=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.07.19 16:31:23 =~=~=~=~=~=~=~=~=~=~=~= <-- SIP read from 192.168.2.201:5060: PUBLISH sip:204@172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-t54tv0lxk6ap;rport From: ;tag=pdyvb1wqe5 To: Call-ID: 3c27f3f72e63-lt5li5s1onh9@snom320-00041324225D CSeq: 1 PUBLISH Max-Forwards: 70 Event: proxy-config Content-Type: application/text Content-Length: 0 Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: PUBLISH sip:204@172.16.0.14 SIP/2.0 (35) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-t54tv0lxk6ap;rport (69) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=pdyvb1wqe5 (42) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (25) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f72e63-lt5li5s1onh9@snom320-00041324225D (55) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 1 PUBLISH (15) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Event: proxy-config (19) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Type: application/text (30) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Content-Length: 0 (17) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 3c27f3f72e63-lt5li5s1onh9@snom320-00041324225D - PUBLISH (No RTP) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received PUBLISH (15) - Command in SIP PUBLISH Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-t54tv0lxk6ap;rport;received=192.168.2.201 From: ;tag=pdyvb1wqe5 To: ;tag=as530f425a Call-ID: 3c27f3f72e63-lt5li5s1onh9@snom320-00041324225D CSeq: 1 PUBLISH User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 --- Jul 19 16:31:26 NOTICE[4196]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '192.168.2.201' Destroying call '3c27f3f72e63-lt5li5s1onh9@snom320-00041324225D' efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: PUBLISH sip:204@172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-3iloz6e9fqmu;rport From: ;tag=ijrx9l1yzd To: Call-ID: 3c27f3f7a875-bsv3d3qucjfe@snom320-00041324225D CSeq: 1 PUBLISH Max-Forwards: 70 Event: number-guessing Content-Type: application/text Content-Length: 25 Number: 30 Max-Hits: 3 Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: PUBLISH sip:204@172.16.0.14 SIP/2.0 (35) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-3iloz6e9fqmu;rport (69) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=ijrx9l1yzd (42) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (25) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f7a875-bsv3d3qucjfe@snom320-00041324225D (55) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 1 PUBLISH (15) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Event: number-guessing (22) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Type: application/text (30) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Content-Length: 25 (18) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: (0) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Number: 30 (10) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Max-Hits: 3 (11) --- (10 headers 2 lines)--- Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 3c27f3f7a875-bsv3d3qucjfe@snom320-00041324225D - PUBLISH (No RTP) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received PUBLISH (15) - Command in SIP PUBLISH Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-3iloz6e9fqmu;rport;received=192.168.2.201 From: ;tag=ijrx9l1yzd To: ;tag=as15fb98ae Call-ID: 3c27f3f7a875-bsv3d3qucjfe@snom320-00041324225D CSeq: 1 PUBLISH User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 --- efw-voiceone-2*CLI> Jul 19 16:31:26 NOTICE[4196]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '192.168.2.201' Destroying call '3c27f3f7a875-bsv3d3qucjfe@snom320-00041324225D' efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: PUBLISH sip:204@172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-ebwtkbjq6wu4;rport From: ;tag=q1y7fd0q7y To: Call-ID: 3c27f3f7cf85-tjy4ya7dpeso@snom320-00041324225D CSeq: 1 PUBLISH Max-Forwards: 70 Event: proxy-config Content-Type: application/text Content-Length: 0 Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: PUBLISH sip:204@172.16.0.14 SIP/2.0 (35) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-ebwtkbjq6wu4;rport (69) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=q1y7fd0q7y (42) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (25) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f7cf85-tjy4ya7dpeso@snom320-00041324225D (55) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 1 PUBLISH (15) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Event: proxy-config (19) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Type: application/text (30) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Content-Length: 0 (17) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: (0) --- (10 headers 0 lines)--- Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 3c27f3f7cf85-tjy4ya7dpeso@snom320-00041324225D - PUBLISH (No RTP) Jul 19 16:31:26 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received PUBLISH (15) - Command in SIP PUBLISH Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-ebwtkbjq6wu4;rport;received=192.168.2.201 From: ;tag=q1y7fd0q7y To: ;tag=as721f902e Call-ID: 3c27f3f7cf85-tjy4ya7dpeso@snom320-00041324225D CSeq: 1 PUBLISH User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 --- Jul 19 16:31:26 NOTICE[4196]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '192.168.2.201' Destroying call '3c27f3f7cf85-tjy4ya7dpeso@snom320-00041324225D' efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: PUBLISH sip:204@172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-t2lccbs6ztyx;rport From: ;tag=7rjnh6l41w To: Call-ID: 3c27f3f85573-0sy8pn6iwor2@snom320-00041324225D CSeq: 1 PUBLISH Max-Forwards: 70 Event: number-guessing Content-Type: application/text Content-Length: 26 Number: 304 Max-Hits: 3 Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: PUBLISH sip:204@172.16.0.14 SIP/2.0 (35) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-t2lccbs6ztyx;rport (69) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=7rjnh6l41w (42) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (25) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f85573-0sy8pn6iwor2@snom320-00041324225D (55) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 1 PUBLISH (15) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Event: number-guessing (22) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Type: application/text (30) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Content-Length: 26 (18) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: (0) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Number: 304 (11) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Max-Hits: 3 (11) --- (10 headers 2 lines)--- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 3c27f3f85573-0sy8pn6iwor2@snom320-00041324225D - PUBLISH (No RTP) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received PUBLISH (15) - Command in SIP PUBLISH Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-t2lccbs6ztyx;rport;received=192.168.2.201 From: ;tag=7rjnh6l41w To: ;tag=as5daa662b Call-ID: 3c27f3f85573-0sy8pn6iwor2@snom320-00041324225D CSeq: 1 PUBLISH User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Accept: application/sdp Content-Length: 0 --- Jul 19 16:31:27 NOTICE[4196]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '192.168.2.201' efw-voiceone-2*CLI> Destroying call '3c27f3f85573-0sy8pn6iwor2@snom320-00041324225D' efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: INVITE sip:304@172.16.0.14;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-c2nwhu4z0jeg;rport From: ;tag=nwdhzhgetw To: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 453 v=0 o=root 1142710222 1142710222 IN IP4 192.168.2.201 s=call c=IN IP4 192.168.2.201 t=0 0 m=audio 65508 RTP/AVP 3 18 8 2 4 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X5UxhUHi/Pu578s0XHo0zjyswieX+a5MebNV6ZxL a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: INVITE sip:304@172.16.0.14;user=phone SIP/2.0 (45) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-c2nwhu4z0jeg;rport (69) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=nwdhzhgetw (42) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (36) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D (55) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 1 INVITE (14) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1 (61) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: P-Key-Flags: keys="3" (21) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: User-Agent: snom320/6.2.2 (25) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: Accept: application/sdp (23) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 12: Allow-Events: talk, hold, refer (31) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 15: Min-SE: 90 (10) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 16: Content-Type: application/sdp (29) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 17: Content-Length: 453 (19) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 18: (0) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: v=0 (3) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: o=root 1142710222 1142710222 IN IP4 192.168.2.201 (49) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: s=call (6) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: c=IN IP4 192.168.2.201 (22) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: t=0 0 (5) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: m=audio 65508 RTP/AVP 3 18 8 2 4 0 101 (38) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X5UxhUHi/Pu578s0XHo0zjyswieX+a5MebNV6ZxL (82) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:3 gsm/8000 (19) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:18 g729/8000 (21) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:8 pcma/8000 (20) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:2 g726-32/8000 (23) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:4 g723/8000 (20) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=fmtp:101 0-16 (15) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=ptime:20 (10) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=encryption:optional (21) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=sendrecv (10) --- (18 headers 18 lines)--- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 3c27f3f6b98c-6df77r06v810@snom320-00041324225D - INVITE (With RTP) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received INVITE (5) - Command in SIP INVITE Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1004 parse_sip_options: Begin: parsing SIP "Supported: timer, 100rel, replaces, callerid" Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1016 parse_sip_options: Found SIP option: -timer- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1022 parse_sip_options: Matched SIP option: timer Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1016 parse_sip_options: Found SIP option: -100rel- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1022 parse_sip_options: Matched SIP option: 100rel Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1016 parse_sip_options: Found SIP option: -replaces- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1022 parse_sip_options: Matched SIP option: replaces Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1016 parse_sip_options: Found SIP option: -callerid- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1027 parse_sip_options: Found no match for SIP option: callerid (Please file bug report!) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1033 parse_sip_options: * SIP extension value: 7 for call 3c27f3f6b98c-6df77r06v810@snom320-00041324225D Using INVITE request as basis request - 3c27f3f6b98c-6df77r06v810@snom320-00041324225D Sending to 192.168.2.201 : 5060 (NAT) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:7149 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-c2nwhu4z0jeg;rport;received=192.168.2.201 From: ;tag=nwdhzhgetw To: ;tag=as52f74d5a Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="341a3ef5" Content-Length: 0 --- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #164 Scheduling destruction of call '3c27f3f6b98c-6df77r06v810@snom320-00041324225D' in 15000 ms Found user '204' efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: ACK sip:304@172.16.0.14;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-c2nwhu4z0jeg;rport From: ;tag=nwdhzhgetw To: ;tag=as52f74d5a Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: ACK sip:304@172.16.0.14;user=phone SIP/2.0 (42) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-c2nwhu4z0jeg;rport (69) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=nwdhzhgetw (42) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: ;tag=as52f74d5a (51) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D (55) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 1 ACK (11) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1 (61) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Length: 0 (17) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received ACK (6) - Command in SIP ACK Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #164 Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c27f3f6b98c-6df77r06v810@snom320-00041324225D' of Response 1: Match Found <-- SIP read from 192.168.2.201:5060: INVITE sip:304@172.16.0.14;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-lu0siza2a6ff;rport From: ;tag=nwdhzhgetw To: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/6.2.2 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="204",realm="asterisk",nonce="341a3ef5",uri="sip:304@172.16.0.14;user=phone",response="a0178e61cdbaba2119726b4ad62ec715",algorithm=md5 Content-Type: application/sdp Content-Length: 453 v=0 o=root 1142710222 1142710222 IN IP4 192.168.2.201 s=call c=IN IP4 192.168.2.201 t=0 0 m=audio 65508 RTP/AVP 3 18 8 2 4 0 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X5UxhUHi/Pu578s0XHo0zjyswieX+a5MebNV6ZxL a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: INVITE sip:304@172.16.0.14;user=phone SIP/2.0 (45) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-lu0siza2a6ff;rport (69) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=nwdhzhgetw (42) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (36) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D (55) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 2 INVITE (14) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1 (61) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: P-Key-Flags: keys="3" (21) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: User-Agent: snom320/6.2.2 (25) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: Accept: application/sdp (23) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 12: Allow-Events: talk, hold, refer (31) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 13: Supported: timer, 100rel, replaces, callerid (44) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 14: Session-Expires: 3600;refresher=uas (35) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 15: Min-SE: 90 (10) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 16: Proxy-Authorization: Digest username="204",realm="asterisk",nonce="341a3ef5",uri="sip:304@172.16.0.14;user=phone",response="a0178e61cdbaba2119726b4ad62ec715",algorithm=md5 (171) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 17: Content-Type: application/sdp (29) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 18: Content-Length: 453 (19) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 19: (0) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: v=0 (3) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: o=root 1142710222 1142710222 IN IP4 192.168.2.201 (49) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: s=call (6) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: c=IN IP4 192.168.2.201 (22) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: t=0 0 (5) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: m=audio 65508 RTP/AVP 3 18 8 2 4 0 101 (38) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:X5UxhUHi/Pu578s0XHo0zjyswieX+a5MebNV6ZxL (82) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:3 gsm/8000 (19) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:18 g729/8000 (21) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:8 pcma/8000 (20) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:2 g726-32/8000 (23) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:4 g723/8000 (20) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:0 pcmu/8000 (20) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=fmtp:101 0-16 (15) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=ptime:20 (10) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=encryption:optional (21) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: a=sendrecv (10) --- (19 headers 18 lines)--- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received INVITE (5) - Command in SIP INVITE Using INVITE request as basis request - 3c27f3f6b98c-6df77r06v810@snom320-00041324225D Sending to 192.168.2.201 : 5060 (NAT) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:7149 check_user_full: Setting NAT on RTP to 0 Found user '204' Found RTP audio format 3 efw-voiceone-2*CLI> Found RTP audio format 18 efw-voiceone-2*CLI> Found RTP audio format 8 efw-voiceone-2*CLI> Found RTP audio format 2 efw-voiceone-2*CLI> Found RTP audio format 4 efw-voiceone-2*CLI> Found RTP audio format 0 efw-voiceone-2*CLI> Found RTP audio format 101 efw-voiceone-2*CLI> Peer audio RTP is at port 192.168.2.201:65508 efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3598 process_sdp: Peer audio RTP is at port 192.168.2.201:65508 efw-voiceone-2*CLI> Found description format gsm efw-voiceone-2*CLI> Found description format g729 Found description format pcma Found description format g726-32 Found description format g723 Found description format pcmu Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:10496 handle_request_invite: Checking SIP call limits for device 204 Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:2206 update_call_counter: Updating call counter for incoming call Looking for 304 in default (domain 172.16.0.14) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:6131 build_route: build_route: Contact hop: ;flow-id=1 list_route: hop: efw-voiceone-2*CLI> Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-lu0siza2a6ff;rport;received=192.168.2.201 From: ;tag=nwdhzhgetw To: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4189]: chan_sip.c:11670 sip_devicestate: Checking device state for peer 204 efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4189]: devicestate.c:187 do_state_change: Changing state for SIP/204 - state 2 (In use) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'Macro' efw-voiceone-2*CLI> -- Executing Macro("SIP/204-08e9d488", "stdexten|SIP/304") in new stack efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '304' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'Set' efw-voiceone-2*CLI> -- Executing Set("SIP/204-08e9d488", "EXTENSION=304") in new stack efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: db.c:200 ast_db_get: Unable to find key '304' in family 'DND' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: func_db.c:69 function_db_read: DB: DND/304 not found in database. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '0' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' efw-voiceone-2*CLI> -- Executing GotoIf("SIP/204-08e9d488", "0?s-BUSY|1") in new stack efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: db.c:200 ast_db_get: Unable to find key '304' in family 'CFU' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: func_db.c:69 function_db_read: DB: CFU/304 not found in database. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '0' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: db.c:200 ast_db_get: Unable to find key 'SIP/304' in family 'CFU' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: func_db.c:69 function_db_read: DB: CFU/SIP/304 not found in database. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '' efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' efw-voiceone-2*CLI> -- Executing GotoIf("SIP/204-08e9d488", "0?default||1") in new stack efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'Dial' efw-voiceone-2*CLI> -- Executing Dial("SIP/204-08e9d488", "SIP/304|20|gtTwW") in new stack efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:1874 create_addr_from_peer: Setting NAT on RTP to 0 efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable STACK-macro-stdexten-s-4. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable MACRO_DEPTH. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable STACK-macro-stdexten-s-3. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable STACK-macro-stdexten-s-2. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable EXTENSION. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable STACK-macro-stdexten-s-1. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable ARG1. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable MACRO_PRIORITY. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable MACRO_CONTEXT. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable MACRO_EXTEN. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable STACK-default-304-1. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable SIPCALLID. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable SIPUSERAGENT. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2856 ast_channel_inherit_variables: Not copying variable SIPURI. efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:2068 sip_call: Outgoing Call for 304 efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:2206 update_call_counter: Updating call counter for outgoing call efw-voiceone-2*CLI> We're at 172.16.0.14 port 16456 efw-voiceone-2*CLI> Adding codec 0x8 (alaw) to SDP efw-voiceone-2*CLI> Adding codec 0x2 (gsm) to SDP efw-voiceone-2*CLI> Adding codec 0x4 (ulaw) to SDP efw-voiceone-2*CLI> Adding non-codec 0x1 (telephone-event) to SDP efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 0: INVITE sip:304@192.168.2.101:40952;rinstance=8b1c93b54f6a0077 SIP/2.0 (69) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport (62) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 2: From: "204" ;tag=as49c9f038 (48) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 3: To: (60) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 4: Contact: (30) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 5: Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 (53) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 6: CSeq: 102 INVITE (16) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 7: User-Agent: Asterisk PBX (24) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 8: Max-Forwards: 70 (16) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 9: Date: Wed, 19 Jul 2006 14:31:27 GMT (35) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 11: Content-Type: application/sdp (29) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 12: Content-Length: 259 (19) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3357 parse_request: Header 13: (0) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: v=0 (3) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: o=root 4150 4150 IN IP4 172.16.0.14 (35) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: s=session (9) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: c=IN IP4 172.16.0.14 (20) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: t=0 0 (5) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: m=audio 16456 RTP/AVP 8 3 0 101 (31) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: a=rtpmap:3 GSM/8000 (19) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: a=rtpmap:0 PCMU/8000 (20) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: a=fmtp:101 0-16 (15) efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:3389 parse_request: Line: a=silenceSupp:off - - - - (25) efw-voiceone-2*CLI> 13 headers, 12 lines efw-voiceone-2*CLI> Reliably Transmitting (no NAT) to 192.168.2.101:40952: INVITE sip:304@192.168.2.101:40952;rinstance=8b1c93b54f6a0077 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport From: "204" ;tag=as49c9f038 To: Contact: Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 19 Jul 2006 14:31:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 4150 4150 IN IP4 172.16.0.14 s=session c=IN IP4 172.16.0.14 t=0 0 m=audio 16456 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #166 efw-voiceone-2*CLI> -- Called 304 efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/304-08ea1360 to read format alaw efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/204-08e9d488 to write format alaw efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/204-08e9d488 to read format alaw efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/304-08ea1360 to write format alaw efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4442]: app_queue.c:490 changethread: Device 'SIP/204' changed to state '2' (In use) but we don't care because they're not a member of any queue. efw-voiceone-2*CLI> <-- SIP read from 192.168.2.101:40952: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport=5060 Contact: To: ;tag=45043174 From: "204";tag=as49c9f038 Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 CSeq: 102 INVITE User-Agent: eyeBeam release 1003l stamp 30936 Content-Length: 0 Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: SIP/2.0 180 Ringing (19) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport=5060 (67) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: Contact: (65) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: ;tag=45043174 (73) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: From: "204";tag=as49c9f038 (47) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 (53) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: CSeq: 102 INVITE (16) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: User-Agent: eyeBeam release 1003l stamp 30936 (45) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Length: 0 (17) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1445 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #166 - INVITE (got response) Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:1454 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14' Request 102: Found Jul 19 16:31:27 DEBUG[4196]: chan_sip.c:9578 handle_response_invite: SIP response 180 to standard invite efw-voiceone-2*CLI> -- SIP/304-08ea1360 is ringing Jul 19 16:31:27 DEBUG[4189]: chan_sip.c:11670 sip_devicestate: Checking device state for peer 304 Jul 19 16:31:27 DEBUG[4189]: devicestate.c:187 do_state_change: Changing state for SIP/304 - state 6 (Ringing) efw-voiceone-2*CLI> Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-lu0siza2a6ff;rport;received=192.168.2.201 From: ;tag=nwdhzhgetw To: ;tag=as00574def Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- efw-voiceone-2*CLI> Jul 19 16:31:27 DEBUG[4443]: app_queue.c:490 changethread: Device 'SIP/304' changed to state '6' (Ringing) but we don't care because they're not a member of any queue. efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: REGISTER sip:172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-66xpqgn7lhre;rport From: ;tag=idnc61o4o0 To: Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D CSeq: 6574 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/6.2.2 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.2.201 WWW-Contact: WWW-Contact: Expires: 60 Content-Length: 0 Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: REGISTER sip:172.16.0.14 SIP/2.0 (32) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-66xpqgn7lhre;rport (69) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=idnc61o4o0 (42) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (25) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D (55) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 6574 REGISTER (19) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" (306) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: User-Agent: snom320/6.2.2 (25) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Supported: gruu (15) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: Allow-Events: dialog (20) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 11: X-Real-IP: 192.168.2.201 (24) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 12: WWW-Contact: (38) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 13: WWW-Contact: (40) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 14: Expires: 60 (11) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 15: Content-Length: 0 (17) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 16: (0) --- (16 headers 0 lines)--- Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for 3c27a27a8647-qqwympeiaqne@snom320-00041324225D - REGISTER (No RTP) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.2.201 : 5060 (NAT) Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-66xpqgn7lhre;rport;received=192.168.2.201 From: ;tag=idnc61o4o0 To: Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D CSeq: 6574 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-66xpqgn7lhre;rport;received=192.168.2.201 From: ;tag=idnc61o4o0 To: ;tag=as4165fb6b Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D CSeq: 6574 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69fecf39" Content-Length: 0 --- Scheduling destruction of call '3c27a27a8647-qqwympeiaqne@snom320-00041324225D' in 15000 ms efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: REGISTER sip:172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-fu536ytcf3nf;rport From: ;tag=idnc61o4o0 To: Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D CSeq: 6575 REGISTER Max-Forwards: 70 Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" User-Agent: snom320/6.2.2 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.2.201 WWW-Contact: WWW-Contact: Authorization: Digest username="204",realm="asterisk",nonce="69fecf39",uri="sip:172.16.0.14",response="602f995461770c7a6b904da51766a698",algorithm=md5 Expires: 60 Content-Length: 0 Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: REGISTER sip:172.16.0.14 SIP/2.0 (32) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-fu536ytcf3nf;rport (69) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=idnc61o4o0 (42) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (25) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D (55) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 6575 REGISTER (19) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1;q=1.0;+sip.instance="";audio;mobility="fixed";duplex="full";description="snom320";actor="principal";events="dialog";methods="INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO" (306) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: User-Agent: snom320/6.2.2 (25) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Supported: gruu (15) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: Allow-Events: dialog (20) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 11: X-Real-IP: 192.168.2.201 (24) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 12: WWW-Contact: (38) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 13: WWW-Contact: (40) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 14: Authorization: Digest username="204",realm="asterisk",nonce="69fecf39",uri="sip:172.16.0.14",response="602f995461770c7a6b904da51766a698",algorithm=md5 (150) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 15: Expires: 60 (11) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 16: Content-Length: 0 (17) Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 17: (0) --- (17 headers 0 lines)--- Jul 19 16:31:28 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received REGISTER (2) - Command in SIP REGISTER Using latest REGISTER request as basis request Sending to 192.168.2.201 : 5060 (NAT) Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-fu536ytcf3nf;rport;received=192.168.2.201 From: ;tag=idnc61o4o0 To: Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D CSeq: 6575 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- efw-voiceone-2*CLI> Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-fu536ytcf3nf;rport;received=192.168.2.201 From: ;tag=idnc61o4o0 To: ;tag=as4165fb6b Call-ID: 3c27a27a8647-qqwympeiaqne@snom320-00041324225D CSeq: 6575 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 60 Contact: ;expires=60 Date: Wed, 19 Jul 2006 14:31:28 GMT Content-Length: 0 --- Scheduling destruction of call '3c27a27a8647-qqwympeiaqne@snom320-00041324225D' in 15000 ms Jul 19 16:31:28 DEBUG[4189]: chan_sip.c:11670 sip_devicestate: Checking device state for peer 204 Jul 19 16:31:28 DEBUG[4189]: devicestate.c:187 do_state_change: Changing state for SIP/204 - state 2 (In use) efw-voiceone-2*CLI> Jul 19 16:31:28 DEBUG[4444]: app_queue.c:490 changethread: Device 'SIP/204' changed to state '2' (In use) but we don't care because they're not a member of any queue. efw-voiceone-2*CLI> <-- SIP read from 192.168.2.101:40952: Jul 19 16:31:29 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: (0) Jul 19 16:31:29 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: (0) --- (0 headers 1 lines)--- efw-voiceone-2*CLI> <-- SIP read from 192.168.2.101:40952: SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport=5060 To: ;tag=45043174 From: "204";tag=as49c9f038 Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 CSeq: 102 INVITE User-Agent: eyeBeam release 1003l stamp 30936 Content-Length: 0 Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: SIP/2.0 480 Temporarily Unavailable (35) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport=5060 (67) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: To: ;tag=45043174 (73) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: From: "204";tag=as49c9f038 (47) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 (53) efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 102 INVITE (16) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: User-Agent: eyeBeam release 1003l stamp 30936 (45) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Content-Length: 0 (17) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: (0) --- (8 headers 0 lines)--- Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:1379 __sip_ack: Acked pending invite 102 Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14' of Request 102: Match Found -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.2.101 Transmitting (no NAT) to 192.168.2.101:40952: ACK sip:304@192.168.2.101:40952;rinstance=8b1c93b54f6a0077 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK5401461e;rport From: "204" ;tag=as49c9f038 To: ;tag=45043174 Contact: Call-ID: 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- efw-voiceone-2*CLI> -- SIP/304-08ea1360 is circuit-busy Jul 19 16:31:31 DEBUG[4441]: channel.c:1336 ast_hangup: Hanging up channel 'SIP/304-08ea1360' Jul 19 16:31:31 DEBUG[4441]: chan_sip.c:2415 sip_hangup: Hangup call SIP/304-08ea1360, SIP callid 2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14) Jul 19 16:31:31 DEBUG[4441]: chan_sip.c:2423 sip_hangup: update_call_counter(304) - decrement call limit counter Jul 19 16:31:31 DEBUG[4441]: chan_sip.c:2206 update_call_counter: Updating call counter for outgoing call == Everyone is busy/congested at this time (1:0/1/0) Jul 19 16:31:31 DEBUG[4441]: app_dial.c:1628 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. Jul 19 16:31:31 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'Goto' -- Executing Goto("SIP/204-08e9d488", "s-CONGESTION|1") in new stack -- Goto (macro-stdexten,s-CONGESTION,1) Jul 19 16:31:31 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'Goto' -- Executing Goto("SIP/204-08e9d488", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) Jul 19 16:31:31 DEBUG[4441]: db.c:200 ast_db_get: Unable to find key '304' in family 'CFNR' Jul 19 16:31:31 DEBUG[4441]: func_db.c:69 function_db_read: DB: CFNR/304 not found in database. efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '' Jul 19 16:31:31 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '0' Jul 19 16:31:31 DEBUG[4441]: db.c:200 ast_db_get: Unable to find key 'SIP/304' in family 'CFNR' Jul 19 16:31:31 DEBUG[4441]: func_db.c:69 function_db_read: DB: CFNR/SIP/304 not found in database. Jul 19 16:31:31 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '' Jul 19 16:31:31 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("SIP/204-08e9d488", "0?default||1") in new stack Jul 19 16:31:31 DEBUG[4441]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch Jul 19 16:31:31 DEBUG[4441]: db.c:200 ast_db_get: Unable to find key '304' in family 'VM' Jul 19 16:31:31 DEBUG[4441]: func_db.c:69 function_db_read: DB: VM/304 not found in database. Jul 19 16:31:31 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '' Jul 19 16:31:31 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '0' Jul 19 16:31:31 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'GotoIf' -- Executing GotoIf("SIP/204-08e9d488", "0?skip-vm") in new stack Jul 19 16:31:31 DEBUG[4441]: pbx.c:6178 pbx_builtin_gotoif: Not taking any branch Jul 19 16:31:31 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'VoiceMail' -- Executing VoiceMail("SIP/204-08e9d488", "u304") in new stack Jul 19 16:31:31 DEBUG[4441]: chan_sip.c:2537 sip_answer: sip_answer(SIP/204-08e9d488) efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: chan_sip.c:11670 sip_devicestate: Checking device state for peer 304 Jul 19 16:31:31 DEBUG[4189]: devicestate.c:187 do_state_change: Changing state for SIP/304 - state 1 (Not in use) Jul 19 16:31:31 DEBUG[4189]: chan_sip.c:11670 sip_devicestate: Checking device state for peer 204 Jul 19 16:31:31 DEBUG[4189]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/204-08e9d488' efw-voiceone-2*CLI> We're at 172.16.0.14 port 17940 efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/204-08e9d488' efw-voiceone-2*CLI> Adding codec 0x8 (alaw) to SDP efw-voiceone-2*CLI> Adding codec 0x2 (gsm) to SDP efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/204-08e9d488' efw-voiceone-2*CLI> Adding codec 0x4 (ulaw) to SDP efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/204-08e9d488' efw-voiceone-2*CLI> Adding non-codec 0x1 (telephone-event) to SDP efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/204-08e9d488' efw-voiceone-2*CLI> Reliably Transmitting (no NAT) to 192.168.2.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-lu0siza2a6ff;rport;received=192.168.2.201 From: ;tag=nwdhzhgetw To: ;tag=as00574def Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 4150 4150 IN IP4 172.16.0.14 s=session c=IN IP4 172.16.0.14 t=0 0 m=audio 17940 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #170 efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'SIP/204-08e9d488' efw-voiceone-2*CLI> Jul 19 16:31:31 WARNING[4441]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '304' efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4189]: devicestate.c:187 do_state_change: Changing state for SIP/204 - state 2 (In use) efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'PlayTones' efw-voiceone-2*CLI> -- Executing PlayTones("SIP/204-08e9d488", "congestion") in new stack efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/204-08e9d488 to write format slin efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:1724 ast_settimeout: Scheduling timer at 160 sample intervals efw-voiceone-2*CLI> == Auto fallthrough, channel 'SIP/204-08e9d488' status is 'CONGESTION' efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:2049 ast_indicate: Driver for channel 'SIP/204-08e9d488' does not support indication 8, emulating it efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/204-08e9d488 to write format alaw efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/204-08e9d488 to write format slin efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:1724 ast_settimeout: Scheduling timer at 160 sample intervals efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4445]: app_queue.c:490 changethread: Device 'SIP/304' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4446]: app_queue.c:490 changethread: Device 'SIP/204' changed to state '2' (In use) but we don't care because they're not a member of any queue. efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: rtp.c:1353 ast_rtp_write: Ooh, format changed from unknown to alaw efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: ACK sip:304@172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-au0s5ywfr2zp;rport From: ;tag=nwdhzhgetw To: ;tag=as00574def Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: ACK sip:304@172.16.0.14 SIP/2.0 (31) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-au0s5ywfr2zp;rport (69) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=nwdhzhgetw (42) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: ;tag=as00574def (51) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D (55) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 2 ACK (11) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1 (61) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Content-Length: 0 (17) Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: (0) --- (9 headers 0 lines)--- Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received ACK (6) - Command in SIP ACK efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #170 Jul 19 16:31:31 DEBUG[4196]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '3c27f3f6b98c-6df77r06v810@snom320-00041324225D' of Response 2: Match Found Destroying call '2a0b89497b9c0dc17dc8dce67b1def45@172.16.0.14' Jul 19 16:31:31 DEBUG[4441]: rtp.c:411 ast_rtcp_read: Got RTCP report of 52 bytes efw-voiceone-2*CLI> Jul 19 16:31:31 DEBUG[4441]: channel.c:1988 ast_read: Generator got voice, switching to phase locked mode Jul 19 16:31:31 DEBUG[4441]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals efw-voiceone-2*CLI> Jul 19 16:31:36 DEBUG[4441]: rtp.c:411 ast_rtcp_read: Got RTCP report of 52 bytes efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: BYE sip:304@172.16.0.14 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-8j75swy7qnvw;rport From: ;tag=nwdhzhgetw To: ;tag=as00574def Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.2.2 RTP-RxStat: Total_Rx_Pkts=258,Rx_Pkts=258,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=249,Tx_Pkts=249,Remote_Tx_Pkts=0 Content-Length: 0 Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: BYE sip:304@172.16.0.14 SIP/2.0 (31) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-8j75swy7qnvw;rport (69) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: ;tag=nwdhzhgetw (42) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: ;tag=as00574def (51) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D (55) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 3 BYE (11) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Max-Forwards: 70 (16) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: Contact: ;flow-id=1 (61) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: User-Agent: snom320/6.2.2 (25) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: RTP-RxStat: Total_Rx_Pkts=258,Rx_Pkts=258,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 (78) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: RTP-TxStat: Total_Tx_Pkts=249,Tx_Pkts=249,Remote_Tx_Pkts=0 (58) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 11: Content-Length: 0 (17) Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 12: (0) --- (12 headers 0 lines)--- Jul 19 16:31:36 DEBUG[4196]: chan_sip.c:11139 handle_request: **** Received BYE (8) - Command in SIP BYE Sending to 192.168.2.201 : 5060 (NAT) Transmitting (NAT) to 192.168.2.201:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK-8j75swy7qnvw;received=192.168.2.201;rport=5060 From: ;tag=nwdhzhgetw To: ;tag=as00574def Call-ID: 3c27f3f6b98c-6df77r06v810@snom320-00041324225D CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Network out of order --- efw-voiceone-2*CLI> Jul 19 16:31:36 DEBUG[4441]: channel.c:2363 set_format: Set channel SIP/204-08e9d488 to write format alaw Jul 19 16:31:36 DEBUG[4441]: channel.c:1724 ast_settimeout: Scheduling timer at 0 sample intervals Jul 19 16:31:36 DEBUG[4441]: pbx.c:1677 pbx_extension_helper: Launching 'NoOp' -- Executing NoOp("SIP/204-08e9d488", "Hangup") in new stack Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '204' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '204' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '304' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'default' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/204-08e9d488' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'SIP/304-08ea1360' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'NoOp' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'Hangup' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-19 16:31:27' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-19 16:31:31' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '2006-07-19 16:31:36' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '9' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '5' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'ANSWERED' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is 'DOCUMENTATION' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '1153319487.18' Jul 19 16:31:36 DEBUG[4441]: pbx.c:1522 pbx_substitute_variables_helper_full: Function result is '(null)' Jul 19 16:31:36 DEBUG[4441]: channel.c:1336 ast_hangup: Hanging up channel 'SIP/204-08e9d488' Jul 19 16:31:36 DEBUG[4441]: chan_sip.c:2415 sip_hangup: Hangup call SIP/204-08e9d488, SIP callid 3c27f3f6b98c-6df77r06v810@snom320-00041324225D) Jul 19 16:31:36 DEBUG[4441]: chan_sip.c:2423 sip_hangup: update_call_counter(204) - decrement call limit counter Jul 19 16:31:36 DEBUG[4441]: chan_sip.c:2206 update_call_counter: Updating call counter for incoming call Jul 19 16:31:36 DEBUG[4189]: chan_sip.c:11670 sip_devicestate: Checking device state for peer 204 efw-voiceone-2*CLI> Jul 19 16:31:36 DEBUG[4189]: devicestate.c:187 do_state_change: Changing state for SIP/204 - state 1 (Not in use) efw-voiceone-2*CLI> Jul 19 16:31:36 DEBUG[4447]: app_queue.c:490 changethread: Device 'SIP/204' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. efw-voiceone-2*CLI> Destroying call '3c27f3f6b98c-6df77r06v810@snom320-00041324225D' efw-voiceone-2*CLI> Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3141 sip_alloc: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: NOTIFY sip:204@192.168.2.201:5060;line=8hscseyw SIP/2.0 (55) efw-voiceone-2*CLI> Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK760cfbd6;rport (62) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: "asterisk" ;tag=as233d8583 (58) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (46) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Contact: (35) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: Call-ID: 430eab1e2a1557355f777642320a8669@172.16.0.14 (53) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: CSeq: 102 NOTIFY (16) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: User-Agent: Asterisk PBX (24) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 8: Max-Forwards: 70 (16) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 9: Event: message-summary (22) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 10: Content-Type: application/simple-message-summary (48) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 11: Content-Length: 86 (18) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 12: (0) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Messages-Waiting: no (20) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Message-Account: sip:*98@172.16.0.14 (36) Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:3389 parse_request: Line: Voice-Message: 0/0 (0/0) (24) 12 headers, 3 lines Reliably Transmitting (no NAT) to 192.168.2.201:5060: NOTIFY sip:204@192.168.2.201:5060;line=8hscseyw SIP/2.0 Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK760cfbd6;rport From: "asterisk" ;tag=as233d8583 To: Contact: Call-ID: 430eab1e2a1557355f777642320a8669@172.16.0.14 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 86 Messages-Waiting: no Message-Account: sip:*98@172.16.0.14 Voice-Message: 0/0 (0/0) --- Jul 19 16:31:37 DEBUG[4196]: chan_sip.c:1293 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #171 Scheduling destruction of call '430eab1e2a1557355f777642320a8669@172.16.0.14' in 15000 ms efw-voiceone-2*CLI> <-- SIP read from 192.168.2.201:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK760cfbd6;rport=5060 From: "asterisk" ;tag=as233d8583 To: Call-ID: 430eab1e2a1557355f777642320a8669@172.16.0.14 CSeq: 102 NOTIFY Content-Length: 0 Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 0: SIP/2.0 200 Ok (14) Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 1: Via: SIP/2.0/UDP 172.16.0.14:5060;branch=z9hG4bK760cfbd6;rport=5060 (67) Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 2: From: "asterisk" ;tag=as233d8583 (58) Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 3: To: (46) Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 4: Call-ID: 430eab1e2a1557355f777642320a8669@172.16.0.14 (53) Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 5: CSeq: 102 NOTIFY (16) Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 6: Content-Length: 0 (17) efw-voiceone-2*CLI> Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:3357 parse_request: Header 7: (0) --- (7 headers 0 lines)--- Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:1390 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #171 Jul 19 16:31:38 DEBUG[4196]: chan_sip.c:1401 __sip_ack: Stopping retransmission on '430eab1e2a1557355f777642320a8669@172.16.0.14' of Request 102: Match Found Destroying call '430eab1e2a1557355f777642320a8669@172.16.0.14' efw-voiceone-2*CLI>