[Jul 25 21:15:39] VERBOSE[26649] logger.c: <--- SIP read from 1.1.1.1:5060 ---> INVITE sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6aca9298;rport From: "litnialex" ;tag=as230df395 To: Contact: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 25 Jul 2007 18:14:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 186 v=0 o=root 32291 32291 IN IP4 1.1.1.1 s=session c=IN IP4 1.1.1.1 t=0 0 m=audio 18710 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 0: INVITE sip:test@2.2.2.2 SIP/2.0 (37) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6aca9298;rport (64) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 2: From: "litnialex" ;tag=as230df395 (56) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 3: To: (28) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 4: Contact: (32) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 (55) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 6: CSeq: 102 INVITE (16) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 9: Date: Wed, 25 Jul 2007 18:14:45 GMT (35) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 11: Supported: replaces (19) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 12: Content-Type: application/sdp (29) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 13: Content-Length: 186 (19) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 14: (0) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: v=0 (3) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: o=root 32291 32291 IN IP4 1.1.1.1 (39) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: s=session (9) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: c=IN IP4 1.1.1.1 (22) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: t=0 0 (5) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: m=audio 18710 RTP/AVP 8 (23) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: a=rtpmap:8 PCMA/8000 (20) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: a=silenceSupp:off - - - - (25) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: a=ptime:20 (10) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Line: a=sendrecv (10) [Jul 25 21:15:39] VERBOSE[26649] logger.c: --- (14 headers 10 lines) --- [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Setting NAT on RTP to On [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Allocating new SIP dialog for 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 - INVITE (With RTP) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Begin: parsing SIP "Supported: replaces" [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Found SIP option: -replaces- [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Matched SIP option: replaces [Jul 25 21:15:39] VERBOSE[26649] logger.c: Sending to 1.1.1.1 : 5060 (NAT) [Jul 25 21:15:39] VERBOSE[26649] logger.c: Using INVITE request as basis request - 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 [Jul 25 21:15:39] VERBOSE[26649] logger.c: Found no matching peer or user for '1.1.1.1:5060' [Jul 25 21:15:39] VERBOSE[26649] logger.c: Found RTP audio format 8 [Jul 25 21:15:39] VERBOSE[26649] logger.c: Peer audio RTP is at port 1.1.1.1:18710 [Jul 25 21:15:39] VERBOSE[26649] logger.c: Found description format PCMA for ID 8 [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: T38 state changed to 0 on channel [Jul 25 21:15:39] VERBOSE[26649] logger.c: Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) [Jul 25 21:15:39] VERBOSE[26649] logger.c: Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Jul 25 21:15:39] VERBOSE[26649] logger.c: Peer audio RTP is at port 1.1.1.1:18710 [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: We're settling with these formats: 0x8 (alaw) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Checking SIP call limits for device [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Updating call counter for incoming call [Jul 25 21:15:39] VERBOSE[26649] logger.c: Looking for test in ivrmanager (domain 2.2.2.2) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: *** Joint capabilities are 0x8 (alaw) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: *** Our capabilities are 0x8 (alaw) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: This channel will not be able to handle video. [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: build_route: Contact hop: [Jul 25 21:15:39] VERBOSE[26649] logger.c: list_route: hop: [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: SIP/1.1.1.1-006c69c0: New call is still down.... Trying... [Jul 25 21:15:39] VERBOSE[26649] logger.c: <--- Transmitting (NAT) to 1.1.1.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6aca9298;received=1.1.1.1;rport=5060 From: "litnialex" ;tag=as230df395 To: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jul 25 21:15:39] DEBUG[26649] devicestate.c: Notification of state change to be queued on device/channel SIP/1.1.1.1-006c69c0 [Jul 25 21:15:39] DEBUG[26628] devicestate.c: No provider found, checking channel drivers for SIP - 1.1.1.1 [Jul 25 21:15:39] DEBUG[26628] chan_sip.c: Checking device state for peer 1.1.1.1 [Jul 25 21:15:39] DEBUG[26628] devicestate.c: Changing state for SIP/1.1.1.1 - state 2 (In use) [Jul 25 21:15:39] DEBUG[28062] pbx.c: Launching 'Answer' [Jul 25 21:15:39] VERBOSE[28062] logger.c: -- Executing [test@ivrmanager:1] Answer("SIP/1.1.1.1-006c69c0", "") in new stack [Jul 25 21:15:39] DEBUG[28062] devicestate.c: Notification of state change to be queued on device/channel SIP/1.1.1.1-006c69c0 [Jul 25 21:15:39] DEBUG[26628] devicestate.c: No provider found, checking channel drivers for SIP - 1.1.1.1 [Jul 25 21:15:39] DEBUG[26628] chan_sip.c: Checking device state for peer 1.1.1.1 [Jul 25 21:15:39] DEBUG[26628] channel.c: Avoiding initial deadlock for channel '0x6a3ef0' [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: SIP answering channel: SIP/1.1.1.1-006c69c0 [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: Setting framing from config on incoming call [Jul 25 21:15:39] DEBUG[28063] app_queue.c: Device 'SIP/1.1.1.1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jul 25 21:15:39] VERBOSE[28062] logger.c: Audio is at 2.2.2.2 port 13718 [Jul 25 21:15:39] VERBOSE[28062] logger.c: Adding codec 0x8 (alaw) to SDP [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: -- Done with adding codecs to SDP [Jul 25 21:15:39] DEBUG[28062] channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=24) [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: Done building SDP. Settling with this capability: 0x8 (alaw) [Jul 25 21:15:39] VERBOSE[28062] logger.c: <--- Reliably Transmitting (NAT) to 1.1.1.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6aca9298;received=1.1.1.1;rport=5060 From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 26624 26624 IN IP4 2.2.2.2 s=session c=IN IP4 2.2.2.2 t=0 0 m=audio 13718 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jul 25 21:15:39] DEBUG[28062] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id #105 [Jul 25 21:15:39] DEBUG[28062] pbx.c: Launching 'Read' [Jul 25 21:15:39] VERBOSE[28062] logger.c: -- Executing [test@ivrmanager:2] Read("SIP/1.1.1.1-006c69c0", "asdf|demo-congrats|1") in new stack [Jul 25 21:15:39] VERBOSE[28062] logger.c: -- Accepting a maximum of 1 digits. [Jul 25 21:15:39] DEBUG[28062] channel.c: Set channel SIP/1.1.1.1-006c69c0 to write format gsm [Jul 25 21:15:39] DEBUG[28062] rtp.c: Ooh, format changed from unknown to alaw [Jul 25 21:15:39] DEBUG[28062] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jul 25 21:15:39] DEBUG[28062] channel.c: Scheduling timer at 160 sample intervals [Jul 25 21:15:39] VERBOSE[28062] logger.c: -- Playing 'demo-congrats' (language 'en') [Jul 25 21:15:39] DEBUG[26628] devicestate.c: Changing state for SIP/1.1.1.1 - state 2 (In use) [Jul 25 21:15:39] DEBUG[28064] app_queue.c: Device 'SIP/1.1.1.1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 25 21:15:39] VERBOSE[26649] logger.c: <--- SIP read from 1.1.1.1:5060 ---> ACK sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6dd4a9ea;rport From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Contact: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 0: ACK sip:test@2.2.2.2 SIP/2.0 (34) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6dd4a9ea;rport (64) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 2: From: "litnialex" ;tag=as230df395 (56) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 3: To: ;tag=as3a54794c (43) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 4: Contact: (32) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 (55) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 6: CSeq: 102 ACK (13) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 9: Content-Length: 0 (17) [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Header 10: (0) [Jul 25 21:15:39] VERBOSE[26649] logger.c: --- (10 headers 0 lines) --- [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: = Found Their Call ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 Their Tag as230df395 Our tag: as3a54794c [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #105 [Jul 25 21:15:39] DEBUG[26649] chan_sip.c: Stopping retransmission on '7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1' of Response 102: Match Not Found [Jul 25 21:15:44] VERBOSE[26649] logger.c: <--- SIP read from 1.1.1.1:5060 ---> INFO sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7dc0f4fd;rport From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Contact: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 103 INFO User-Agent: Asterisk PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 31 Signal=1 Duration=4271283118 <-------------> [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 0: INFO sip:test@2.2.2.2 SIP/2.0 (35) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7dc0f4fd;rport (64) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 2: From: "litnialex" ;tag=as230df395 (56) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 3: To: ;tag=as3a54794c (43) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 4: Contact: (32) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 (55) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 6: CSeq: 103 INFO (14) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 9: Content-Type: application/dtmf-relay (36) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 10: Content-Length: 31 (18) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Header 11: (0) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Line: Signal=1 (8) [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: Line: Duration=4271283118 (19) [Jul 25 21:15:44] VERBOSE[26649] logger.c: --- (11 headers 2 lines) --- [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: = Found Their Call ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 Their Tag as230df395 Our tag: as3a54794c [Jul 25 21:15:44] DEBUG[26649] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jul 25 21:15:44] VERBOSE[26649] logger.c: Receiving INFO! [Jul 25 21:15:44] VERBOSE[26649] logger.c: * DTMF-relay event received: 1 [Jul 25 21:15:44] VERBOSE[26649] logger.c: <--- Transmitting (NAT) to 1.1.1.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7dc0f4fd;received=1.1.1.1;rport=5060 From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 103 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jul 25 21:15:44] DTMF[28062] channel.c: DTMF end '1' received on SIP/1.1.1.1-006c69c0, duration 4271283118 ms [Jul 25 21:15:44] DTMF[28062] channel.c: DTMF begin emulation of '1' with duration -23684178 queued on SIP/1.1.1.1-006c69c0 [Jul 25 21:15:44] DEBUG[28062] rtp.c: Got RTCP report of 64 bytes [Jul 25 21:15:49] VERBOSE[26649] logger.c: <--- SIP read from 1.1.1.1:5060 ---> INFO sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK10814c39;rport From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Contact: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 104 INFO User-Agent: Asterisk PBX Max-Forwards: 70 Content-Type: application/dtmf-relay Content-Length: 25 Signal=2 Duration=5199 <-------------> [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 0: INFO sip:test@2.2.2.2 SIP/2.0 (35) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK10814c39;rport (64) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 2: From: "litnialex" ;tag=as230df395 (56) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 3: To: ;tag=as3a54794c (43) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 4: Contact: (32) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 5: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 (55) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 6: CSeq: 104 INFO (14) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 7: User-Agent: Asterisk PBX (24) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 8: Max-Forwards: 70 (16) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 9: Content-Type: application/dtmf-relay (36) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 10: Content-Length: 25 (18) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Header 11: (0) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Line: Signal=2 (8) [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: Line: Duration=5199 (13) [Jul 25 21:15:49] VERBOSE[26649] logger.c: --- (11 headers 2 lines) --- [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: = Found Their Call ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 Their Tag as230df395 Our tag: as3a54794c [Jul 25 21:15:49] DEBUG[26649] chan_sip.c: **** Received INFO (13) - Command in SIP INFO [Jul 25 21:15:49] VERBOSE[26649] logger.c: Receiving INFO! [Jul 25 21:15:49] VERBOSE[26649] logger.c: * DTMF-relay event received: 2 [Jul 25 21:15:49] VERBOSE[26649] logger.c: <--- Transmitting (NAT) to 1.1.1.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK10814c39;received=1.1.1.1;rport=5060 From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 104 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jul 25 21:15:49] DTMF[28062] channel.c: DTMF end '2' received on SIP/1.1.1.1-006c69c0, duration 5199 ms [Jul 25 21:15:49] DEBUG[28062] rtp.c: Got RTCP report of 64 bytes [Jul 25 21:15:54] DEBUG[28062] rtp.c: Got RTCP report of 64 bytes [Jul 25 21:15:55] VERBOSE[26649] logger.c: <--- SIP read from 1.1.1.1:5060 ---> BYE sip:test@2.2.2.2 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK342c82d2;rport From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 <-------------> [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 0: BYE sip:test@2.2.2.2 SIP/2.0 (34) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 1: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK342c82d2;rport (64) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 2: From: "litnialex" ;tag=as230df395 (56) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 3: To: ;tag=as3a54794c (43) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 4: Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 (55) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 5: CSeq: 105 BYE (13) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 7: Max-Forwards: 70 (16) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 8: Content-Length: 0 (17) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Header 9: (0) [Jul 25 21:15:55] VERBOSE[26649] logger.c: --- (9 headers 0 lines) --- [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: = Found Their Call ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 Their Tag as230df395 Our tag: as3a54794c [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jul 25 21:15:55] VERBOSE[26649] logger.c: Sending to 1.1.1.1 : 5060 (NAT) [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Setting SIP_ALREADYGONE on dialog 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 [Jul 25 21:15:55] DEBUG[26649] chan_sip.c: Received bye, issuing owner hangup [Jul 25 21:15:55] VERBOSE[26649] logger.c: <--- Transmitting (NAT) to 1.1.1.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK342c82d2;received=1.1.1.1;rport=5060 From: "litnialex" ;tag=as230df395 To: ;tag=as3a54794c Call-ID: 7988e4a57fa9b1c91dc045b72a07c3f6@1.1.1.1 CSeq: 105 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0