[Jul 12 12:26:56] VERBOSE[12452] config.c: == Parsing '/etc/asterisk/logger.conf': [Jul 12 12:26:56] DEBUG[12452] config.c: Parsing /etc/asterisk/logger.conf [Jul 12 12:26:56] VERBOSE[12452] config.c: == Found [Jul 12 12:26:56] VERBOSE[12452] logger.c: Asterisk Event Logger restarted [Jul 12 12:26:56] VERBOSE[12452] logger.c: Asterisk Queue Logger restarted [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> REGISTER sip:isoemo.com SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK19780302732688623031;rport From: "Valery Komarov" ;tag=9523946 To: "Valery Komarov" Call-ID: 256983013825581-191182952428353@192.168.5.13 CSeq: 3 REGISTER Contact: Max-Forwards: 70 Expires: 60 Supported: path User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 0 [ 31]: REGISTER sip:isoemo.com SIP/2.0 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 1 [ 75]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK19780302732688623031;rport [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 2 [ 55]: From: "Valery Komarov" ;tag=9523946 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 3 [ 41]: To: "Valery Komarov" [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 4 [ 53]: Call-ID: 256983013825581-191182952428353@192.168.5.13 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 5 [ 16]: CSeq: 3 REGISTER [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 8 [ 11]: Expires: 60 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 9 [ 15]: Supported: path [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 10 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 12 [ 0]: [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: --- (12 headers 0 lines) --- [Jul 12 12:27:00] DEBUG[12485] acl.c: Found IP address for this socket [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.18:5060 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Allocating new SIP dialog for 256983013825581-191182952428353@192.168.5.13 - REGISTER (No RTP) [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Initializing initreq for method REGISTER - callid 256983013825581-191182952428353@192.168.5.13 [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: Sending to 192.168.5.13 : 5060 (NAT) [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Allocating new SIP dialog for 362648567eac1ac3205c48366563a390@127.0.0.1 - OPTIONS (No RTP) [Jul 12 12:27:00] DEBUG[12485] acl.c: Found IP address for this socket [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.18:5060 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Initializing initreq for method OPTIONS - callid 32e1deb01a96e8e02f0197f1007c323c@192.168.1.18 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 0 [ 41]: OPTIONS sip:360@192.168.5.13:5060 SIP/2.0 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK27b106b9;rport [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 3 [ 59]: From: "asterisk" ;tag=as6dbbd922 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 4 [ 31]: To: [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 5 [ 36]: Contact: [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 6 [ 54]: Call-ID: 32e1deb01a96e8e02f0197f1007c323c@192.168.1.18 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 7 [ 17]: CSeq: 102 OPTIONS [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 9 [ 35]: Date: Mon, 12 Jul 2010 08:27:00 GMT [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 12 [ 17]: Content-Length: 0 [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.13:5060: OPTIONS sip:360@192.168.5.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK27b106b9;rport Max-Forwards: 70 From: "asterisk" ;tag=as6dbbd922 To: Contact: Call-ID: 32e1deb01a96e8e02f0197f1007c323c@192.168.1.18 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.9 Date: Mon, 12 Jul 2010 08:27:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #18 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: > Saved useragent "Voip Phone 1.0" for peer 360 [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK19780302732688623031;received=192.168.5.13;rport=5060 From: "Valery Komarov" ;tag=9523946 To: "Valery Komarov" ;tag=as507d674a Call-ID: 256983013825581-191182952428353@192.168.5.13 CSeq: 3 REGISTER Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 60 Contact: ;expires=60 Date: Mon, 12 Jul 2010 08:27:00 GMT Content-Length: 0 <------------> [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:00] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 12 12:27:00] DEBUG[12457] chan_sip.c: Checking device state for peer 360 [Jul 12 12:27:00] DEBUG[12457] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 12 12:27:00] DEBUG[12457] devicestate.c: device 'SIP/360' state '1' [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: Scheduling destruction of SIP dialog '256983013825581-191182952428353@192.168.5.13' in 32000 ms (Method: REGISTER) [Jul 12 12:27:00] DEBUG[12486] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK27b106b9;rport From: "asterisk" ;tag=as6dbbd922 To: Call-ID: 32e1deb01a96e8e02f0197f1007c323c@192.168.1.18 CSeq: 102 OPTIONS Contact: Supported: 100rel, replaces, timer Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Accept: application/sdp, message/sipfrag, application/dtmf-relay Content-Length: 0 <-------------> [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK27b106b9;rport [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 2 [ 59]: From: "asterisk" ;tag=as6dbbd922 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 3 [ 31]: To: [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 32e1deb01a96e8e02f0197f1007c323c@192.168.1.18 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 5 [ 17]: CSeq: 102 OPTIONS [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 7 [ 34]: Supported: 100rel, replaces, timer [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 8 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 9 [ 64]: Accept: application/sdp, message/sipfrag, application/dtmf-relay [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Header 11 [ 0]: [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: --- (11 headers 0 lines) --- [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #18 [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Stopping retransmission on '32e1deb01a96e8e02f0197f1007c323c@192.168.1.18' of Request 102: Match Found [Jul 12 12:27:00] DEBUG[12485] chan_sip.c: Destroying SIP dialog 32e1deb01a96e8e02f0197f1007c323c@192.168.1.18 [Jul 12 12:27:00] VERBOSE[12485] chan_sip.c: Really destroying SIP dialog '32e1deb01a96e8e02f0197f1007c323c@192.168.1.18' Method: OPTIONS [Jul 12 12:27:03] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.1.41:9630 ---> <-------------> [Jul 12 12:27:03] DEBUG[12485] chan_sip.c: Header 0 [ 0]: [Jul 12 12:27:03] DEBUG[12485] chan_sip.c: Body 0 [ 0]: [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.1.41:9630 ---> INVITE sip:360@isoemo.com;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-30682c3d3d29ab4d-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=9afad9c2 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 327 v=0 o=- 12923396833605045 12923396833605045 IN IP4 192.168.1.41 s=Counterpath Bria 3.0 c=IN IP4 192.168.1.41 t=0 0 m=audio 49580 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.41 49580 typ host a=candidate:1 2 UDP 659134 192.168.1.41 49581 typ host <-------------> [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 0 [ 47]: INVITE sip:360@isoemo.com;transport=udp SIP/2.0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 1 [ 91]: Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-30682c3d3d29ab4d-1---d8754z-;rport [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 3 [ 50]: Contact: [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 4 [ 24]: To: [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 5 [ 39]: From: ;tag=9afad9c2 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 6 [ 53]: Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 7 [ 14]: CSeq: 1 INVITE [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 10 [ 48]: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 11 [ 19]: Content-Length: 327 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 12 [ 0]: [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 1 [ 59]: o=- 12923396833605045 12923396833605045 IN IP4 192.168.1.41 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 2 [ 22]: s=Counterpath Bria 3.0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.41 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 5 [ 29]: m=audio 49580 RTP/AVP 0 8 101 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 6 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 7 [ 15]: a=fmtp:101 0-15 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 8 [ 10]: a=sendrecv [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 9 [ 54]: a=candidate:1 1 UDP 659136 192.168.1.41 49580 typ host [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Body 10 [ 54]: a=candidate:1 2 UDP 659134 192.168.1.41 49581 typ host [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: --- (12 headers 11 lines) --- [Jul 12 12:27:08] DEBUG[12485] acl.c: Found IP address for this socket [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.18:5060 [Jul 12 12:27:08] VERBOSE[12485] netsock.c: == Using SIP RTP CoS mark 5 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Setting NAT on RTP to On [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Allocating new SIP dialog for NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. - INVITE (With RTP) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Sending to 192.168.1.41 : 9630 (NAT) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Initializing initreq for method INVITE - callid NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Using INVITE request as basis request - NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Found peer '361' for '361' from 192.168.1.41:9630 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Setting NAT on RTP to On [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing session-level SDP o=- 12923396833605045 12923396833605045 IN IP4 192.168.1.41... UNSUPPORTED. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing session-level SDP s=Counterpath Bria 3.0... UNSUPPORTED. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.1.41... OK. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Found RTP audio format 0 [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Found RTP audio format 8 [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Found RTP audio format 101 [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=candidate:1 1 UDP 659136 192.168.1.41 49580 typ host... UNSUPPORTED. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=candidate:1 2 UDP 659134 192.168.1.41 49581 typ host... UNSUPPORTED. [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Peer audio RTP is at port 192.168.1.41:49580 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Checking SIP call limits for device 361 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Updating call counter for incoming call [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: Looking for 360 in internal (domain isoemo.com) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: This channel will not be able to handle video. [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: build_route: Contact hop: [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: list_route: hop: [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: SIP/361-00000000: New call is still down.... Trying... [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.41:9630 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-30682c3d3d29ab4d-1---d8754z-;received=192.168.1.41;rport=9630 From: ;tag=9afad9c2 To: Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.41:9630 [Jul 12 12:27:08] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 12 12:27:08] DEBUG[12457] chan_sip.c: Checking device state for peer 361 [Jul 12 12:27:08] DEBUG[12457] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 12 12:27:08] DEBUG[12457] devicestate.c: device 'SIP/361' state '1' [Jul 12 12:27:08] DEBUG[12490] pbx.c: Launching 'Macro' [Jul 12 12:27:08] VERBOSE[12490] pbx.c: -- Executing [360@internal:1] Macro("SIP/361-00000000", "stdexten,360,SIP/360") in new stack [Jul 12 12:27:08] DEBUG[12490] pbx.c: Function result is '0' [Jul 12 12:27:08] DEBUG[12490] pbx.c: Expression result is '0' [Jul 12 12:27:08] DEBUG[12490] pbx.c: Launching 'GotoIf' [Jul 12 12:27:08] VERBOSE[12490] pbx.c: -- Executing [s@macro-stdexten:1] GotoIf("SIP/361-00000000", "0?3") in new stack [Jul 12 12:27:08] DEBUG[12490] pbx.c: Not taking any branch [Jul 12 12:27:08] DEBUG[12490] app_macro.c: Executed application: GotoIf [Jul 12 12:27:08] DEBUG[12490] pbx.c: Function result is '0' [Jul 12 12:27:08] DEBUG[12490] pbx.c: Expression result is '1' [Jul 12 12:27:08] DEBUG[12490] pbx.c: Launching 'GotoIf' [Jul 12 12:27:08] VERBOSE[12490] pbx.c: -- Executing [s@macro-stdexten:2] GotoIf("SIP/361-00000000", "1?4") in new stack [Jul 12 12:27:08] VERBOSE[12490] pbx.c: -- Goto (macro-stdexten,s,4) [Jul 12 12:27:08] DEBUG[12490] app_macro.c: Executed application: GotoIf [Jul 12 12:27:08] DEBUG[12490] pbx.c: Launching 'Dial' [Jul 12 12:27:08] VERBOSE[12490] pbx.c: -- Executing [s@macro-stdexten:4] Dial("SIP/361-00000000", "SIP/360,60,tTrRwW") in new stack [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jul 12 12:27:08] VERBOSE[12490] netsock.c: == Using SIP RTP CoS mark 5 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Allocating new SIP dialog for 2aa378e7215a8c8c26a4813843cd0aad@127.0.0.1 - INVITE (With RTP) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Setting NAT on RTP to On [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jul 12 12:27:08] DEBUG[12490] acl.c: Found IP address for this socket [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.18:5060 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: This channel will not be able to handle video. [Jul 12 12:27:08] DEBUG[12490] rtp.c: Seeded SDP of 'SIP/360-00000001' with that of 'SIP/361-00000000' [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable DIALEDTIME. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable ANSWEREDTIME. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable DIALEDPEERNAME. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable DIALEDPEERNUMBER. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable DIALSTATUS. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable MACRO_DEPTH. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable ARG2. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable ARG1. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable MACRO_PRIORITY. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable MACRO_CONTEXT. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable MACRO_EXTEN. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable SIPCALLID. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable SIPDOMAIN. [Jul 12 12:27:08] DEBUG[12490] channel.c: Not copying variable SIPURI. [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Outgoing Call for 360 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Updating call counter for outgoing call [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 12 12:27:08] VERBOSE[12490] chan_sip.c: Audio is at 192.168.1.18 port 15984 [Jul 12 12:27:08] VERBOSE[12490] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 12 12:27:08] VERBOSE[12490] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: -- Done with adding codecs to SDP [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Initializing initreq for method INVITE - callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 0 [ 40]: INVITE sip:360@192.168.5.13:5060 SIP/2.0 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 3 [ 49]: From: "361" ;tag=as41a472f1 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 4 [ 31]: To: [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 5 [ 31]: Contact: [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 6 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 8 [ 32]: User-Agent: Asterisk PBX 1.6.2.9 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 9 [ 35]: Date: Mon, 12 Jul 2010 08:27:08 GMT [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 13 [ 19]: Content-Length: 263 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Header 14 [ 0]: [Jul 12 12:27:08] DEBUG[12486] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 1 [ 48]: o=root 2035354530 2035354530 IN IP4 192.168.1.18 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.18 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 5 [ 27]: m=audio 15984 RTP/AVP 0 101 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 9 [ 25]: a=silenceSupp:off - - - - [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 10 [ 10]: a=ptime:20 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Body 11 [ 10]: a=sendrecv [Jul 12 12:27:08] VERBOSE[12490] chan_sip.c: Reliably Transmitting (NAT) to 192.168.5.13:5060: INVITE sip:360@192.168.5.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport Max-Forwards: 70 From: "361" ;tag=as41a472f1 To: Contact: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9 Date: Mon, 12 Jul 2010 08:27:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 2035354530 2035354530 IN IP4 192.168.1.18 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.1.18 t=0 0 m=audio 15984 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #23 [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:08] VERBOSE[12490] app_dial.c: -- Called 360 [Jul 12 12:27:08] VERBOSE[12490] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.41:9630 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-30682c3d3d29ab4d-1---d8754z-;received=192.168.1.41;rport=9630 From: ;tag=9afad9c2 To: ;tag=as65be0566 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.1.41:9630 [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport From: "361" ;tag=as41a472f1 To: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 2 [ 49]: From: "361" ;tag=as41a472f1 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 3 [ 31]: To: [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 6 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 8 [ 0]: [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: --- (8 headers 0 lines) --- [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: *** SIP TIMER: Cancelling retransmission #23 - INVITE (got response) [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '53a1521f1efb40db568fcdc229402935@192.168.1.18' Request 102: Found [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: SIP response 100 to standard invite [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport From: "361" ;tag=as41a472f1 To: ;tag=150919305 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 102 INVITE Contact: Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 <-------------> [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 2 [ 49]: From: "361" ;tag=as41a472f1 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 3 [ 45]: To: ;tag=150919305 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 7 [ 22]: Server: Voip Phone 1.0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 8 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: Header 10 [ 0]: [Jul 12 12:27:08] VERBOSE[12485] chan_sip.c: --- (10 headers 0 lines) --- [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '53a1521f1efb40db568fcdc229402935@192.168.1.18' Request 102: Found [Jul 12 12:27:08] DEBUG[12485] chan_sip.c: SIP response 180 to standard invite [Jul 12 12:27:08] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 12 12:27:08] DEBUG[12457] chan_sip.c: Checking device state for peer 360 [Jul 12 12:27:08] DEBUG[12457] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 12 12:27:08] DEBUG[12457] devicestate.c: device 'SIP/360' state '1' [Jul 12 12:27:08] DEBUG[12486] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) [Jul 12 12:27:08] VERBOSE[12490] app_dial.c: -- SIP/360-00000001 is ringing [Jul 12 12:27:08] VERBOSE[12490] chan_sip.c: <--- Transmitting (NAT) to 192.168.1.41:9630 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-30682c3d3d29ab4d-1---d8754z-;received=192.168.1.41;rport=9630 From: ;tag=9afad9c2 To: ;tag=as65be0566 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:08] DEBUG[12490] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.1.41:9630 [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport From: "361" ;tag=as41a472f1 To: ;tag=150919305 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 102 INVITE Contact: Supported: 100rel, replaces, timer Server: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 213 v=0 o=360 20946106 28627838 IN IP4 192.168.5.13 s=A conversation c=IN IP4 192.168.5.13 t=0 0 m=audio 10042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK68612f8d;rport [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 2 [ 49]: From: "361" ;tag=as41a472f1 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 3 [ 45]: To: ;tag=150919305 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 7 [ 34]: Supported: 100rel, replaces, timer [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 8 [ 22]: Server: Voip Phone 1.0 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 9 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 11 [ 19]: Content-Length: 213 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 12 [ 0]: [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 1 [ 43]: o=360 20946106 28627838 IN IP4 192.168.5.13 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 2 [ 16]: s=A conversation [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.5.13 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 5 [ 27]: m=audio 10042 RTP/AVP 0 101 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: --- (12 headers 10 lines) --- [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Acked pending invite 102 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Stopping retransmission on '53a1521f1efb40db568fcdc229402935@192.168.1.18' of Request 102: Match Found [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: SIP response 200 to standard invite [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing session-level SDP o=360 20946106 28627838 IN IP4 192.168.5.13... UNSUPPORTED. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.5.13... OK. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Found RTP audio format 0 [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Found RTP audio format 101 [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Peer audio RTP is at port 192.168.5.13:10042 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: We have an owner, now see if we need to change this call [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Updating call counter for outgoing call [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: build_route: Contact hop: [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: list_route: hop: [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Strict routing enforced for session 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: set_destination: set destination to 192.168.5.13, port 5060 [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: Transmitting (NAT) to 192.168.5.13:5060: ACK sip:360@192.168.5.13:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK44e4eb70;rport Max-Forwards: 70 From: "361" ;tag=as41a472f1 To: ;tag=150919305 Contact: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9 Content-Length: 0 --- [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Trying to put 'ACK sip:360' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:09] VERBOSE[12490] app_dial.c: -- SIP/360-00000001 answered SIP/361-00000000 [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: SIP answering channel: SIP/361-00000000 [Jul 12 12:27:09] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: Setting framing from config on incoming call [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 12 12:27:09] VERBOSE[12490] chan_sip.c: Audio is at 192.168.1.18 port 12408 [Jul 12 12:27:09] VERBOSE[12490] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 12 12:27:09] VERBOSE[12490] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: -- Done with adding codecs to SDP [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 12 12:27:09] VERBOSE[12490] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.1.41:9630 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-30682c3d3d29ab4d-1---d8754z-;received=192.168.1.41;rport=9630 From: ;tag=9afad9c2 To: ;tag=as65be0566 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 1424650008 1424650008 IN IP4 192.168.1.18 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.1.18 t=0 0 m=audio 12408 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #27 [Jul 12 12:27:09] DEBUG[12490] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.41:9630 [Jul 12 12:27:09] DEBUG[12490] features.c: bridge answer set, chan answer set [Jul 12 12:27:09] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:09] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:09] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 12 12:27:09] DEBUG[12457] chan_sip.c: Checking device state for peer 360 [Jul 12 12:27:09] DEBUG[12457] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 12 12:27:09] DEBUG[12457] devicestate.c: device 'SIP/360' state '1' [Jul 12 12:27:09] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 12 12:27:09] DEBUG[12457] chan_sip.c: Checking device state for peer 361 [Jul 12 12:27:09] DEBUG[12457] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 12 12:27:09] DEBUG[12457] devicestate.c: device 'SIP/361' state '1' [Jul 12 12:27:09] DEBUG[12486] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) [Jul 12 12:27:09] DEBUG[12486] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) [Jul 12 12:27:09] DEBUG[12490] rtp.c: Got RTCP report of 132 bytes [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.1.41:9630 ---> ACK sip:360@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-cf146f7fb50ca47c-1---d8754z-;rport Max-Forwards: 70 Contact: To: ;tag=as65be0566 From: ;tag=9afad9c2 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 1 ACK User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 0 <-------------> [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 0 [ 32]: ACK sip:360@192.168.1.18 SIP/2.0 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 1 [ 91]: Via: SIP/2.0/UDP 192.168.1.41:9630;branch=z9hG4bK-d8754z-cf146f7fb50ca47c-1---d8754z-;rport [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 3 [ 50]: Contact: [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 4 [ 39]: To: ;tag=as65be0566 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 5 [ 39]: From: ;tag=9afad9c2 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 6 [ 53]: Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 7 [ 11]: CSeq: 1 ACK [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 8 [ 48]: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Header 10 [ 0]: [Jul 12 12:27:09] VERBOSE[12485] chan_sip.c: --- (10 headers 0 lines) --- [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #27 [Jul 12 12:27:09] DEBUG[12485] chan_sip.c: Stopping retransmission on 'NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc.' of Response 1: Match Found [Jul 12 12:27:09] DEBUG[12490] rtp.c: Ooh, format changed from unknown to ulaw [Jul 12 12:27:09] DEBUG[12490] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jul 12 12:27:10] DEBUG[12490] rtp.c: Ooh, format changed from unknown to ulaw [Jul 12 12:27:10] DEBUG[12490] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> INVITE sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK425142861259928793;rport From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 1 INVITE Contact: Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 208 v=0 o=360 20946106 28627839 IN IP4 192.168.5.13 s=A conversation c=IN IP4 0.0.0.0 t=0 0 m=audio 10042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 0 [ 35]: INVITE sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK425142861259928793;rport [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 5 [ 14]: CSeq: 1 INVITE [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 8 [ 31]: Supported: replaces, join, path [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 9 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 10 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 12 [ 19]: Content-Length: 208 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 13 [ 0]: [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 1 [ 43]: o=360 20946106 28627839 IN IP4 192.168.5.13 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 2 [ 16]: s=A conversation [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 3 [ 16]: c=IN IP4 0.0.0.0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 5 [ 27]: m=audio 10042 RTP/AVP 0 101 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Body 9 [ 10]: a=sendonly [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: --- (13 headers 10 lines) --- [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Begin: parsing SIP "Supported: replaces, join, path" [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Found SIP option: -replaces- [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Matched SIP option: replaces [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Found SIP option: -join- [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Matched SIP option: join [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Found SIP option: -path- [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Matched SIP option: path [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Sending to 192.168.5.13 : 5060 (NAT) [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Initializing initreq for method INVITE - callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing session-level SDP o=360 20946106 28627839 IN IP4 192.168.5.13... UNSUPPORTED. [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing session-level SDP c=IN IP4 0.0.0.0... OK. [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Found RTP audio format 0 [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Found RTP audio format 101 [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Found audio description format PCMU for ID 0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Found audio description format telephone-event for ID 101 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED. [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Processing media-level (audio) SDP a=sendonly... OK. [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Peer audio RTP is at port 0.0.0.0:10042 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: We have an owner, now see if we need to change this call [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Got a SIP re-invite for call 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: SIP/360-00000001: This call is UP.... [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK425142861259928793;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Setting framing from config on incoming call [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Audio is at 192.168.1.18 port 15984 [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: -- Done with adding codecs to SDP [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK425142861259928793;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 1 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2035354530 2035354531 IN IP4 192.168.1.18 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.1.18 t=0 0 m=audio 15984 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #30 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:10] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:10] VERBOSE[12490] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/361-00000000 [Jul 12 12:27:10] DEBUG[12490] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 12 12:27:10] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:10] DEBUG[12490] rtp.c: RTP NAT: Got audio from other end. Now sending to address 192.168.5.13:10042 [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> ACK sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK27145178501935522358 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 1 ACK Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 0 [ 32]: ACK sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK27145178501935522358 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 5 [ 11]: CSeq: 1 ACK [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 7 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Header 9 [ 0]: [Jul 12 12:27:10] VERBOSE[12485] chan_sip.c: --- (9 headers 0 lines) --- [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #30 [Jul 12 12:27:10] DEBUG[12485] chan_sip.c: Stopping retransmission on '53a1521f1efb40db568fcdc229402935@192.168.1.18' of Response 1: Match Found [Jul 12 12:27:10] DEBUG[12490] channel.c: Generator got voice, switching to phase locked mode [Jul 12 12:27:10] DEBUG[12490] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 12 12:27:10] DEBUG[12490] channel.c: Set channel SIP/361-00000000 to write format slin [Jul 12 12:27:10] DEBUG[12490] res_musiconhold.c: SIP/361-00000000 Opened file 0 '/var/lib/asterisk/moh/08-Chet Atkins-Blue Angel' [Jul 12 12:27:10] WARNING[12490] mp3/interface.c: Junk at the beginning of frame 49443303 [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> INVITE sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK587330182108036952;rport From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 2 INVITE Contact: Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 215 v=0 o=360 123456 654322 IN IP4 192.168.5.13 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.5.13 t=0 0 m=audio 10042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 0 [ 35]: INVITE sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK587330182108036952;rport [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 5 [ 14]: CSeq: 2 INVITE [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 8 [ 31]: Supported: replaces, join, path [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 9 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 10 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 12 [ 19]: Content-Length: 215 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 13 [ 0]: [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 1 [ 39]: o=360 123456 654322 IN IP4 192.168.5.13 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.5.13 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 5 [ 27]: m=audio 10042 RTP/AVP 0 101 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: --- (13 headers 10 lines) --- [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: Sending to 192.168.5.13 : 5060 (NAT) [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Initializing initreq for method INVITE - callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Call 53a1521f1efb40db568fcdc229402935@192.168.1.18 responded to our reinvite without changing SDP version; ignoring SDP. [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Got a SIP re-invite for call 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: SIP/360-00000001: This call is UP.... [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK587330182108036952;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Setting framing from config on incoming call [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: Audio is at 192.168.1.18 port 15984 [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: -- Done with adding codecs to SDP [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK587330182108036952;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 2 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2035354530 2035354531 IN IP4 192.168.1.18 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.1.18 t=0 0 m=audio 15984 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #32 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:11] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> ACK sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK1380358150671276 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 2 ACK Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 0 [ 32]: ACK sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK1380358150671276 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 5 [ 11]: CSeq: 2 ACK [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 7 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Header 9 [ 0]: [Jul 12 12:27:11] VERBOSE[12485] chan_sip.c: --- (9 headers 0 lines) --- [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #32 [Jul 12 12:27:11] DEBUG[12485] chan_sip.c: Stopping retransmission on '53a1521f1efb40db568fcdc229402935@192.168.1.18' of Response 2: Match Found [Jul 12 12:27:11] DEBUG[12490] rtp.c: Forcing Marker bit, because SSRC has changed [Jul 12 12:27:11] DEBUG[12490] rtp.c: Changing ssrc from 105139232 to 1291587583 due to a source change [Jul 12 12:27:12] DEBUG[12490] rtp.c: Got RTCP report of 200 bytes [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> INVITE sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK2241117588228227032;rport From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 3 INVITE Contact: Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 210 v=0 o=360 123456 654322 IN IP4 192.168.5.13 s=Asterisk PBX 1.6.2.9 c=IN IP4 0.0.0.0 t=0 0 m=audio 10042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 0 [ 35]: INVITE sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 1 [ 74]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK2241117588228227032;rport [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 5 [ 14]: CSeq: 3 INVITE [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 8 [ 31]: Supported: replaces, join, path [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 9 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 10 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 12 [ 19]: Content-Length: 210 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 13 [ 0]: [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 1 [ 39]: o=360 123456 654322 IN IP4 192.168.5.13 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 3 [ 16]: c=IN IP4 0.0.0.0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 5 [ 27]: m=audio 10042 RTP/AVP 0 101 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Body 9 [ 10]: a=sendonly [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: --- (13 headers 10 lines) --- [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: Sending to 192.168.5.13 : 5060 (NAT) [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Initializing initreq for method INVITE - callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Call 53a1521f1efb40db568fcdc229402935@192.168.1.18 responded to our reinvite without changing SDP version; ignoring SDP. [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Got a SIP re-invite for call 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: SIP/360-00000001: This call is UP.... [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK2241117588228227032;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 3 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Setting framing from config on incoming call [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: Audio is at 192.168.1.18 port 15984 [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: -- Done with adding codecs to SDP [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK2241117588228227032;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 3 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2035354530 2035354531 IN IP4 192.168.1.18 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.1.18 t=0 0 m=audio 15984 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #33 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:13] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> ACK sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK87422702628316690 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 3 ACK Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 0 [ 32]: ACK sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK87422702628316690 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 5 [ 11]: CSeq: 3 ACK [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 7 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Header 9 [ 0]: [Jul 12 12:27:13] VERBOSE[12485] chan_sip.c: --- (9 headers 0 lines) --- [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #33 [Jul 12 12:27:13] DEBUG[12485] chan_sip.c: Stopping retransmission on '53a1521f1efb40db568fcdc229402935@192.168.1.18' of Response 3: Match Found [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> INVITE sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK116689523497412225;rport From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 4 INVITE Contact: Max-Forwards: 70 Supported: replaces, join, path User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 215 v=0 o=360 123456 654322 IN IP4 192.168.5.13 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.5.13 t=0 0 m=audio 10042 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 0 [ 35]: INVITE sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 1 [ 73]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK116689523497412225;rport [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 5 [ 14]: CSeq: 4 INVITE [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 6 [ 36]: Contact: [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 8 [ 31]: Supported: replaces, join, path [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 9 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 10 [ 85]: Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 11 [ 29]: Content-Type: application/sdp [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 12 [ 19]: Content-Length: 215 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 13 [ 0]: [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 0 [ 3]: v=0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 1 [ 39]: o=360 123456 654322 IN IP4 192.168.5.13 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 2 [ 22]: s=Asterisk PBX 1.6.2.9 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.5.13 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 4 [ 5]: t=0 0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 5 [ 27]: m=audio 10042 RTP/AVP 0 101 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-15 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: --- (13 headers 10 lines) --- [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: Sending to 192.168.5.13 : 5060 (NAT) [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Initializing initreq for method INVITE - callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Call 53a1521f1efb40db568fcdc229402935@192.168.1.18 responded to our reinvite without changing SDP version; ignoring SDP. [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Got a SIP re-invite for call 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: SIP/360-00000001: This call is UP.... [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK116689523497412225;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 4 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Setting framing from config on incoming call [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: Audio is at 192.168.1.18 port 15984 [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: -- Done with adding codecs to SDP [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK116689523497412225;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 4 INVITE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 263 v=0 o=root 2035354530 2035354531 IN IP4 192.168.1.18 s=Asterisk PBX 1.6.2.9 c=IN IP4 192.168.1.18 t=0 0 m=audio 15984 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #34 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:14] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> ACK sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK140151792737454564 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 4 ACK Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 0 [ 32]: ACK sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK140151792737454564 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 5 [ 11]: CSeq: 4 ACK [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 7 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Header 9 [ 0]: [Jul 12 12:27:14] VERBOSE[12485] chan_sip.c: --- (9 headers 0 lines) --- [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #34 [Jul 12 12:27:14] DEBUG[12485] chan_sip.c: Stopping retransmission on '53a1521f1efb40db568fcdc229402935@192.168.1.18' of Response 4: Match Found [Jul 12 12:27:14] DEBUG[12490] rtp.c: Forcing Marker bit, because SSRC has changed [Jul 12 12:27:14] DEBUG[12490] rtp.c: Changing ssrc from 1291587583 to 738106436 due to a source change [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 12 12:27:15] DTMF[12490] channel.c: DTMF begin '1' received on SIP/360-00000001 [Jul 12 12:27:15] DTMF[12490] channel.c: DTMF begin passthrough '1' on SIP/360-00000001 [Jul 12 12:27:15] DEBUG[12490] channel.c: Got DTMF begin on channel (SIP/360-00000001) [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] channel.c: Bridge stops bridging channels SIP/361-00000000 and SIP/360-00000001 [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12490] rtp.c: Sending dtmf: 49 (1), at 192.168.5.13 [Jul 12 12:27:15] DTMF[12490] channel.c: DTMF end '1' received on SIP/360-00000001, duration 110 ms [Jul 12 12:27:15] DTMF[12490] channel.c: DTMF end accepted with begin '1' on SIP/360-00000001 [Jul 12 12:27:15] DTMF[12490] channel.c: DTMF end passthrough '1' on SIP/360-00000001 [Jul 12 12:27:15] DEBUG[12490] channel.c: Got DTMF end on channel (SIP/360-00000001) [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] channel.c: Bridge stops bridging channels SIP/361-00000000 and SIP/360-00000001 [Jul 12 12:27:15] DEBUG[12490] features.c: Feature interpret: chan=SIP/361-00000000, peer=SIP/360-00000001, code=1, sense=2, features=18, dynamic=monkeys#automon#nway-start#monkeys#automon#nway-start [Jul 12 12:27:15] VERBOSE[12490] res_musiconhold.c: -- Stopped music on hold on SIP/361-00000000 [Jul 12 12:27:15] DEBUG[12490] channel.c: Set channel SIP/361-00000000 to write format ulaw [Jul 12 12:27:15] DEBUG[12490] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Jul 12 12:27:15] DEBUG[12490] channel.c: Started silence generator on 'SIP/361-00000000' [Jul 12 12:27:15] DEBUG[12490] channel.c: Generator got voice, switching to phase locked mode [Jul 12 12:27:15] DEBUG[12490] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] DEBUG[12491] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] DEBUG[12491] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] WARNING[12490] chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) [Jul 12 12:27:15] DEBUG[12490] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Jul 12 12:27:15] DEBUG[12490] channel.c: Stopped silence generator on 'SIP/361-00000000' [Jul 12 12:27:15] DEBUG[12490] channel.c: Set channel SIP/361-00000000 to write format slin [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:15] DEBUG[12490] channel.c: Set channel SIP/361-00000000 to write format ulaw [Jul 12 12:27:15] DEBUG[12490] rtp.c: Got RTCP report of 224 bytes [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.5.13:5060 ---> BYE sip:361@192.168.1.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK3151011175313947984;rport From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 5 BYE Max-Forwards: 70 User-Agent: Voip Phone 1.0 Content-Length: 0 <-------------> [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 0 [ 32]: BYE sip:361@192.168.1.18 SIP/2.0 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 1 [ 74]: Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK3151011175313947984;rport [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 2 [ 47]: From: ;tag=150919305 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 3 [ 47]: To: "361" ;tag=as41a472f1 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 4 [ 54]: Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 5 [ 11]: CSeq: 5 BYE [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 7 [ 26]: User-Agent: Voip Phone 1.0 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 9 [ 0]: [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: --- (9 headers 0 lines) --- [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Initializing initreq for method BYE - callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: Sending to 192.168.5.13 : 5060 (NAT) [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Setting SIP_ALREADYGONE on dialog 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Received bye, issuing owner hangup [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: <--- Transmitting (NAT) to 192.168.5.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.13:5060;branch=z9hG4bK3151011175313947984;received=192.168.5.13;rport=5060 From: ;tag=150919305 To: "361" ;tag=as41a472f1 Call-ID: 53a1521f1efb40db568fcdc229402935@192.168.1.18 CSeq: 5 BYE Server: Asterisk PBX 1.6.2.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.5.13:5060 [Jul 12 12:27:16] DEBUG[12490] channel.c: Didn't get a frame from channel: SIP/360-00000001 [Jul 12 12:27:16] DEBUG[12490] rtp.c: Setting the marker bit due to a source update [Jul 12 12:27:16] DEBUG[12490] channel.c: Bridge stops bridging channels SIP/361-00000000 and SIP/360-00000001 [Jul 12 12:27:16] DEBUG[12490] channel.c: Hanging up channel 'SIP/360-00000001' [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: Hangup call SIP/360-00000001, SIP callid 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: update_call_counter(360) - decrement call limit counter on hangup [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: Updating call counter for outgoing call [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: Call to peer '360' removed from call limit 0 [Jul 12 12:27:16] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 12 12:27:16] DEBUG[12490] rtp.c: Channel '' has no RTP, not doing anything [Jul 12 12:27:16] DEBUG[12490] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jul 12 12:27:16] DEBUG[12457] chan_sip.c: Checking device state for peer 360 [Jul 12 12:27:16] DEBUG[12490] app_macro.c: Spawn extension (macro-stdexten,s,4) exited non-zero on 'SIP/361-00000000' in macro 'stdexten' [Jul 12 12:27:16] VERBOSE[12490] app_macro.c: == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'SIP/361-00000000' in macro 'stdexten' [Jul 12 12:27:16] DEBUG[12490] pbx.c: Spawn extension (internal,360,1) exited non-zero on 'SIP/361-00000000' [Jul 12 12:27:16] DEBUG[12457] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 12 12:27:16] DEBUG[12457] devicestate.c: device 'SIP/360' state '1' [Jul 12 12:27:16] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 12 12:27:16] DEBUG[12457] chan_sip.c: Checking device state for peer 360 [Jul 12 12:27:16] DEBUG[12457] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 12 12:27:16] VERBOSE[12490] pbx.c: == Spawn extension (internal, 360, 1) exited non-zero on 'SIP/361-00000000' [Jul 12 12:27:16] DEBUG[12490] channel.c: Soft-Hanging up channel 'SIP/361-00000000' [Jul 12 12:27:16] DEBUG[12490] channel.c: Hanging up channel 'SIP/361-00000000' [Jul 12 12:27:16] DEBUG[12486] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: Hangup call SIP/361-00000000, SIP callid NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:16] DEBUG[12457] devicestate.c: device 'SIP/360' state '1' [Jul 12 12:27:16] VERBOSE[12490] chan_sip.c: Scheduling destruction of SIP dialog 'NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc.' in 6400 ms (Method: ACK) [Jul 12 12:27:16] DEBUG[12486] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: Strict routing enforced for session NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:16] VERBOSE[12490] chan_sip.c: set_destination: Parsing for address/port to send to [Jul 12 12:27:16] VERBOSE[12490] chan_sip.c: set_destination: set destination to 192.168.1.41, port 9630 [Jul 12 12:27:16] VERBOSE[12490] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.41:9630: BYE sip:361@192.168.1.41:9630;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK3eaebf46;rport Max-Forwards: 70 From: ;tag=as65be0566 To: ;tag=9afad9c2 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.2.9 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #36 [Jul 12 12:27:16] DEBUG[12490] chan_sip.c: Trying to put 'BYE sip:361' onto UDP socket destined for 192.168.1.41:9630 [Jul 12 12:27:16] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 361 [Jul 12 12:27:16] DEBUG[12457] chan_sip.c: Checking device state for peer 361 [Jul 12 12:27:16] DEBUG[12457] devicestate.c: Changing state for SIP/361 - state 1 (Not in use) [Jul 12 12:27:16] DEBUG[12457] devicestate.c: device 'SIP/361' state '1' [Jul 12 12:27:16] DEBUG[12486] app_queue.c: Device 'SIP/361' changed to state '1' (Not in use) [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: <--- SIP read from UDP:192.168.1.41:9630 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK3eaebf46;rport=5060 Contact: To: ;tag=9afad9c2 From: ;tag=as65be0566 Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. CSeq: 102 BYE User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 Content-Length: 0 <-------------> [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 1 [ 68]: Via: SIP/2.0/UDP 192.168.1.18:5060;branch=z9hG4bK3eaebf46;rport=5060 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 2 [ 50]: Contact: [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 3 [ 37]: To: ;tag=9afad9c2 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 4 [ 41]: From: ;tag=as65be0566 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 5 [ 53]: Call-ID: NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 6 [ 13]: CSeq: 102 BYE [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 7 [ 48]: User-Agent: Bria 3.0 release 3.0.1.1 stamp 56993 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Header 9 [ 0]: [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: --- (9 headers 0 lines) --- [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #36 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Stopping retransmission on 'NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc.' of Request 102: Match Found [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Destroying SIP dialog 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: Really destroying SIP dialog '53a1521f1efb40db568fcdc229402935@192.168.1.18' Method: BYE [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Updating call counter for outgoing call [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Call to peer '360' removed from call limit 0 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: This call did not properly clean up call limits. Call ID 53a1521f1efb40db568fcdc229402935@192.168.1.18 [Jul 12 12:27:16] DEBUG[12485] chan_sip.c: Destroying SIP dialog NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc. [Jul 12 12:27:16] VERBOSE[12485] chan_sip.c: Really destroying SIP dialog 'NDM4Y2Y2MjhkMGI2Yjk3NDFiOGJkYTYwMWEyYmQ2Nzc.' Method: ACK [Jul 12 12:27:16] DEBUG[12457] devicestate.c: No provider found, checking channel drivers for SIP - 360 [Jul 12 12:27:16] DEBUG[12457] chan_sip.c: Checking device state for peer 360 [Jul 12 12:27:16] DEBUG[12457] devicestate.c: Changing state for SIP/360 - state 1 (Not in use) [Jul 12 12:27:16] DEBUG[12457] devicestate.c: device 'SIP/360' state '1' [Jul 12 12:27:16] DEBUG[12486] app_queue.c: Device 'SIP/360' changed to state '1' (Not in use) [Jul 12 12:27:18] VERBOSE[12489] asterisk.c: Beginning asterisk shutdown.... [Jul 12 12:27:18] VERBOSE[12489] asterisk.c: Executing last minute cleanups [Jul 12 12:27:18] VERBOSE[12489] res_musiconhold.c: == Destroying musiconhold processes [Jul 12 12:27:18] DEBUG[12489] res_musiconhold.c: Destroying MOH class 'default' [Jul 12 12:27:18] VERBOSE[12489] asterisk.c: Asterisk cleanly ending (0). [Jul 12 12:27:18] DEBUG[12489] asterisk.c: Asterisk ending (0).