[Jan 29 18:12:01] VERBOSE[2956] config.c: == Parsing '/etc/asterisk/logger.conf': [Jan 29 18:12:01] DEBUG[2956] config.c: Parsing /etc/asterisk/logger.conf [Jan 29 18:12:01] VERBOSE[2956] config.c: == Found [Jan 29 18:12:01] VERBOSE[2956] logger.c: Asterisk Event Logger restarted [Jan 29 18:12:01] VERBOSE[2956] logger.c: Asterisk Queue Logger restarted [Jan 29 18:12:04] DEBUG[2946] chan_sip.c: Auto destroying SIP dialog '466181098' [Jan 29 18:12:04] DEBUG[2946] chan_sip.c: Destroying SIP dialog 466181098 [Jan 29 18:12:04] VERBOSE[2946] chan_sip.c: Really destroying SIP dialog '466181098' Method: OPTIONS [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.103:5062 ---> INVITE sip:cit02@sax3n03.local:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1809766208 From: ;tag=2047668150 To: Call-ID: 380593170 CSeq: 20 INVITE Contact: Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Subject: Phone call Content-Length: 423 v=0 o=cit03 123456 654321 IN IP4 192.168.12.103 s=A conversation c=IN IP4 192.168.12.103 t=0 0 m=audio 7080 RTP/AVP 112 111 110 3 0 8 101 a=rtpmap:112 speex/32000/1 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000/1 a=fmtp:111 vbr=vad a=rtpmap:110 speex/8000/1 a=fmtp:110 vbr=vad a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 0 [ 43]: INVITE sip:cit02@sax3n03.local:5062 SIP/2.0 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1809766208 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 2 [ 51]: From: ;tag=2047668150 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 3 [ 34]: To: [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 4 [ 18]: Call-ID: 380593170 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 5 [ 15]: CSeq: 20 INVITE [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 6 [ 40]: Contact: [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 9 [ 16]: Max-Forwards: 70 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 10 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 11 [ 19]: Subject: Phone call [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 12 [ 21]: Content-Length: 423 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 13 [ 0]: [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 1 [ 43]: o=cit03 123456 654321 IN IP4 192.168.12.103 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.103 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 5 [ 42]: m=audio 7080 RTP/AVP 112 111 110 3 0 8 101 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 6 [ 26]: a=rtpmap:112 speex/32000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 7 [ 17]: a=fmtp:112 vbr=on [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 8 [ 26]: a=rtpmap:111 speex/16000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 9 [ 18]: a=fmtp:111 vbr=vad [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 10 [ 25]: a=rtpmap:110 speex/8000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 11 [ 18]: a=fmtp:110 vbr=vad [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 12 [ 21]: a=rtpmap:3 GSM/8000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 13 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 14 [ 22]: a=rtpmap:8 PCMA/8000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 15 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 16 [ 15]: a=fmtp:101 0-11 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Body 17 [ 10]: a=sendrecv [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: --- (13 headers 18 lines) --- [Jan 29 18:12:08] DEBUG[2946] acl.c: Found IP address for this socket [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.12.103:5060 [Jan 29 18:12:08] VERBOSE[2946] netsock.c: == Using SIP RTP CoS mark 5 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Setting NAT on RTP to Off [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Allocating new SIP dialog for 380593170 - INVITE (With RTP) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Sending to 192.168.12.103 : 5062 (no NAT) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Initializing initreq for method INVITE - callid 380593170 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Using INVITE request as basis request - 380593170 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found peer 'cit03' for 'cit03' from 192.168.12.103:5062 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Setting NAT on RTP to Off [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing session-level SDP o=cit03 123456 654321 IN IP4 192.168.12.103... UNSUPPORTED. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.12.103... OK. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 112 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 111 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 110 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 3 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 0 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 8 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found RTP audio format 101 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format speex for ID 112 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 speex/32000/1... OK. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=fmtp:112 vbr=on... UNSUPPORTED. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format speex for ID 111 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 speex/16000/1... OK. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=fmtp:111 vbr=vad... UNSUPPORTED. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format speex for ID 110 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000/1... OK. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 vbr=vad... UNSUPPORTED. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format GSM for ID 3 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000/1... OK. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000/1... OK. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000/1... OK. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0xa1e (gsm|ulaw|alaw|g726|speex|g726aal2)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Peer audio RTP is at port 192.168.12.103:7080 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Checking SIP call limits for device [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Updating call counter for incoming call [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: Looking for cit02 in sax3nlocal (domain sax3n03.local) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: This channel will not be able to handle video. [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: build_route: Contact hop: [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: list_route: hop: [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: SIP/cit03-00000002: New call is still down.... Trying... [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK1809766208;received=192.168.12.103;rport=5062 From: ;tag=2047668150 To: Call-ID: 380593170 CSeq: 20 INVITE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:12:08] DEBUG[2940] devicestate.c: No provider found, checking channel drivers for SIP - cit03 [Jan 29 18:12:08] DEBUG[2940] chan_sip.c: Checking device state for peer cit03 [Jan 29 18:12:08] DEBUG[2940] devicestate.c: Changing state for SIP/cit03 - state 1 (Not in use) [Jan 29 18:12:08] DEBUG[2940] devicestate.c: device 'SIP/cit03' state '1' [Jan 29 18:12:08] DEBUG[2960] pbx.c: Launching 'NoOp' [Jan 29 18:12:08] VERBOSE[2960] pbx.c: -- Executing [cit02@sax3nlocal:1] NoOp("SIP/cit03-00000002", "Call to cit02") in new stack [Jan 29 18:12:08] DEBUG[2960] pbx.c: Launching 'Dial' [Jan 29 18:12:08] VERBOSE[2960] pbx.c: -- Executing [cit02@sax3nlocal:2] Dial("SIP/cit03-00000002", "SIP/cit02") in new stack [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [Jan 29 18:12:08] VERBOSE[2960] netsock.c: == Using SIP RTP CoS mark 5 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Allocating new SIP dialog for 40b96bc9060b36ff5fb6847d1be27510@192.168.12.103 - INVITE (With RTP) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Setting NAT on RTP to Off [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: OBPROXY: Not applying OBproxy to this call [Jan 29 18:12:08] DEBUG[2960] acl.c: Found IP address for this socket [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.12.103:5060 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: *** Our capabilities are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: *** Our preferred formats from the incoming channel are 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: This channel will not be able to handle video. [Jan 29 18:12:08] DEBUG[2960] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Jan 29 18:12:08] DEBUG[2960] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable DIALEDTIME. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable ANSWEREDTIME. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable DIALEDPEERNAME. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable DIALEDPEERNUMBER. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable DIALSTATUS. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable SIPCALLID. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable SIPDOMAIN. [Jan 29 18:12:08] DEBUG[2960] channel.c: Not copying variable SIPURI. [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Outgoing Call for [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Updating call counter for outgoing call [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [Jan 29 18:12:08] VERBOSE[2960] chan_sip.c: Audio is at 192.168.12.103 port 16948 [Jan 29 18:12:08] VERBOSE[2960] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 29 18:12:08] VERBOSE[2960] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: -- Done with adding codecs to SDP [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Initializing initreq for method INVITE - callid 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 0 [ 37]: INVITE sip:sax3n02.local:5062 SIP/2.0 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 3 [ 55]: From: "cit03" ;tag=as4b49de71 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 4 [ 28]: To: [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 5 [ 35]: Contact: [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 6 [ 56]: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 8 [ 34]: User-Agent: Asterisk PBX 1.6.2.9-2 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 9 [ 35]: Date: Mon, 29 Jan 2007 18:12:08 GMT [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 10 [ 72]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 12 [ 29]: Content-Type: application/sdp [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 13 [ 19]: Content-Length: 242 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Header 14 [ 0]: [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 1 [ 50]: o=root 1542424413 1542424413 IN IP4 192.168.12.103 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 2 [ 24]: s=Asterisk PBX 1.6.2.9-2 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.103 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 5 [ 27]: m=audio 16948 RTP/AVP 0 101 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 6 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 7 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-16 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 9 [ 10]: a=ptime:20 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Body 10 [ 10]: a=sendrecv [Jan 29 18:12:08] VERBOSE[2960] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.12.102:5062: INVITE sip:sax3n02.local:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport Max-Forwards: 70 From: "cit03" ;tag=as4b49de71 To: Contact: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.9-2 Date: Mon, 29 Jan 2007 18:12:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 242 v=0 o=root 1542424413 1542424413 IN IP4 192.168.12.103 s=Asterisk PBX 1.6.2.9-2 c=IN IP4 192.168.12.103 t=0 0 m=audio 16948 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #36 [Jan 29 18:12:08] DEBUG[2960] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:12:08] VERBOSE[2960] app_dial.c: -- Called cit02 [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 From: "cit03" ;tag=as4b49de71 To: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 102 INVITE User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as4b49de71 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 3 [ 28]: To: [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 4 [ 56]: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 6 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 8 [ 0]: [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: --- (8 headers 0 lines) --- [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: *** SIP TIMER: Cancelling retransmission #36 - INVITE (got response) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' Request 102: Found [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: SIP response 100 to standard invite [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 101 Dialog Establishement Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 From: "cit03" ;tag=as4b49de71 To: ;tag=718914218 Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 0 [ 33]: SIP/2.0 101 Dialog Establishement [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as4b49de71 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 3 [ 42]: To: ;tag=718914218 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 4 [ 56]: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 6 [ 34]: Contact: [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 7 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: Header 9 [ 0]: [Jan 29 18:12:08] VERBOSE[2946] chan_sip.c: --- (9 headers 0 lines) --- [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' Request 102: Found [Jan 29 18:12:08] DEBUG[2946] chan_sip.c: SIP response 101 to standard invite [Jan 29 18:12:08] DEBUG[2949] app_queue.c: Device 'SIP/cit03' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:12:09] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> OPTIONS sip:192.168.12.103:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK2066992218 Route: "cit03" ;tag=as4b49de71 From: ;tag=1952214518 To: "cit03" ;tag=as4b49de71 Call-ID: 463398228 CSeq: 20 OPTIONS Accept: application/sdp Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Expires: 120 Content-Length: 0 <-------------> [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 0 [ 39]: OPTIONS sip:192.168.12.103:5060 SIP/2.0 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.102:5062;rport;branch=z9hG4bK2066992218 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 2 [ 56]: Route: "cit03" ;tag=as4b49de71 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 3 [ 45]: From: ;tag=1952214518 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 4 [ 53]: To: "cit03" ;tag=as4b49de71 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 5 [ 18]: Call-ID: 463398228 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 6 [ 16]: CSeq: 20 OPTIONS [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 7 [ 23]: Accept: application/sdp [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 9 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 10 [ 12]: Expires: 120 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 11 [ 17]: Content-Length: 0 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Header 12 [ 0]: [Jan 29 18:12:09] VERBOSE[2946] chan_sip.c: --- (12 headers 0 lines) --- [Jan 29 18:12:09] DEBUG[2946] acl.c: Found IP address for this socket [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.12.103:5060 [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Allocating new SIP dialog for 463398228 - OPTIONS (No RTP) [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS [Jan 29 18:12:09] VERBOSE[2946] chan_sip.c: Looking for s in sax3nvoip (domain 192.168.12.103) [Jan 29 18:12:09] VERBOSE[2946] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.102:5062 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.12.102:5062;branch=z9hG4bK2066992218;received=192.168.12.102;rport=5062 From: ;tag=1952214518 To: "cit03" ;tag=as4b49de71 Call-ID: 463398228 CSeq: 20 OPTIONS Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:12:09] VERBOSE[2946] chan_sip.c: Scheduling destruction of SIP dialog '463398228' in 32000 ms (Method: OPTIONS) [Jan 29 18:12:09] DEBUG[2946] chan_sip.c: SIP message could not be handled, bad request: 463398228 [Jan 29 18:12:10] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 From: "cit03" ;tag=as4b49de71 To: ;tag=718914218 Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 102 INVITE Contact: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as4b49de71 [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 3 [ 42]: To: ;tag=718914218 [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 4 [ 56]: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 6 [ 34]: Contact: [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 7 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: Header 9 [ 0]: [Jan 29 18:12:10] VERBOSE[2946] chan_sip.c: --- (9 headers 0 lines) --- [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' Request 102: Found [Jan 29 18:12:10] DEBUG[2946] chan_sip.c: SIP response 180 to standard invite [Jan 29 18:12:10] DEBUG[2940] devicestate.c: No provider found, checking channel drivers for SIP - cit02 [Jan 29 18:12:10] DEBUG[2940] chan_sip.c: Checking device state for peer cit02 [Jan 29 18:12:10] DEBUG[2940] devicestate.c: Changing state for SIP/cit02 - state 1 (Not in use) [Jan 29 18:12:10] DEBUG[2940] devicestate.c: device 'SIP/cit02' state '1' [Jan 29 18:12:10] VERBOSE[2960] app_dial.c: -- SIP/cit02-00000003 is ringing [Jan 29 18:12:10] VERBOSE[2960] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK1809766208;received=192.168.12.103;rport=5062 From: ;tag=2047668150 To: ;tag=as539f0966 Call-ID: 380593170 CSeq: 20 INVITE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 29 18:12:10] DEBUG[2960] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:12:10] DEBUG[2949] app_queue.c: Device 'SIP/cit02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 From: "cit03" ;tag=as4b49de71 To: ;tag=718914218 Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 102 INVITE Contact: Content-Type: application/sdp User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 218 v=0 o=cit02 123456 654321 IN IP4 192.168.12.102 s=A conversation c=IN IP4 192.168.12.102 t=0 0 m=audio 7080 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-11 a=sendrecv <-------------> [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK418b975d;rport=5060 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as4b49de71 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 3 [ 42]: To: ;tag=718914218 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 4 [ 56]: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 5 [ 16]: CSeq: 102 INVITE [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 6 [ 34]: Contact: [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 7 [ 29]: Content-Type: application/sdp [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 9 [ 21]: Content-Length: 218 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Header 10 [ 0]: [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 0 [ 3]: v=0 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 1 [ 43]: o=cit02 123456 654321 IN IP4 192.168.12.102 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 2 [ 16]: s=A conversation [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 3 [ 23]: c=IN IP4 192.168.12.102 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 5 [ 26]: m=audio 7080 RTP/AVP 0 101 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 6 [ 22]: a=rtpmap:0 PCMU/8000/1 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 7 [ 35]: a=rtpmap:101 telephone-event/8000/1 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 8 [ 15]: a=fmtp:101 0-11 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Body 9 [ 10]: a=sendrecv [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: --- (10 headers 10 lines) --- [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Acked pending invite 102 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Stopping retransmission on '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' of Request 102: Match Found [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: SIP response 200 to standard invite [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED. [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing session-level SDP o=cit02 123456 654321 IN IP4 192.168.12.102... UNSUPPORTED. [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED. [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.12.102... OK. [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED. [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Found RTP audio format 0 [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Found RTP audio format 101 [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000/1... OK. [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000/1... OK. [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED. [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Peer audio RTP is at port 192.168.12.102:7080 [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: We're settling with these formats: 0x4 (ulaw) [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: We have an owner, now see if we need to change this call [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Updating call counter for outgoing call [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: build_route: Contact hop: [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: list_route: hop: [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Strict routing enforced for session 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: set_destination: set destination to 192.168.12.102, port 5062 [Jan 29 18:12:12] VERBOSE[2946] chan_sip.c: Transmitting (no NAT) to 192.168.12.102:5062: ACK sip:192.168.12.102:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK7416a25e;rport Max-Forwards: 70 From: "cit03" ;tag=as4b49de71 To: ;tag=718914218 Contact: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.9-2 Content-Length: 0 --- [Jan 29 18:12:12] DEBUG[2946] chan_sip.c: Trying to put 'ACK sip:192' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:12:12] VERBOSE[2960] app_dial.c: -- SIP/cit02-00000003 answered SIP/cit03-00000002 [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: SIP answering channel: SIP/cit03-00000002 [Jan 29 18:12:12] DEBUG[2960] rtp.c: Setting the marker bit due to a source update [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: Setting framing from config on incoming call [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 29 18:12:12] VERBOSE[2960] chan_sip.c: Audio is at 192.168.12.103 port 14160 [Jan 29 18:12:12] VERBOSE[2960] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 29 18:12:12] VERBOSE[2960] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: -- Done with adding codecs to SDP [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw) [Jan 29 18:12:12] VERBOSE[2960] chan_sip.c: <--- Reliably Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK1809766208;received=192.168.12.103;rport=5062 From: ;tag=2047668150 To: ;tag=as539f0966 Call-ID: 380593170 CSeq: 20 INVITE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 843800493 843800493 IN IP4 192.168.12.103 s=Asterisk PBX 1.6.2.9-2 c=IN IP4 192.168.12.103 t=0 0 m=audio 14160 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #40 [Jan 29 18:12:12] DEBUG[2960] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:12:12] DEBUG[2960] features.c: bridge answer set, chan answer set [Jan 29 18:12:12] DEBUG[2960] rtp.c: Setting the marker bit due to a source update [Jan 29 18:12:12] DEBUG[2960] rtp.c: Setting the marker bit due to a source update [Jan 29 18:12:12] VERBOSE[2960] rtp.c: -- Packet2Packet bridging SIP/cit03-00000002 and SIP/cit02-00000003 [Jan 29 18:12:12] DEBUG[2940] devicestate.c: No provider found, checking channel drivers for SIP - cit02 [Jan 29 18:12:12] DEBUG[2940] chan_sip.c: Checking device state for peer cit02 [Jan 29 18:12:12] DEBUG[2940] devicestate.c: Changing state for SIP/cit02 - state 1 (Not in use) [Jan 29 18:12:12] DEBUG[2940] devicestate.c: device 'SIP/cit02' state '1' [Jan 29 18:12:12] DEBUG[2940] devicestate.c: No provider found, checking channel drivers for SIP - cit03 [Jan 29 18:12:12] DEBUG[2940] chan_sip.c: Checking device state for peer cit03 [Jan 29 18:12:12] DEBUG[2940] devicestate.c: Changing state for SIP/cit03 - state 1 (Not in use) [Jan 29 18:12:12] DEBUG[2940] devicestate.c: device 'SIP/cit03' state '1' [Jan 29 18:12:12] DEBUG[2949] app_queue.c: Device 'SIP/cit02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:12:12] DEBUG[2949] app_queue.c: Device 'SIP/cit03' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:12:13] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.103:5062 ---> ACK sip:cit02@192.168.12.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1196938710 From: ;tag=2047668150 To: ;tag=as539f0966 Call-ID: 380593170 CSeq: 20 ACK Contact: Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 0 [ 36]: ACK sip:cit02@192.168.12.103 SIP/2.0 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1196938710 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 2 [ 51]: From: ;tag=2047668150 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 3 [ 49]: To: ;tag=as539f0966 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 4 [ 18]: Call-ID: 380593170 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 5 [ 12]: CSeq: 20 ACK [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 6 [ 40]: Contact: [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Header 10 [ 0]: [Jan 29 18:12:13] VERBOSE[2946] chan_sip.c: --- (10 headers 0 lines) --- [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #40 [Jan 29 18:12:13] DEBUG[2946] chan_sip.c: Stopping retransmission on '380593170' of Response 20: Match Found [Jan 29 18:12:17] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:18] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:22] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:23] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:27] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:28] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:32] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:33] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:37] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:38] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:41] DEBUG[2946] chan_sip.c: Auto destroying SIP dialog '463398228' [Jan 29 18:12:41] DEBUG[2946] chan_sip.c: Destroying SIP dialog 463398228 [Jan 29 18:12:41] VERBOSE[2946] chan_sip.c: Really destroying SIP dialog '463398228' Method: OPTIONS [Jan 29 18:12:42] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:43] DEBUG[2960] rtp.c: Got RTCP report of 144 bytes [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.103:5062 ---> BYE sip:cit02@192.168.12.103 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1529429269 From: ;tag=2047668150 To: ;tag=as539f0966 Call-ID: 380593170 CSeq: 21 BYE Contact: Max-Forwards: 70 User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 0 [ 36]: BYE sip:cit02@192.168.12.103 SIP/2.0 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 192.168.12.103:5062;rport;branch=z9hG4bK1529429269 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 2 [ 51]: From: ;tag=2047668150 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 3 [ 49]: To: ;tag=as539f0966 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 4 [ 18]: Call-ID: 380593170 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 5 [ 12]: CSeq: 21 BYE [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 6 [ 40]: Contact: [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 8 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 9 [ 17]: Content-Length: 0 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 10 [ 0]: [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: --- (10 headers 0 lines) --- [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Initializing initreq for method BYE - callid 380593170 [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: Sending to 192.168.12.103 : 5062 (no NAT) [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Setting SIP_ALREADYGONE on dialog 380593170 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Received bye, issuing owner hangup [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: <--- Transmitting (no NAT) to 192.168.12.103:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5062;branch=z9hG4bK1529429269;received=192.168.12.103;rport=5062 From: ;tag=2047668150 To: ;tag=as539f0966 Call-ID: 380593170 CSeq: 21 BYE Server: Asterisk PBX 1.6.2.9-2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.12.103:5062 [Jan 29 18:12:47] DEBUG[2960] rtp.c: p2p-rtp-bridge: Ooh, got a hangup [Jan 29 18:12:47] DEBUG[2960] channel.c: Returning from native bridge, channels: SIP/cit03-00000002, SIP/cit02-00000003 [Jan 29 18:12:47] DEBUG[2960] pbx.c: Launching 'NoOp' [Jan 29 18:12:47] VERBOSE[2960] pbx.c: -- Executing [h@sax3nlocal:1] NoOp("SIP/cit03-00000002", "HANGUP!!!!") in new stack [Jan 29 18:12:47] DEBUG[2960] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 29 18:12:47] DEBUG[2960] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('cit03','cit03','cit02','sax3nlocal','SIP/cit03-00000002','SIP/cit02-00000003','Dial','SIP/cit02','2007-01-29 18:12:08','2007-01-29 18:12:12','2007-01-29 18:12:47','39','35','ANSWERED','DOCUMENTATION','1170094328.2') [Jan 29 18:12:47] DEBUG[2960] channel.c: Hanging up channel 'SIP/cit02-00000003' [Jan 29 18:12:47] DEBUG[2960] chan_sip.c: Hangup call SIP/cit02-00000003, SIP callid 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:47] VERBOSE[2960] chan_sip.c: Scheduling destruction of SIP dialog '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' in 32000 ms (Method: INVITE) [Jan 29 18:12:47] DEBUG[2960] chan_sip.c: Strict routing enforced for session 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:47] VERBOSE[2960] chan_sip.c: set_destination: Parsing for address/port to send to [Jan 29 18:12:47] VERBOSE[2960] chan_sip.c: set_destination: set destination to 192.168.12.102, port 5062 [Jan 29 18:12:47] VERBOSE[2960] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.12.102:5062: BYE sip:192.168.12.102:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5c3adff4;rport Max-Forwards: 70 From: "cit03" ;tag=as4b49de71 To: ;tag=718914218 Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 103 BYE User-Agent: Asterisk PBX 1.6.2.9-2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jan 29 18:12:47] DEBUG[2960] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #42 [Jan 29 18:12:47] DEBUG[2960] chan_sip.c: Trying to put 'BYE sip:192' onto UDP socket destined for 192.168.12.102:5062 [Jan 29 18:12:47] DEBUG[2960] rtp.c: Channel '' has no RTP, not doing anything [Jan 29 18:12:47] DEBUG[2960] app_dial.c: Exiting with DIALSTATUS=ANSWER. [Jan 29 18:12:47] DEBUG[2960] pbx.c: Spawn extension (sax3nlocal,cit02,2) exited non-zero on 'SIP/cit03-00000002' [Jan 29 18:12:47] VERBOSE[2960] pbx.c: == Spawn extension (sax3nlocal, cit02, 2) exited non-zero on 'SIP/cit03-00000002' [Jan 29 18:12:47] DEBUG[2960] channel.c: Soft-Hanging up channel 'SIP/cit03-00000002' [Jan 29 18:12:47] DEBUG[2960] channel.c: Hanging up channel 'SIP/cit03-00000002' [Jan 29 18:12:47] DEBUG[2960] chan_sip.c: Hangup call SIP/cit03-00000002, SIP callid 380593170 [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: <--- SIP read from UDP:192.168.12.102:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5c3adff4;rport=5060 From: "cit03" ;tag=as4b49de71 To: ;tag=718914218 Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 CSeq: 103 BYE User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) Content-Length: 0 <-------------> [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 1 [ 70]: Via: SIP/2.0/UDP 192.168.12.103:5060;branch=z9hG4bK5c3adff4;rport=5060 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 2 [ 55]: From: "cit03" ;tag=as4b49de71 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 3 [ 42]: To: ;tag=718914218 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 4 [ 56]: Call-ID: 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 6 [ 45]: User-Agent: Linphone/3.3.99.7 (eXosip2/3.3.0) [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Header 8 [ 0]: [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: --- (8 headers 0 lines) --- [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #42 [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Stopping retransmission on '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' of Request 103: Match Found [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Destroying SIP dialog 380593170 [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: Really destroying SIP dialog '380593170' Method: BYE [Jan 29 18:12:47] DEBUG[2946] chan_sip.c: Destroying SIP dialog 0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103 [Jan 29 18:12:47] VERBOSE[2946] chan_sip.c: Really destroying SIP dialog '0bb0d0345be9d1cc18349c7f2d9e28e5@192.168.12.103' Method: INVITE [Jan 29 18:12:47] DEBUG[2940] devicestate.c: No provider found, checking channel drivers for SIP - cit02 [Jan 29 18:12:47] DEBUG[2940] chan_sip.c: Checking device state for peer cit02 [Jan 29 18:12:47] DEBUG[2940] devicestate.c: Changing state for SIP/cit02 - state 1 (Not in use) [Jan 29 18:12:47] DEBUG[2940] devicestate.c: device 'SIP/cit02' state '1' [Jan 29 18:12:47] DEBUG[2940] devicestate.c: No provider found, checking channel drivers for SIP - cit03 [Jan 29 18:12:47] DEBUG[2940] chan_sip.c: Checking device state for peer cit03 [Jan 29 18:12:47] DEBUG[2940] devicestate.c: Changing state for SIP/cit03 - state 1 (Not in use) [Jan 29 18:12:47] DEBUG[2940] devicestate.c: device 'SIP/cit03' state '1' [Jan 29 18:12:47] DEBUG[2949] app_queue.c: Device 'SIP/cit02' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 29 18:12:47] DEBUG[2949] app_queue.c: Device 'SIP/cit03' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.