### Call originated from Asterisk to sip:ibc@aliax.net ### CLI>console dial ibc [Dec 13 13:35:28] DEBUG[26331] pbx.c: Launching 'Dial' [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Asked to create a SIP channel with formats: 0x40 (slin) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Setting NAT on RTP to On [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Dec 13 13:35:28] DEBUG[26331] acl.c: Found IP address for this socket [Dec 13 13:35:28] DEBUG[26331] frame.c: Could not find preferred codec - Going for the best codec [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** Our native formats are 0x40 (slin) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** Joint capabilities are 0x40 (slin) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** Our capabilities are 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140) [Dec 13 13:35:28] DEBUG[26331] frame.c: Could not find preferred codec - Going for the best codec [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x40 (slin) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: This channel will not be able to handle video. [Dec 13 13:35:28] DEBUG[26331] rtp.c: Channel 'OSS/dsp' has no RTP, not doing anything [Dec 13 13:35:28] DEBUG[26331] channel.c: Not copying variable STACK-default-ibc-1. [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Outgoing Call for ibc [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Updating call counter for outgoing call [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Our T38 capability (0), joint T38 capability (0) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: ** Our capability: 0x27f9cfe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140) Video flag: False Text flag: False [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: ** Our prefcodec: 0x40 (slin) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: This call needs video offers, but there's no video support enabled! [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: This call needs text offers, but there's no text support enabled ! [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: -- Done with adding codecs to SDP [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Done building SDP. Settling with this capability: 0x27f9cfe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Initializing initreq for method INVITE - callid 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 0 [ 32]: INVITE sip:ibc@aliax.net SIP/2.0 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 1 [ 62]: Via: SIP/2.0/UDP 88.95.0.111:5060;branch=z9hG4bK71896ec5;rport [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 3 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 4 [ 23]: To: [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 5 [ 35]: Contact: [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 6 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 8 [ 30]: User-Agent: Asterisk SVN trunk [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 9 [ 35]: Date: Thu, 13 Dec 2007 12:35:28 GMT [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 10 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 11 [ 19]: Supported: replaces [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 13 [ 19]: Content-Length: 526 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 14 [ 0]: [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 0 [ 3]: v=0 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 1 [ 47]: o=root 1743511099 1743511099 IN IP4 88.95.0.111 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 2 [ 31]: s=Asterisk PBX SVN-trunk-r89407 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 3 [ 20]: c=IN IP4 88.95.0.111 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 4 [ 5]: t=0 0 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 5 [ 51]: m=audio 10298 RTP/AVP 10 8 111 3 0 112 5 7 97 9 101 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 6 [ 20]: a=rtpmap:10 L16/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 7 [ 20]: a=rtpmap:8 PCMA/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 8 [ 25]: a=rtpmap:111 G726-32/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 9 [ 19]: a=rtpmap:3 GSM/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 10 [ 20]: a=rtpmap:0 PCMU/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 11 [ 30]: a=rtpmap:112 AAL2-G726-32/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 12 [ 20]: a=rtpmap:5 DVI4/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 13 [ 19]: a=rtpmap:7 LPC/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 14 [ 21]: a=rtpmap:97 iLBC/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 15 [ 17]: a=fmtp:97 mode=30 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 16 [ 21]: a=rtpmap:9 G722/16000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 17 [ 33]: a=rtpmap:101 telephone-event/8000 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 18 [ 15]: a=fmtp:101 0-16 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 19 [ 25]: a=silenceSupp:off - - - - [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 20 [ 10]: a=ptime:20 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Body 21 [ 10]: a=sendrecv [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #12924 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 0 [ 52]: SIP/2.0 100 Trying -- your money is important for me [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;branch=z9hG4bK71896ec5;rport=5060 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 3 [ 23]: To: [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 6 [ 45]: Server: OpenSER (1.3.0-pre1-tls (i386/linux)) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: *** SIP TIMER: Cancelling retransmission #12924 - INVITE (got response) [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f10b486661309627b68b4f971c1cd27@88.95.0.111' Request 102: Found [Dec 13 13:35:28] DEBUG[26331] chan_sip.c: SIP response 100 to standard invite [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;rport=5060;branch=z9hG4bK71896ec5 [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 2 [ 33]: To: ;tag=veuje [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 3 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 6 [ 19]: Server: Twinkle/1.1 [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7f10b486661309627b68b4f971c1cd27@88.95.0.111' Request 102: Found [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: SIP response 180 to standard invite [Dec 13 13:35:29] DEBUG[26331] devicestate.c: Notification of state change to be queued on device/channel SIP/aliax.net-08228b38 [Dec 13 13:35:29] DEBUG[26331] devicestate.c: No provider found, checking channel drivers for SIP - aliax.net [Dec 13 13:35:29] DEBUG[26331] chan_sip.c: Checking device state for peer aliax.net [Dec 13 13:35:29] DEBUG[26331] devicestate.c: Changing state for SIP/aliax.net - state 1 (Not in use) [Dec 13 13:35:29] DEBUG[26331] app_queue.c: Device 'SIP/aliax.net' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. ### CLI> console hangup [Dec 13 13:35:36] DEBUG[26331] channel.c: Hanging up channel 'SIP/aliax.net-08228b38' [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Hangup call SIP/aliax.net-08228b38, SIP callid 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Hanging up channel in state Ringing (not UP) [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Acked pending invite 102 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Stopping retransmission on '7f10b486661309627b68b4f971c1cd27@88.95.0.111' of Request 102: Match Found [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #12930 [Dec 13 13:35:36] DEBUG[26331] devicestate.c: Notification of state change to be queued on device/channel SIP/aliax.net-08228b38 [Dec 13 13:35:36] DEBUG[26331] app_dial.c: Exiting with DIALSTATUS=CANCEL. [Dec 13 13:35:36] DEBUG[26331] pbx.c: Spawn extension (default,ibc,1) exited non-zero on 'OSS/dsp' [Dec 13 13:35:36] DEBUG[26331] channel.c: Soft-Hanging up channel 'OSS/dsp' [Dec 13 13:35:36] DEBUG[26331] channel.c: Hanging up channel 'OSS/dsp' [Dec 13 13:35:36] DEBUG[26331] devicestate.c: Notification of state change to be queued on device/channel OSS/dsp [Dec 13 13:35:36] DEBUG[26331] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 0 [ 21]: SIP/2.0 200 canceling [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;branch=z9hG4bK71896ec5;rport=5060 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 3 [ 65]: To: ;tag=373b8885156114ecb2c4bd665f9faf0b-f9aa [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 6 [ 45]: Server: OpenSER (1.3.0-pre1-tls (i386/linux)) [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: That's odd... Got a response on a call we dont know about. Callid 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Invalid SIP message - rejected , no callid, len 350 [Dec 13 13:35:36] DEBUG[26331] devicestate.c: No provider found, checking channel drivers for SIP - aliax.net [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Checking device state for peer aliax.net [Dec 13 13:35:36] DEBUG[26331] devicestate.c: Changing state for SIP/aliax.net - state 1 (Not in use) [Dec 13 13:35:36] DEBUG[26331] devicestate.c: No provider found, checking channel drivers for OSS - dsp [Dec 13 13:35:36] DEBUG[26331] devicestate.c: Changing state for OSS/dsp - state 4 (Invalid) [Dec 13 13:35:36] DEBUG[26331] app_queue.c: Device 'SIP/aliax.net' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Dec 13 13:35:36] DEBUG[26331] app_queue.c: Device 'OSS/dsp' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 0 [ 30]: SIP/2.0 487 Request Terminated [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;rport=5060;branch=z9hG4bK71896ec5 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 2 [ 33]: To: ;tag=veuje [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 3 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 6 [ 19]: Server: Twinkle/1.1 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Stopping retransmission on '7f10b486661309627b68b4f971c1cd27@88.95.0.111' of Request 102: Match Not Found [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: SIP response 487 to standard invite [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Updating call counter for outgoing call [Dec 13 13:35:36] DEBUG[26331] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: SIP TIMER: Rescheduling retransmission #12930 (1) CANCEL - 14 [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #12930)) [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 0 [ 21]: SIP/2.0 200 canceling [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;branch=z9hG4bK71896ec5;rport=5060 [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 3 [ 65]: To: ;tag=373b8885156114ecb2c4bd665f9faf0b-f9aa [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 6 [ 45]: Server: OpenSER (1.3.0-pre1-tls (i386/linux)) [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: That's odd... Got a response on a call we dont know about. Callid 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:37] DEBUG[26331] chan_sip.c: Invalid SIP message - rejected , no callid, len 350 [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: SIP TIMER: Rescheduling retransmission #12930 (2) CANCEL - 14 [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: ** SIP timers: Rescheduling retransmission 3 to 2000 ms (t1 500 ms (Retrans id #12930)) [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 0 [ 21]: SIP/2.0 200 canceling [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;branch=z9hG4bK71896ec5;rport=5060 [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 3 [ 65]: To: ;tag=373b8885156114ecb2c4bd665f9faf0b-f9aa [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 6 [ 45]: Server: OpenSER (1.3.0-pre1-tls (i386/linux)) [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: That's odd... Got a response on a call we dont know about. Callid 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:38] DEBUG[26331] chan_sip.c: Invalid SIP message - rejected , no callid, len 350 [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: SIP TIMER: Rescheduling retransmission #12930 (3) CANCEL - 14 [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: ** SIP timers: Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id #12930)) [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 0 [ 21]: SIP/2.0 200 canceling [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.95.0.111:5060;branch=z9hG4bK71896ec5;rport=5060 [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 2 [ 58]: From: "asterisk" ;tag=as5c87f2dd [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 3 [ 65]: To: ;tag=373b8885156114ecb2c4bd665f9faf0b-f9aa [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 4 [ 53]: Call-ID: 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 5 [ 16]: CSeq: 102 CANCEL [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 6 [ 45]: Server: OpenSER (1.3.0-pre1-tls (i386/linux)) [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: That's odd... Got a response on a call we dont know about. Callid 7f10b486661309627b68b4f971c1cd27@88.95.0.111 [Dec 13 13:35:40] DEBUG[26331] chan_sip.c: Invalid SIP message - rejected , no callid, len 350 [Dec 13 13:35:44] DEBUG[26331] chan_sip.c: SIP TIMER: Rescheduling retransmission #12930 (4) CANCEL - 14 [Dec 13 13:35:44] DEBUG[26331] chan_sip.c: ** SIP timers: Rescheduling retransmission 5 to 4000 ms (t1 500 ms (Retrans id #12930)) [Dec 13 13:35:48] DEBUG[26331] chan_sip.c: SIP TIMER: Rescheduling retransmission #12930 (5) CANCEL - 14 [Dec 13 13:35:48] DEBUG[26331] chan_sip.c: ** SIP timers: Rescheduling retransmission 6 to 4000 ms (t1 500 ms (Retrans id #12930)) [Dec 13 13:35:52] DEBUG[26331] chan_sip.c: SIP TIMER: Rescheduling retransmission #12930 (6) CANCEL - 14 [Dec 13 13:35:52] DEBUG[26331] chan_sip.c: ** SIP timers: Rescheduling retransmission 7 to 4000 ms (t1 500 ms (Retrans id #12930))