Sep 25 14:38:37 VERBOSE[14089] logger.c: Asterisk Ready. Asterisk Event Logger restarted Sep 25 14:38:41 DEBUG[14110] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - NOTIFY (No RTP) Sep 25 14:38:41 VERBOSE[14110] logger.c: Destroying call '411c87440cddd1ee63b8f8d34a823464@sip.babytel.ca' Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 192.168.100.199:5060: INVITE sip:95149313015@192.168.100.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-dca8bea4 From: ;tag=567ef9047499efa4o0 To: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 101 INVITE Max-Forwards: 70 Contact: Expires: 240 User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 240 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 66137 66137 IN IP4 192.168.100.199 s=- c=IN IP4 192.168.100.199 t=0 0 m=audio 16428 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: INVITE sip:95149313015@192.168.100.30 SIP/2.0 (45) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-dca8bea4 (61) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: ;tag=567ef9047499efa4o0 (57) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: (36) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 (42) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 101 INVITE (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Max-Forwards: 70 (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Contact: (43) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Expires: 240 (12) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: User-Agent: Sipura/SPA3000-2.0.13(GWg) (38) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 240 (19) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Supported: x-sipura (19) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Type: application/sdp (29) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: v=0 (3) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: o=- 66137 66137 IN IP4 192.168.100.199 (38) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: s=- (3) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: c=IN IP4 192.168.100.199 (24) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: t=0 0 (5) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: m=audio 16428 RTP/AVP 18 100 101 (32) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=rtpmap:18 G729a/8000 (22) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=rtpmap:100 NSE/8000 (21) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=fmtp:101 0-15 (15) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=ptime:30 (10) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=sendrecv (10) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (14 headers 12 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (14 headers 12 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Allocating new SIP dialog for e6cf3c84-5f1a8224@192.168.100.199 - INVITE (With RTP) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Begin: parsing SIP "Supported: x-sipura" Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Found SIP option: -x-sipura- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Found no match for SIP option: x-sipura (Please file bug report!) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: * SIP extension value: 0 for call e6cf3c84-5f1a8224@192.168.100.199 Sep 25 14:38:45 VERBOSE[14110] logger.c: Using INVITE request as basis request - e6cf3c84-5f1a8224@192.168.100.199 Sep 25 14:38:45 VERBOSE[14110] logger.c: Sending to 192.168.100.199 : 5060 (non-NAT) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Setting NAT on RTP to 524288 Sep 25 14:38:45 VERBOSE[14110] logger.c: Reliably Transmitting (NAT) to 192.168.100.199:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-dca8bea4;received=192.168.100.199 From: ;tag=567ef9047499efa4o0 To: ;tag=as7fbc9922 Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="75e6b1d4" Content-Length: 0 --- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #14 Sep 25 14:38:45 VERBOSE[14110] logger.c: Scheduling destruction of call 'e6cf3c84-5f1a8224@192.168.100.199' in 15000 ms Sep 25 14:38:45 VERBOSE[14110] logger.c: Found user 'testata' Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 192.168.100.199:5060: ACK sip:95149313015@192.168.100.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-dca8bea4 From: ;tag=567ef9047499efa4o0 To: ;tag=as7fbc9922 Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 101 ACK Max-Forwards: 70 Contact: User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: ACK sip:95149313015@192.168.100.30 SIP/2.0 (42) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-dca8bea4 (61) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: ;tag=567ef9047499efa4o0 (57) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: ;tag=as7fbc9922 (51) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 (42) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 101 ACK (13) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Max-Forwards: 70 (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Contact: (43) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: User-Agent: Sipura/SPA3000-2.0.13(GWg) (38) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (10 headers 0 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (10 headers 0 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 25 14:38:45 DEBUG[14110] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Stopping retransmission on 'e6cf3c84-5f1a8224@192.168.100.199' of Response 101: Match Found Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 192.168.100.199:5060: INVITE sip:95149313015@192.168.100.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-9363ff4 From: ;tag=567ef9047499efa4o0 To: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="testata",realm="asterisk",nonce="75e6b1d4",uri="sip:95149313015@192.168.100.30",algorithm=MD5,response="d4580e8adb3963dad6f347dac4257777" Contact: Expires: 240 User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 240 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 66137 66137 IN IP4 192.168.100.199 s=- c=IN IP4 192.168.100.199 t=0 0 m=audio 16428 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: INVITE sip:95149313015@192.168.100.30 SIP/2.0 (45) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-9363ff4 (60) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: ;tag=567ef9047499efa4o0 (57) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: (36) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 (42) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 102 INVITE (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Max-Forwards: 70 (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Proxy-Authorization: Digest username="testata",realm="asterisk",nonce="75e6b1d4",uri="sip:95149313015@192.168.100.30",algorithm=MD5,response="d4580e8adb3963dad6f347dac4257777" (175) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Contact: (43) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Expires: 240 (12) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: User-Agent: Sipura/SPA3000-2.0.13(GWg) (38) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 240 (19) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Supported: x-sipura (19) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Type: application/sdp (29) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: v=0 (3) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: o=- 66137 66137 IN IP4 192.168.100.199 (38) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: s=- (3) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: c=IN IP4 192.168.100.199 (24) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: t=0 0 (5) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: m=audio 16428 RTP/AVP 18 100 101 (32) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=rtpmap:18 G729a/8000 (22) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=rtpmap:100 NSE/8000 (21) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=fmtp:101 0-15 (15) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=ptime:30 (10) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Line: a=sendrecv (10) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (15 headers 12 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (15 headers 12 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Begin: parsing SIP "Supported: x-sipura" Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Found SIP option: -x-sipura- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Found no match for SIP option: x-sipura (Please file bug report!) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: * SIP extension value: 0 for call e6cf3c84-5f1a8224@192.168.100.199 Sep 25 14:38:45 VERBOSE[14110] logger.c: Using INVITE request as basis request - e6cf3c84-5f1a8224@192.168.100.199 Sep 25 14:38:45 VERBOSE[14110] logger.c: Sending to 192.168.100.199 : 5060 (NAT) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Setting NAT on RTP to 524288 Sep 25 14:38:45 VERBOSE[14110] logger.c: Found user 'testata' Sep 25 14:38:45 VERBOSE[14110] logger.c: Found RTP audio format 18 Sep 25 14:38:45 VERBOSE[14110] logger.c: Found RTP audio format 100 Sep 25 14:38:45 VERBOSE[14110] logger.c: Found RTP audio format 101 Sep 25 14:38:45 VERBOSE[14110] logger.c: Peer audio RTP is at port 192.168.100.199:16428 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Peer audio RTP is at port 192.168.100.199:16428 Sep 25 14:38:45 VERBOSE[14110] logger.c: Found description format G729a Sep 25 14:38:45 VERBOSE[14110] logger.c: Found description format NSE Sep 25 14:38:45 VERBOSE[14110] logger.c: Found description format telephone-event Sep 25 14:38:45 VERBOSE[14110] logger.c: Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Sep 25 14:38:45 VERBOSE[14110] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Checking SIP call limits for device testata Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Updating call counter for incoming call Sep 25 14:38:45 VERBOSE[14110] logger.c: Looking for 95149313015 in company2-internal Sep 25 14:38:45 DEBUG[14110] chan_sip.c: build_route: Contact hop: Sep 25 14:38:45 VERBOSE[14110] logger.c: list_route: hop: Sep 25 14:38:45 VERBOSE[14110] logger.c: Transmitting (NAT) to 192.168.100.199:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-9363ff4;received=192.168.100.199 From: ;tag=567ef9047499efa4o0 To: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for peer testata Sep 25 14:38:45 DEBUG[14094] channel.c: Avoiding initial deadlock for 'SIP/testata-4cd3' Sep 25 14:38:45 DEBUG[14115] pbx.c: Launching 'Dial' Sep 25 14:38:45 VERBOSE[14115] logger.c: -- Executing Dial("SIP/testata-4cd3", "SIP/5149313015@babytel-out") in new stack Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Setting NAT on RTP to 524288 Sep 25 14:38:45 DEBUG[14094] devicestate.c: Changing state for SIP/testata - state 2 (In use) Sep 25 14:38:45 DEBUG[14117] app_queue.c: Device 'SIP/testata' changed to state '2' (In use) Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for peer testata Sep 25 14:38:45 DEBUG[14115] channel.c: Not copying variable STACK-company2-internal-95149313015-1. Sep 25 14:38:45 DEBUG[14115] channel.c: Not copying variable SIPCALLID. Sep 25 14:38:45 DEBUG[14115] channel.c: Not copying variable SIPUSERAGENT. Sep 25 14:38:45 DEBUG[14115] channel.c: Not copying variable SIPDOMAIN. Sep 25 14:38:45 DEBUG[14115] channel.c: Not copying variable SIPURI. Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Outgoing Call for 5149313015 Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Updating call counter for outgoing call Sep 25 14:38:45 VERBOSE[14115] logger.c: We're at 192.168.100.30 port 14412 Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering/Requesting with root capability 0x100 (g729) Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering with preferred capability 0x2 (gsm) Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering with preferred capability 0x4 (ulaw) Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering with preferred capability 0x8 (alaw) Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering with preferred capability 0x400 (ilbc) Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering with preferred capability 0x1 (g723) Sep 25 14:38:45 VERBOSE[14115] logger.c: Answering with non-codec capability 0x1 (telephone-event) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: INVITE sip:5149313015@sip.babytel.ca:5065 SIP/2.0 (49) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport (65) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: From: "asterisk" ;tag=as72e80853 (64) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: To: (40) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Contact: (41) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca (56) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: CSeq: 102 INVITE (16) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: User-Agent: Asterisk PBX (24) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Date: Sun, 25 Sep 2005 18:38:45 GMT (35) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Content-Type: application/sdp (29) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: Content-Length: 343 (19) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: v=0 (3) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: o=root 14115 14115 IN IP4 192.168.100.30 (40) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: s=session (9) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: c=IN IP4 192.168.100.30 (23) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: t=0 0 (5) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: m=audio 14412 RTP/AVP 18 3 0 8 97 4 101 (39) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:18 G729/8000 (21) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:3 GSM/8000 (19) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:0 PCMU/8000 (20) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:97 iLBC/8000 (21) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:4 G723/8000 (20) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=fmtp:101 0-16 (15) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Line: a=silenceSupp:off - - - - (25) Sep 25 14:38:45 VERBOSE[14115] logger.c: 12 headers, 15 lines Sep 25 14:38:45 VERBOSE[14115] logger.c: Reliably Transmitting (NAT) to 216.18.125.7:5065: INVITE sip:5149313015@sip.babytel.ca:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport From: "asterisk" ;tag=as72e80853 To: Contact: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sun, 25 Sep 2005 18:38:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 343 v=0 o=root 14115 14115 IN IP4 192.168.100.30 s=session c=IN IP4 192.168.100.30 t=0 0 m=audio 14412 RTP/AVP 18 3 0 8 97 4 101 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 25 14:38:45 DEBUG[14115] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #16 Sep 25 14:38:45 VERBOSE[14115] logger.c: -- Called 5149313015@babytel-out Sep 25 14:38:45 DEBUG[14115] channel.c: Set channel SIP/babytel-out-406a to read format slin Sep 25 14:38:45 DEBUG[14115] channel.c: Set channel SIP/testata-4cd3 to write format slin Sep 25 14:38:45 DEBUG[14115] channel.c: Set channel SIP/testata-4cd3 to read format slin Sep 25 14:38:45 DEBUG[14115] channel.c: Set channel SIP/babytel-out-406a to write format slin Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 216.18.125.7:5065: SIP/2.0 100 Trying To: From: "asterisk";tag=as72e80853 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport=8604;received=69.157.240.5 Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 102 INVITE Server: JasomiNetworks-PeerPoint/3.1.7 rc-39 Content-Length: 0 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: SIP/2.0 100 Trying (18) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: (40) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: "asterisk";tag=as72e80853 (63) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport=8604;received=69.157.240.5 (92) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca (56) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 102 INVITE (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Server: JasomiNetworks-PeerPoint/3.1.7 rc-39 (44) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (8 headers 0 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (8 headers 0 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: n uri sip:5149313015@sip.babytel.ca:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: m uri sip:5149313015@sip.babytel.ca:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: f uri sip:5149313015@sip.babytel.ca:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: *** SIP TIMER: Cancelling retransmission #16 - INVITE (got response) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca' Request 102: Found Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: g uri sip:5149313015@sip.babytel.ca:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: SIP response 100 to standard invite Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 216.18.125.7:5065: SIP/2.0 407 Proxy Authentication Required To: ;tag=ef6fff77 From: "asterisk";tag=as72e80853 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport=8604;received=69.157.240.5 Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 102 INVITE Contact: Proxy-Authenticate: Digest realm="sip.babytel.ca", nonce="4336ef62c1ca1344a0137629ac6ee63710aa2d65" Content-Length: 0 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: SIP/2.0 407 Proxy Authentication Required (41) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: ;tag=ef6fff77 (53) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: "asterisk";tag=as72e80853 (63) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport=8604;received=69.157.240.5 (92) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca (56) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 102 INVITE (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Contact: (48) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Proxy-Authenticate: Digest realm="sip.babytel.ca", nonce="4336ef62c1ca1344a0137629ac6ee63710aa2d65" (99) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (9 headers 0 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (9 headers 0 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: n uri sip:5149313015@sip.babytel.ca:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: m uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: f uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Acked pending invite 102 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Stopping retransmission on '28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca' of Request 102: Match Found Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: g uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: SIP response 407 to standard invite Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: h uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: i ok full our Sep 25 14:38:45 VERBOSE[14110] logger.c: Transmitting (NAT) to 216.18.125.7:5065: ACK sip:5149313015@sip.babytel.ca:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK7d0a8d78;rport From: "asterisk" ;tag=as72e80853 To: ;tag=ef6fff77 Contact: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: j ok full our Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: k uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Auth attempt 1 on INVITE Sep 25 14:38:45 VERBOSE[14110] logger.c: We're at 192.168.100.30 port 14412 Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering/Requesting with root capability 0x100 (g729) Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering with preferred capability 0x2 (gsm) Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering with preferred capability 0x4 (ulaw) Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering with preferred capability 0x8 (alaw) Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering with preferred capability 0x400 (ilbc) Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering with preferred capability 0x1 (g723) Sep 25 14:38:45 VERBOSE[14110] logger.c: Answering with non-codec capability 0x1 (telephone-event) Sep 25 14:38:45 VERBOSE[14110] logger.c: Reliably Transmitting (NAT) to 216.18.125.7:5065: INVITE sip:hQ9-L-101454101@216.18.125.7:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK30fbfc05;rport From: "asterisk" ;tag=as72e80853 To: Contact: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="15144484051", realm="sip.babytel.ca", algorithm=MD5, uri="sip:hQ9-L-101454101@216.18.125.7:5065", nonce="4336ef62c1ca1344a0137629ac6ee63710aa2d65", response="c42af35dc21a0553d931edf7dcce55bb", opaque="" Date: Sun, 25 Sep 2005 18:38:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 343 v=0 o=root 14115 14116 IN IP4 192.168.100.30 s=session c=IN IP4 192.168.100.30 t=0 0 m=audio 14412 RTP/AVP 18 3 0 8 97 4 101 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #17 Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 216.18.125.7:5065: SIP/2.0 100 Trying To: From: "asterisk";tag=as72e80853 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK30fbfc05;rport=8604;received=69.157.240.5 Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 103 INVITE Server: JasomiNetworks-PeerPoint/3.1.7 rc-39 Content-Length: 0 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: SIP/2.0 100 Trying (18) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: (40) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: "asterisk";tag=as72e80853 (63) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK30fbfc05;rport=8604;received=69.157.240.5 (92) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca (56) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 103 INVITE (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Server: JasomiNetworks-PeerPoint/3.1.7 rc-39 (44) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (8 headers 0 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (8 headers 0 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: n uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: m uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: f uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: *** SIP TIMER: Cancelling retransmission #17 - INVITE (got response) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca' Request 103: Found Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: g uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: SIP response 100 to standard invite Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 216.18.125.7:5065: SIP/2.0 503 Service Unavailable To: ;tag=1f480401 From: "asterisk";tag=as72e80853 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK30fbfc05;rport=8604;received=69.157.240.5 Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 103 INVITE Contact: Warning: 499 nat-001.mtl.pop.vds.ca "No other DNS entries to try" Content-Length: 0 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: SIP/2.0 503 Service Unavailable (31) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: ;tag=1f480401 (53) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: "asterisk";tag=as72e80853 (63) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK30fbfc05;rport=8604;received=69.157.240.5 (92) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca (56) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 103 INVITE (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Contact: (48) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Warning: 499 nat-001.mtl.pop.vds.ca "No other DNS entries to try" (65) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (9 headers 0 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (9 headers 0 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: n uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: m uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: f uri sip:hQ9-L-101454101@216.18.125.7:5065 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Acked pending invite 103 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Stopping retransmission on '28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca' of Request 103: Match Found Sep 25 14:38:45 VERBOSE[14110] logger.c: -- Got SIP response 503 "Service Unavailable" back from 216.18.125.7 Sep 25 14:38:45 VERBOSE[14110] logger.c: Transmitting (NAT) to 216.18.125.7:5065: ACK sip:5149313015@sip.babytel.ca:5065 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK30fbfc05;rport From: "asterisk" ;tag=as72e80853 To: ;tag=1f480401 Contact: Call-ID: 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 --- Sep 25 14:38:45 VERBOSE[14115] logger.c: -- SIP/babytel-out-406a is circuit-busy Sep 25 14:38:45 DEBUG[14115] channel.c: Hanging up channel 'SIP/babytel-out-406a' Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Hangup call SIP/babytel-out-406a, SIP callid 28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: update_call_counter(5149313015) - decrement call limit counter Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Updating call counter for outgoing call Sep 25 14:38:45 VERBOSE[14115] logger.c: == Everyone is busy/congested at this time (1:0/1/0) Sep 25 14:38:45 DEBUG[14115] app_dial.c: Exiting with DIALSTATUS=CONGESTION. Sep 25 14:38:45 VERBOSE[14115] logger.c: == Auto fallthrough, channel 'SIP/testata-4cd3' status is 'CONGESTION' Sep 25 14:38:45 VERBOSE[14115] logger.c: Transmitting (NAT) to 192.168.100.199:5060: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-9363ff4;received=192.168.100.199 From: ;tag=567ef9047499efa4o0 To: ;tag=as4fae6a55 Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Circuit/channel congestion --- Sep 25 14:38:45 DEBUG[14115] channel.c: Soft-Hanging up channel 'SIP/testata-4cd3' Sep 25 14:38:45 VERBOSE[14110] logger.c: <-- SIP read from 192.168.100.199:5060: ACK sip:95149313015@192.168.100.30 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-9363ff4 From: ;tag=567ef9047499efa4o0 To: ;tag=as4fae6a55 Call-ID: e6cf3c84-5f1a8224@192.168.100.199 CSeq: 102 ACK Max-Forwards: 70 Proxy-Authorization: Digest username="testata",realm="asterisk",nonce="75e6b1d4",uri="sip:95149313015@192.168.100.30",algorithm=MD5,response="f2dde2a9ba665e55e2cac4904a419710" Contact: User-Agent: Sipura/SPA3000-2.0.13(GWg) Content-Length: 0 Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: ACK sip:95149313015@192.168.100.30 SIP/2.0 (42) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.199:5060;branch=z9hG4bK-9363ff4 (60) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: From: ;tag=567ef9047499efa4o0 (57) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: To: ;tag=as4fae6a55 (51) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Call-ID: e6cf3c84-5f1a8224@192.168.100.199 (42) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: CSeq: 102 ACK (13) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Max-Forwards: 70 (16) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Proxy-Authorization: Digest username="testata",realm="asterisk",nonce="75e6b1d4",uri="sip:95149313015@192.168.100.30",algorithm=MD5,response="f2dde2a9ba665e55e2cac4904a419710" (175) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Contact: (43) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: User-Agent: Sipura/SPA3000-2.0.13(GWg) (38) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (11 headers 0 lines)Sep 25 14:38:45 VERBOSE[14110] logger.c: --- (11 headers 0 lines)--- Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Failed to grab lock, trying again... Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Failed to grab lock, trying again... Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for DNS host babytel Sep 25 14:38:45 DEBUG[14094] devicestate.c: Changing state for SIP/babytel - state 4 (Invalid) Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for peer testata Sep 25 14:38:45 DEBUG[14094] channel.c: Avoiding initial deadlock for 'SIP/testata-4cd3' Sep 25 14:38:45 DEBUG[14118] app_queue.c: Device 'SIP/babytel' changed to state '4' (Invalid) Sep 25 14:38:45 DEBUG[14094] channel.c: Avoiding initial deadlock for 'SIP/testata-4cd3' Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Failed to grab lock, trying again... Sep 25 14:38:45 DEBUG[14094] channel.c: Avoiding initial deadlock for 'SIP/testata-4cd3' Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Failed to grab lock, trying again... Sep 25 14:38:45 VERBOSE[14115] logger.c: > cdr_odbc: Query Successful! Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'testata' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'testata' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '95149313015' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'company2-internal' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'SIP/testata-4cd3' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'SIP/babytel-out-406a' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'Dial' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'SIP/5149313015@babytel-out' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '2005-09-25 14:38:45' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '(null)' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '2005-09-25 14:38:45' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '0' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '0' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'BUSY' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is 'DOCUMENTATION' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '(null)' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '1127673525.0' Sep 25 14:38:45 DEBUG[14115] pbx.c: Function result is '(null)' Sep 25 14:38:45 DEBUG[14115] channel.c: Hanging up channel 'SIP/testata-4cd3' Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Hangup call SIP/testata-4cd3, SIP callid e6cf3c84-5f1a8224@192.168.100.199) Sep 25 14:38:45 DEBUG[14115] chan_sip.c: update_call_counter(testata) - decrement call limit counter Sep 25 14:38:45 DEBUG[14115] chan_sip.c: Updating call counter for incoming call Sep 25 14:38:45 DEBUG[14094] devicestate.c: Changing state for SIP/testata - state 1 (Not in use) Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for peer testata Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for peer testata Sep 25 14:38:45 DEBUG[14094] devicestate.c: Changing state for SIP/testata - state 1 (Not in use) Sep 25 14:38:45 DEBUG[14119] app_queue.c: Device 'SIP/testata' changed to state '1' (Not in use) Sep 25 14:38:45 DEBUG[14094] chan_sip.c: Checking device state for peer testata Sep 25 14:38:45 DEBUG[14120] app_queue.c: Device 'SIP/testata' changed to state '1' (Not in use) Sep 25 14:38:45 DEBUG[14110] chan_sip.c: **** Received ACK (6) - Command in SIP ACK Sep 25 14:38:45 DEBUG[14110] chan_sip.c: Stopping retransmission on 'e6cf3c84-5f1a8224@192.168.100.199' of Response 102: Match Not Found Sep 25 14:38:45 VERBOSE[14110] logger.c: Destroying call '28ef82dc3f2047291fa3e46c4a9cc9d0@sip.babytel.ca' Sep 25 14:38:45 VERBOSE[14110] logger.c: Destroying call 'e6cf3c84-5f1a8224@192.168.100.199' Sep 25 14:38:53 NOTICE[14110] chan_sip.c: -- Re-registration for 15144484051@sip.babytel.ca Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Allocating new SIP dialog for 06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca - REGISTER (No RTP) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Scheduled a registration timeout for sip.babytel.ca id #18 Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: REGISTER sip:sip.babytel.ca SIP/2.0 (35) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK02d9b42b (59) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: From: ;tag=as0bbd46cb (53) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: To: (36) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Call-ID: 06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca (56) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: CSeq: 103 REGISTER (18) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: User-Agent: Asterisk PBX (24) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Expires: 100 (12) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Contact: (41) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Event: registration (19) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:53 VERBOSE[14110] logger.c: REGISTER 11 headers, 0 lines Sep 25 14:38:53 VERBOSE[14110] logger.c: REGISTER attempt 1 to 15144484051@sip.babytel.ca Sep 25 14:38:53 VERBOSE[14110] logger.c: Reliably Transmitting (no NAT) to 216.18.125.7:5065: REGISTER sip:sip.babytel.ca SIP/2.0 Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK02d9b42b From: ;tag=as0bbd46cb To: Call-ID: 06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca CSeq: 103 REGISTER User-Agent: Asterisk PBX Expires: 100 Contact: Event: registration Content-Length: 0 --- Sep 25 14:38:53 DEBUG[14110] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #19 Sep 25 14:38:53 VERBOSE[14110] logger.c: <-- SIP read from 216.18.125.7:5065: SIP/2.0 200 OK To: ;tag=8bf3da4b From: ;tag=as0bbd46cb Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK02d9b42b;received=69.157.240.5 Call-ID: 06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca CSeq: 103 REGISTER Contact: ;expires=38 Content-Length: 0 Sep 25 14:38:53 DEBUG[14110] chan_sip.c: XDEBUG: e uri ii Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: SIP/2.0 200 OK (14) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: To: ;tag=8bf3da4b (49) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: From: ;tag=as0bbd46cb (53) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Via: SIP/2.0/UDP 192.168.100.30:5060;branch=z9hG4bK02d9b42b;received=69.157.240.5 (81) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Call-ID: 06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca (56) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: CSeq: 103 REGISTER (18) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Contact: ;expires=38 (52) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: Content-Length: 0 (17) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Header: (0) Sep 25 14:38:53 DEBUG[14110] chan_sip.c: XDEBUG: z uri ii Sep 25 14:38:53 VERBOSE[14110] logger.c: --- (8 headers 0 lines)Sep 25 14:38:53 VERBOSE[14110] logger.c: --- (8 headers 0 lines)--- Sep 25 14:38:53 DEBUG[14110] chan_sip.c: XDEBUG: n uri sip:sip.babytel.ca Sep 25 14:38:53 DEBUG[14110] chan_sip.c: XDEBUG: m uri sip:15144484051@192.168.100.30 Sep 25 14:38:53 DEBUG[14110] chan_sip.c: XDEBUG: f uri sip:15144484051@192.168.100.30 Sep 25 14:38:53 DEBUG[14110] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #19 Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Stopping retransmission on '06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca' of Request 103: Match Found Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Registration successful Sep 25 14:38:53 DEBUG[14110] chan_sip.c: Cancelling timeout 18 Sep 25 14:38:53 VERBOSE[14110] logger.c: Scheduling destruction of call '06361b3f6b313a3b0d01b60b24948113@sip.babytel.ca' in 32000 ms Sep 25 14:38:53 NOTICE[14110] chan_sip.c: Outbound Registration: Expiry for sip.babytel.ca is 38 sec (Scheduling reregistration in 23 s)