.254.128 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:8622 check_user_full: Setting NAT on RTP to Off [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:8632 check_user_full: Setting NAT on UDPTL to Off Reliably Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-11e73c62;received=192.168.254.128 From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as142daf02 Call-ID: 98eef612-c584a2cf@192.168.254.128 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="4213e0c2" Content-Length: 0 --- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1911 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #19 Scheduling destruction of SIP dialog '98eef612-c584a2cf@192.168.254.128' in 32000 ms (Method: INVITE) Found user '201' 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-11e73c62 From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as142daf02 Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 101 ACK Max-Forwards: 70 Contact: 201 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: ACK sip:200@192.168.254.96 SIP/2.0 (34) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-11e73c62 (61) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: 201 ;tag=79ba15362ebd6dfbo0 (57) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: To: ;tag=as142daf02 (43) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: Call-ID: 98eef612-c584a2cf@192.168.254.128 (42) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: CSeq: 101 ACK (13) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Max-Forwards: 70 (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Contact: 201 (43) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: User-Agent: Sipura/SPA2100-3.2.5(d) (35) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: Content-Length: 0 (17) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: (0) --- (10 headers 0 lines)--- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:13979 handle_request: **** Received ACK (6) - Command in SIP ACK [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:2004 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #19 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:2014 __sip_ack: Stopping retransmission on '98eef612-c584a2cf@192.168.254.128' of Response 101: Match Not Found 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-d224afee From: 201 ;tag=79ba15362ebd6dfbo0 To: Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4213e0c2",uri="sip:200@192.168.254.96",algorithm=MD5,response="2909701ccaf140944a722ffdf8e1bde2" Contact: 201 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=- 81722 81722 IN IP4 192.168.254.128 s=- c=IN IP4 192.168.254.128 t=0 0 m=audio 16436 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 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: INVITE sip:200@192.168.254.96 SIP/2.0 (37) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-d224afee (61) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: 201 ;tag=79ba15362ebd6dfbo0 (57) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: To: (28) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: Call-ID: 98eef612-c584a2cf@192.168.254.128 (42) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: CSeq: 102 INVITE (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Max-Forwards: 70 (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4213e0c2",uri="sip:200@192.168.254.96",algorithm=MD5,response="2909701ccaf140944a722ffdf8e1bde2" (163) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: Contact: 201 (43) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: Expires: 240 (12) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: User-Agent: Sipura/SPA2100-3.2.5(d) (35) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 11: Content-Length: 446 (19) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 12: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 13: Supported: x-sipura (19) localhost*CLI> [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 14: Content-Type: application/sdp (29) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 15: (0) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: v=0 (3) localhost*CLI> [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: o=- 81722 81722 IN IP4 192.168.254.128 (38) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: s=- (3) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: c=IN IP4 192.168.254.128 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: t=0 0 (5) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101 (49) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:2 G726-32/8000 (23) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:4 G723/8000 (20) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:18 G729a/8000 (22) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:96 G726-40/8000 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:97 G726-24/8000 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:98 G726-16/8000 (24) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:100 NSE/8000 (21) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=fmtp:100 192-193 (18) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=fmtp:101 0-15 (15) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=ptime:30 (10) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=sendrecv (10) --- (15 headers 20 lines)--- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:13979 handle_request: **** Received INVITE (5) - Command in SIP INVITE [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1620 parse_sip_options: Begin: parsing SIP "Supported: x-sipura" [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1628 parse_sip_options: Found SIP option: -x-sipura- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:1640 parse_sip_options: Found private SIP option, not supported: x-sipura Sending to 192.168.254.128 : 5060 (no NAT) Using INVITE request as basis request - 98eef612-c584a2cf@192.168.254.128 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:8622 check_user_full: Setting NAT on RTP to Off [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:8632 check_user_full: Setting NAT on UDPTL to Off Found user '201' 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 192.168.254.128:16436 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4645 process_sdp: Peer doesn't provide T.38 UDPTL 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 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:ptime in SDP offer [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4831 process_sdp: T38 state changed to 0 on channel 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 192.168.254.128:16436 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4901 process_sdp: We're settling with these formats: 0xd0d (g723|ulaw|alaw|g726|g729|ilbc) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:12792 handle_request_invite: Checking SIP call limits for device 201 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:2879 update_call_counter: Updating call counter for incoming call Looking for 200 in default (domain 192.168.254.96) [Aug 8 08:56:08] DEBUG[21144]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:3608 sip_new: *** Our native formats are 0x4 (ulaw) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:3609 sip_new: *** Joint capabilities are 0xd0d (g723|ulaw|alaw|g726|g729|ilbc) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:3610 sip_new: *** Our capabilities are 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264) [Aug 8 08:56:08] DEBUG[21144]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:3611 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:3634 sip_new: This channel will not be able to handle video. [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:7587 build_route: build_route: Contact hop: 201 list_route: hop: [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:12862 handle_request_invite: SIP/201-081c9700: New call is still down.... Trying... 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-d224afee;received=192.168.254.128 From: 201 ;tag=79ba15362ebd6dfbo0 To: Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Aug 8 08:56:08] DEBUG[21144]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/201-081c9700 [Aug 8 08:56:08] DEBUG[21137]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 201 [Aug 8 08:56:08] DEBUG[21137]: chan_sip.c:14563 sip_devicestate: Checking device state for peer 201 [Aug 8 08:56:08] DEBUG[21137]: devicestate.c:287 do_state_change: Changing state for SIP/201 - state 1 (Not in use) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: pbx.c:1675 pbx_extension_helper: Launching 'Dial' localhost*CLI> -- Executing [200@default:1] Dial("SIP/201-081c9700", "SIP/200") in new stack localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:14619 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4106 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:2510 create_addr_from_peer: Our T38 capability (3840) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:2520 create_addr_from_peer: Setting NAT on RTP to Off localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:2532 create_addr_from_peer: Setting NAT on UDPTL to Off localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:3608 sip_new: *** Our native formats are 0x4 (ulaw) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:3609 sip_new: *** Joint capabilities are 0x0 (nothing) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:3610 sip_new: *** Our capabilities are 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: frame.c:1037 ast_codec_choose: Could not find preferred codec - Going for the best codec localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:3611 sip_new: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:3613 sip_new: *** Our preferred formats from the incoming channel are 0x4 (ulaw) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:3634 sip_new: This channel will not be able to handle video. localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: channel.c:3019 ast_channel_inherit_variables: Not copying variable STACK-default-200-1. localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: channel.c:3019 ast_channel_inherit_variables: Not copying variable SIPCALLID. localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: channel.c:3019 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: channel.c:3019 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: channel.c:3019 ast_channel_inherit_variables: Not copying variable SIPURI. localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:2718 sip_call: Outgoing Call for 200 localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:2879 update_call_counter: Updating call counter for outgoing call localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:2726 sip_call: Our T38 capability (3840), joint T38 capability (3840) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:5806 add_sdp: ** Our capability: 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264) Video flag: False localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:5807 add_sdp: ** Our prefcodec: 0x4 (ulaw) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:5822 add_sdp: This call needs video offers, but there's no video support enabled ! localhost*CLI> Audio is at 192.168.254.96 port 17724 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> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:5959 add_sdp: -- Done with adding codecs to SDP localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:5998 add_sdp: Done building SDP. Settling with this capability: 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 0: INVITE sip:200@192.168.254.126:5060 SIP/2.0 (43) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca;rport (65) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 2: From: "201" ;tag=as7676e651 (51) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 3: To: (34) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 4: Contact: (33) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 5: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 (56) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 6: CSeq: 102 INVITE (16) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 7: User-Agent: Asterisk PBX (24) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 8: Max-Forwards: 70 (16) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 9: Date: Tue, 08 Aug 2006 08:56:08 GMT (35) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 11: Supported: replaces (19) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 12: Content-Type: application/sdp (29) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 13: Content-Length: 563 (19) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4326 parse_request: Header 14: (0) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: v=0 (3) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: o=root 21187 21187 IN IP4 192.168.254.96 (40) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: s=session (9) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: c=IN IP4 192.168.254.96 (23) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: t=0 0 (5) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: m=audio 17724 RTP/AVP 0 4 3 8 112 5 10 7 18 110 97 111 101 (58) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:4 G723/8000 (20) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:3 GSM/8000 (19) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:112 AAL2-G726-32/8000 (30) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:5 DVI4/8000 (20) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:10 L16/8000 (20) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:7 LPC/8000 (19) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:18 G729/8000 (21) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=fmtp:18 annexb=no (19) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:110 speex/8000 (23) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:97 iLBC/8000 (21) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=fmtp:97 mode=20 (17) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:111 G726-32/8000 (25) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=fmtp:101 0-16 (15) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=silenceSupp:off - - - - (25) localhost*CLI> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:4358 parse_request: Line: a=sendrecv (10) localhost*CLI> Reliably Transmitting (no NAT) to 192.168.254.126:5060: INVITE sip:200@192.168.254.126:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca;rport From: "201" ;tag=as7676e651 To: Contact: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 08 Aug 2006 08:56:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 563 v=0 o=root 21187 21187 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 17724 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> [Aug 8 08:56:08] DEBUG[21187]: chan_sip.c:1911 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #21 localhost*CLI> -- Called 200 localhost*CLI> [Aug 8 08:56:08] DEBUG[21188]: app_queue.c:536 changethread: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. localhost*CLI> <-- SIP read from 192.168.254.126:5060: SIP/2.0 100 Trying To: From: "201" ;tag=as7676e651 Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: SIP/2.0 100 Trying (18) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: To: (34) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: "201" ;tag=as7676e651 (51) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 (56) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: CSeq: 102 INVITE (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca (59) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Server: Sipura/SPA2100-3.2.5(d) (31) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Content-Length: 0 (17) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: (0) --- (8 headers 0 lines)--- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:2047 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #21 - INVITE (got response) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:2056 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '49665b1f346167a20a588e9d32eee1f8@192.168.254.96' Request 102: Found [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:11150 handle_response_invite: SIP response 100 to standard invite localhost*CLI> <-- SIP read from 192.168.254.126:5060: SIP/2.0 180 Ringing To: ;tag=8f0f9d44ed87d035i0 From: "201" ;tag=as7676e651 Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca Server: Sipura/SPA2100-3.2.5(d) Remote-Party-ID: 200 ;screen=yes;party=called Content-Length: 0 [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: SIP/2.0 180 Ringing (19) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: To: ;tag=8f0f9d44ed87d035i0 (57) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: "201" ;tag=as7676e651 (51) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 (56) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: CSeq: 102 INVITE (16) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca (59) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Server: Sipura/SPA2100-3.2.5(d) (31) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Remote-Party-ID: 200 ;screen=yes;party=called (69) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: Content-Length: 0 (17) [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: (0) --- (9 headers 0 lines)--- [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:2056 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '49665b1f346167a20a588e9d32eee1f8@192.168.254.96' Request 102: Found localhost*CLI> [Aug 8 08:56:08] DEBUG[21144]: chan_sip.c:11150 handle_response_invite: SIP response 180 to standard invite [Aug 8 08:56:08] DEBUG[21144]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/200-081dd2e8 localhost*CLI> [Aug 8 08:56:08] DEBUG[21137]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 200 [Aug 8 08:56:08] DEBUG[21137]: chan_sip.c:14563 sip_devicestate: Checking device state for peer 200 [Aug 8 08:56:08] DEBUG[21137]: devicestate.c:287 do_state_change: Changing state for SIP/200 - state 1 (Not in use) localhost*CLI> -- SIP/200-081dd2e8 is ringing localhost*CLI> Transmitting (no NAT) to 192.168.254.128:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-d224afee;received=192.168.254.128 From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as042f405f Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- localhost*CLI> [Aug 8 08:56:08] DEBUG[21189]: app_queue.c:536 changethread: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. localhost*CLI> <-- SIP read from 192.168.254.126:5060: SIP/2.0 200 OK To: ;tag=8f0f9d44ed87d035i0 From: "201" ;tag=as7676e651 Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca Contact: 200 Server: Sipura/SPA2100-3.2.5(d) Remote-Party-ID: 200 ;screen=yes;party=called Content-Length: 257 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 80426 80426 IN IP4 192.168.254.126 s=- c=IN IP4 192.168.254.126 t=0 0 m=audio 16436 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 [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: SIP/2.0 200 OK (14) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: To: ;tag=8f0f9d44ed87d035i0 (57) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: "201" ;tag=as7676e651 (51) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 (56) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: CSeq: 102 INVITE (16) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK067c50ca (59) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Contact: 200 (43) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Server: Sipura/SPA2100-3.2.5(d) (31) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: Remote-Party-ID: 200 ;screen=yes;party=called (69) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: Content-Length: 257 (19) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 11: Supported: x-sipura (19) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 12: Content-Type: application/sdp (29) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 13: (0) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: v=0 (3) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: o=- 80426 80426 IN IP4 192.168.254.126 (38) localhost*CLI> [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: s=- (3) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: c=IN IP4 192.168.254.126 (24) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: t=0 0 (5) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: m=audio 16436 RTP/AVP 0 100 101 (31) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:100 NSE/8000 (21) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=fmtp:100 192-193 (18) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=fmtp:101 0-15 (15) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=ptime:30 (10) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4358 parse_request: Line: a=sendrecv (10) --- (13 headers 13 lines)--- [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:1996 __sip_ack: Acked pending invite 102 [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:2014 __sip_ack: Stopping retransmission on '49665b1f346167a20a588e9d32eee1f8@192.168.254.96' of Request 102: Match Not Found [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:11150 handle_response_invite: SIP response 200 to standard invite Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.126:16436 [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4645 process_sdp: Peer doesn't provide T.38 UDPTL Found description format PCMU for ID 0 Found description format NSE for ID 100 Got unsupported a:fmtp in SDP offer Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Got unsupported a:ptime in SDP offer [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4831 process_sdp: T38 state changed to 0 on channel SIP/200-081dd2e8 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 192.168.254.126:16436 [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4901 process_sdp: We're settling with these formats: 0x4 (ulaw) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4908 process_sdp: We have an owner, now see if we need to change this call localhost*CLI> [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:2879 update_call_counter: Updating call counter for outgoing call [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:7587 build_route: build_route: Contact hop: 200 list_route: hop: [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:5337 reqprep: Strict routing enforced for session 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.126, port 5060 Transmitting (no NAT) to 192.168.254.126:5060: ACK sip:200@192.168.254.126:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK0be15f98;rport From: "201" ;tag=as7676e651 To: ;tag=8f0f9d44ed87d035i0 Contact: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Aug 8 08:56:13] DEBUG[21187]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/200-081dd2e8 -- SIP/200-081dd2e8 answered SIP/201-081c9700 [Aug 8 08:56:13] DEBUG[21187]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/201-081c9700 [Aug 8 08:56:13] DEBUG[21187]: chan_sip.c:3319 sip_answer: SIP answering channel: SIP/201-081c9700 [Aug 8 08:56:13] DEBUG[21187]: chan_sip.c:5806 add_sdp: ** Our capability: 0xd0d (g723|ulaw|alaw|g726|g729|ilbc) Video flag: True [Aug 8 08:56:13] DEBUG[21187]: chan_sip.c:5807 add_sdp: ** Our prefcodec: 0x0 (nothing) Audio is at 192.168.254.96 port 10856 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 [Aug 8 08:56:13] DEBUG[21187]: chan_sip.c:5959 add_sdp: -- Done with adding codecs to SDP [Aug 8 08:56:13] DEBUG[21187]: chan_sip.c:5998 add_sdp: Done building SDP. Settling with this capability: 0xd0d (g723|ulaw|alaw|g726|g729|ilbc) 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-d224afee;received=192.168.254.128 From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as042f405f Call-ID: 98eef612-c584a2cf@192.168.254.128 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 21187 21187 IN IP4 192.168.254.96 s=session c=IN IP4 192.168.254.96 t=0 0 m=audio 10856 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 --- [Aug 8 08:56:13] DEBUG[21187]: chan_sip.c:1911 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #23 [Aug 8 08:56:13] DEBUG[21137]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 200 [Aug 8 08:56:13] DEBUG[21137]: chan_sip.c:14563 sip_devicestate: Checking device state for peer 200 [Aug 8 08:56:13] DEBUG[21137]: devicestate.c:287 do_state_change: Changing state for SIP/200 - state 1 (Not in use) localhost*CLI> [Aug 8 08:56:13] DEBUG[21137]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 201 [Aug 8 08:56:13] DEBUG[21137]: chan_sip.c:14563 sip_devicestate: Checking device state for peer 201 [Aug 8 08:56:13] DEBUG[21137]: devicestate.c:287 do_state_change: Changing state for SIP/201 - state 1 (Not in use) localhost*CLI> [Aug 8 08:56:13] DEBUG[21190]: app_queue.c:536 changethread: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. localhost*CLI> [Aug 8 08:56:13] DEBUG[21191]: app_queue.c:536 changethread: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. 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-7e18d4a4 From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as042f405f Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4213e0c2",uri="sip:200@192.168.254.96",algorithm=MD5,response="edf4c882ffec251408147485a3bed470" Contact: 201 User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: ACK sip:200@192.168.254.96 SIP/2.0 (34) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-7e18d4a4 (61) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: 201 ;tag=79ba15362ebd6dfbo0 (57) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: To: ;tag=as042f405f (43) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: Call-ID: 98eef612-c584a2cf@192.168.254.128 (42) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: CSeq: 102 ACK (13) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Max-Forwards: 70 (16) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4213e0c2",uri="sip:200@192.168.254.96",algorithm=MD5,response="edf4c882ffec251408147485a3bed470" (163) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: Contact: 201 (43) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: User-Agent: Sipura/SPA2100-3.2.5(d) (35) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: Content-Length: 0 (17) [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 11: (0) --- (11 headers 0 lines)--- [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:13979 handle_request: **** Received ACK (6) - Command in SIP ACK [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:2004 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #23 [Aug 8 08:56:13] DEBUG[21144]: chan_sip.c:2014 __sip_ack: Stopping retransmission on '98eef612-c584a2cf@192.168.254.128' of Response 102: Match Not Found localhost*CLI> [Aug 8 08:56:13] DEBUG[21187]: rtp.c:2343 ast_rtp_write: Ooh, format changed from unknown to ulaw localhost*CLI> [Aug 8 08:56:13] DEBUG[21187]: rtp.c:2343 ast_rtp_write: Ooh, format changed from unknown to ulaw localhost*CLI> <-- SIP read from 192.168.254.128:5060: BYE sip:200@192.168.254.96 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-60110f9b From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as042f405f Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 103 BYE Max-Forwards: 70 Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4213e0c2",uri="sip:200@192.168.254.96",algorithm=MD5,response="6d524e8e000b444efd4c0dd9ec85c585" User-Agent: Sipura/SPA2100-3.2.5(d) Content-Length: 0 [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: BYE sip:200@192.168.254.96 SIP/2.0 (34) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-60110f9b (61) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: 201 ;tag=79ba15362ebd6dfbo0 (57) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: To: ;tag=as042f405f (43) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: Call-ID: 98eef612-c584a2cf@192.168.254.128 (42) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: CSeq: 103 BYE (13) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Max-Forwards: 70 (16) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Proxy-Authorization: Digest username="201",realm="asterisk",nonce="4213e0c2",uri="sip:200@192.168.254.96",algorithm=MD5,response="6d524e8e000b444efd4c0dd9ec85c585" (163) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: User-Agent: Sipura/SPA2100-3.2.5(d) (35) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: Content-Length: 0 (17) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: (0) --- (10 headers 0 lines)--- [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:13979 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.254.128 : 5060 (no NAT) 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-60110f9b;received=192.168.254.128 From: 201 ;tag=79ba15362ebd6dfbo0 To: ;tag=as042f405f Call-ID: 98eef612-c584a2cf@192.168.254.128 CSeq: 103 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- [Aug 8 08:56:34] DEBUG[21187]: channel.c:3490 ast_generic_bridge: Didn't get a frame from channel: SIP/201-081c9700 [Aug 8 08:56:34] DEBUG[21187]: channel.c:3784 ast_channel_bridge: Bridge stops bridging channels SIP/201-081c9700 and SIP/200-081dd2e8 localhost*CLI> [Aug 8 08:56:34] DEBUG[21187]: channel.c:1527 ast_hangup: Hanging up channel 'SIP/200-081dd2e8' [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:3180 sip_hangup: Hangup call SIP/200-081dd2e8, SIP callid 49665b1f346167a20a588e9d32eee1f8@192.168.254.96) [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:3188 sip_hangup: update_call_counter(200) - decrement call limit counter on hangup [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:2879 update_call_counter: Updating call counter for outgoing call [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:5337 reqprep: Strict routing enforced for session 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.126, port 5060 Reliably Transmitting (no NAT) to 192.168.254.126:5060: BYE sip:200@192.168.254.126:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK6b64b2f9;rport From: "201" ;tag=as7676e651 To: ;tag=8f0f9d44ed87d035i0 Contact: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:1911 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #26 [Aug 8 08:56:34] DEBUG[21187]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/200-081dd2e8 [Aug 8 08:56:34] DEBUG[21187]: rtp.c:1317 ast_rtp_early_bridge: Channel '' has no RTP, not doing anything [Aug 8 08:56:34] DEBUG[21187]: app_dial.c:1624 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Aug 8 08:56:34] DEBUG[21187]: pbx.c:2271 __ast_pbx_run: Spawn extension (default,200,1) exited non-zero on 'SIP/201-081c9700' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '"201" <201>' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '201' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '200' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'default' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/201-081c9700' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/200-081dd2e8' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'Dial' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'SIP/200' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-08 08:56:08' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-08 08:56:13' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '2006-08-08 08:56:34' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '26' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '21' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '201' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '1155027368.0' [Aug 8 08:56:34] DEBUG[21187]: pbx.c:1529 pbx_substitute_variables_helper_full: Function result is '' [Aug 8 08:56:34] DEBUG[21187]: channel.c:1527 ast_hangup: Hanging up channel 'SIP/201-081c9700' [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:3180 sip_hangup: Hangup call SIP/201-081c9700, SIP callid 98eef612-c584a2cf@192.168.254.128) [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:3188 sip_hangup: update_call_counter(201) - decrement call limit counter on hangup [Aug 8 08:56:34] DEBUG[21187]: chan_sip.c:2879 update_call_counter: Updating call counter for incoming call [Aug 8 08:56:34] DEBUG[21187]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/201-081c9700 [Aug 8 08:56:34] DEBUG[21137]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 200 [Aug 8 08:56:34] DEBUG[21137]: chan_sip.c:14563 sip_devicestate: Checking device state for peer 200 [Aug 8 08:56:34] DEBUG[21137]: devicestate.c:287 do_state_change: Changing state for SIP/200 - state 1 (Not in use) localhost*CLI> [Aug 8 08:56:34] DEBUG[21137]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 201 [Aug 8 08:56:34] DEBUG[21137]: chan_sip.c:14563 sip_devicestate: Checking device state for peer 201 [Aug 8 08:56:34] DEBUG[21137]: devicestate.c:287 do_state_change: Changing state for SIP/201 - state 1 (Not in use) localhost*CLI> [Aug 8 08:56:34] DEBUG[21192]: app_queue.c:536 changethread: Device 'SIP/200' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. localhost*CLI> [Aug 8 08:56:34] DEBUG[21193]: app_queue.c:536 changethread: Device 'SIP/201' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. localhost*CLI> <-- SIP read from 192.168.254.126:5060: SIP/2.0 200 OK To: ;tag=8f0f9d44ed87d035i0 From: "201" ;tag=as7676e651 Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 CSeq: 103 BYE Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK6b64b2f9 Server: Sipura/SPA2100-3.2.5(d) Content-Length: 0 [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: SIP/2.0 200 OK (14) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: To: ;tag=8f0f9d44ed87d035i0 (57) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: "201" ;tag=as7676e651 (51) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: Call-ID: 49665b1f346167a20a588e9d32eee1f8@192.168.254.96 (56) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: CSeq: 103 BYE (13) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: Via: SIP/2.0/UDP 192.168.254.96:5060;branch=z9hG4bK6b64b2f9 (59) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Server: Sipura/SPA2100-3.2.5(d) (31) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Content-Length: 0 (17) [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: (0) --- (8 headers 0 lines)--- [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:2004 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #26 [Aug 8 08:56:34] DEBUG[21144]: chan_sip.c:2014 __sip_ack: Stopping retransmission on '49665b1f346167a20a588e9d32eee1f8@192.168.254.96' of Request 103: Match Not Found Really destroying SIP dialog '49665b1f346167a20a588e9d32eee1f8@192.168.254.96' Method: INVITE Really destroying SIP dialog '98eef612-c584a2cf@192.168.254.128' Method: BYE localhost*CLI> quit <-- 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-3e14976 From: 201 ;tag=bca95f553c4cd52bo0 To: 201 Call-ID: f561d3bc-55d912d7@192.168.254.12 CSeq: 59011 REGISTER Max-Forwards: 70 Authorization: Digest username="201",realm="asterisk",nonce="777dedd1",uri="sip:192.168.254.96",algorithm=MD5,response="6f5fed50a75c95a325d1287d42fb7592" 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 [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 0: REGISTER sip:192.168.254.96 SIP/2.0 (35) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.254.128:5060;branch=z9hG4bK-3e14976 (60) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 2: From: 201 ;tag=bca95f553c4cd52bo0 (57) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 3: To: 201 (32) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 4: Call-ID: f561d3bc-55d912d7@192.168.254.12 (41) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 5: CSeq: 59011 REGISTER (20) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 6: Max-Forwards: 70 (16) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 7: Authorization: Digest username="201",realm="asterisk",nonce="777dedd1",uri="sip:192.168.254.96",algorithm=MD5,response="6f5fed50a75c95a325d1287d42fb7592" (153) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 8: Contact: 201 ;expires=3600 (56) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 9: User-Agent: Sipura/SPA2100-3.2.5(d) (35) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 10: Content-Length: 0 (17) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 11: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 12: Supported: x-sipura (19) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4326 parse_request: Header 13: (0) --- (13 headers 0 lines)--- [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:4106 sip_alloc: Allocating new SIP dialog for f561d3bc-55d912d7@192.168.254.12 - REGISTER (No RTP) [Aug 8 08:57:21] DEBUG[21144]: chan_sip.c:13979 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.254.128 : 5060 (no NAT) localhost*CLI> quit 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-3e14976;received=192.168.254.128 From: 201 ;tag=bca95f553c4cd52bo0 To: 201 Call-ID: f561d3bc-55d912d7@192.168.254.12 CSeq: 59011 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-3e14976;received=192.168.254.128 From: 201 ;tag=bca95f553c4cd52bo0 To: 201 ;tag=as72276efa Call-ID: f561d3bc-55d912d7@192.168.254.12 CSeq: 59011 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="0163874d" Content-Length: 0 --- Scheduling destruction of SIP dialog 'f561d3bc-55d912d7@192.168.254.12' in 32000 ms (Method: REGISTER) localhost*CLI> quit localhost ~ # exit exit Script done on Tue Aug 8 08:57:23 2006