#### sip.conf: #### [iam_friend] #### ... #### [iam_peer] #### ... [Jan 9 16:49:38] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> INVITE sip:test@88.99.0.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKyxfxlkrr Max-Forwards: 70 To: From: "Anonimo" ;tag=ixiwr Call-ID: xrfovqbpqnpadhc@192.168.1.58 CSeq: 286 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.1 Content-Length: 309 v=0 o=anonimo 1586468422 1669568257 IN IP4 192.168.1.58 s=- c=IN IP4 192.168.1.58 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 0 [ 35]: INVITE sip:test@88.99.0.111 SIP/2.0 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKyxfxlkrr [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 3 [ 26]: To: [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 4 [ 55]: From: "Anonimo" ;tag=ixiwr [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 5 [ 37]: Call-ID: xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 6 [ 16]: CSeq: 286 INVITE [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 7 [ 35]: Contact: [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 9 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 10 [ 37]: Supported: replaces,norefersub,100rel [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 11 [ 23]: User-Agent: Twinkle/1.1 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 12 [ 19]: Content-Length: 309 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 13 [ 0]: [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 0 [ 3]: v=0 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 1 [ 51]: o=anonimo 1586468422 1669568257 IN IP4 192.168.1.58 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 2 [ 3]: s=- [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.58 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 5 [ 36]: m=audio 8000 RTP/AVP 98 97 8 0 3 101 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 6 [ 23]: a=rtpmap:98 speex/16000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 7 [ 22]: a=rtpmap:97 speex/8000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Body 13 [ 10]: a=ptime:20 [Jan 9 16:49:38] VERBOSE[16653] logger.c: --- (13 headers 14 lines) --- [Jan 9 16:49:38] DEBUG[16653] acl.c: Found IP address for this socket [Jan 9 16:49:38] VERBOSE[16653] logger.c: == Using SIP RTP CoS mark 5 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Setting NAT on RTP to Off [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Allocating new SIP dialog for xrfovqbpqnpadhc@192.168.1.58 - INVITE (With RTP) [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel" [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Found SIP option: -replaces- [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Matched SIP option: replaces [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Found SIP option: -norefersub- [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Matched SIP option: norefersub [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Found SIP option: -100rel- [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Matched SIP option: 100rel [Jan 9 16:49:38] VERBOSE[16653] logger.c: Sending to 222.111.235.18 : 1033 (NAT) [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Initializing initreq for method INVITE - callid xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Using INVITE request as basis request - xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:38] VERBOSE[16653] logger.c: No user 'anonimo' in SIP users list [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found peer 'iam_peer' for 'anonimo' from 222.111.235.18:1033 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Setting NAT on RTP to On [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found RTP audio format 98 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found RTP audio format 97 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found RTP audio format 8 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found RTP audio format 0 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found RTP audio format 3 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found RTP audio format 101 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Peer audio RTP is at port 192.168.1.58:8000 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found audio description format speex for ID 98 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found audio description format speex for ID 97 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found audio description format PCMA for ID 8 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found audio description format PCMU for ID 0 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found audio description format GSM for ID 3 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Found audio description format telephone-event for ID 101 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Got unsupported a:fmtp in SDP offer [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: T38 state changed to 0 on channel [Jan 9 16:49:38] VERBOSE[16653] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 9 16:49:38] VERBOSE[16653] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 9 16:49:38] VERBOSE[16653] logger.c: Peer audio RTP is at port 192.168.1.58:8000 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Checking SIP call limits for device iam_peer [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Updating call counter for incoming call [Jan 9 16:49:38] VERBOSE[16653] logger.c: Looking for test in test (domain 88.99.0.111) [Jan 9 16:49:38] DEBUG[16653] frame.c: Could not find preferred codec - Going for the best codec [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Jan 9 16:49:38] DEBUG[16653] frame.c: Could not find preferred codec - Going for the best codec [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: This channel will not be able to handle video. [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: build_route: Contact hop: [Jan 9 16:49:38] VERBOSE[16653] logger.c: list_route: hop: [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: SIP/iam_peer-081fed38: New call is still down.... Trying... [Jan 9 16:49:38] VERBOSE[16653] logger.c: <--- Transmitting (NAT) to 222.111.235.18:1033 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKyxfxlkrr;received=222.111.235.18;rport=1033 From: "Anonimo" ;tag=ixiwr To: Call-ID: xrfovqbpqnpadhc@192.168.1.58 CSeq: 286 INVITE User-Agent: Asterisk SVN trunk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 9 16:49:38] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer-081fed38 [Jan 9 16:49:38] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer [Jan 9 16:49:38] DEBUG[16653] pbx.c: Launching 'Playback' [Jan 9 16:49:38] VERBOSE[16653] logger.c: -- Executing [test@test:1] Playback("SIP/iam_peer-081fed38", "demo-thanks") in new stack [Jan 9 16:49:38] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer-081fed38 [Jan 9 16:49:38] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: SIP answering channel: SIP/iam_peer-081fed38 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Setting framing from config on incoming call [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 9 16:49:38] VERBOSE[16653] logger.c: Audio is at 88.99.0.111 port 10458 [Jan 9 16:49:38] VERBOSE[16653] logger.c: Adding codec 0x2 (gsm) to SDP [Jan 9 16:49:38] VERBOSE[16653] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 9 16:49:38] VERBOSE[16653] logger.c: Adding codec 0x8 (alaw) to SDP [Jan 9 16:49:38] VERBOSE[16653] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: -- Done with adding codecs to SDP [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 9 16:49:38] VERBOSE[16653] logger.c: <--- Reliably Transmitting (NAT) to 222.111.235.18:1033 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKyxfxlkrr;received=222.111.235.18;rport=1033 From: "Anonimo" ;tag=ixiwr To: ;tag=as2f255ac5 Call-ID: xrfovqbpqnpadhc@192.168.1.58 CSeq: 286 INVITE User-Agent: Asterisk SVN trunk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 317 v=0 o=root 1880878628 1880878628 IN IP4 88.99.0.111 s=Asterisk PBX SVN-trunk-r96988 c=IN IP4 88.99.0.111 t=0 0 m=audio 10458 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #3 [Jan 9 16:49:38] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer-081fed38 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer-081fed38 [Jan 9 16:49:38] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> ACK sip:test@88.99.0.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKfzoweesw Max-Forwards: 70 To: ;tag=as2f255ac5 From: "Anonimo" ;tag=ixiwr Call-ID: xrfovqbpqnpadhc@192.168.1.58 CSeq: 286 ACK User-Agent: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 0 [ 32]: ACK sip:test@88.99.0.111 SIP/2.0 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKfzoweesw [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 3 [ 41]: To: ;tag=as2f255ac5 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 4 [ 55]: From: "Anonimo" ;tag=ixiwr [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 5 [ 37]: Call-ID: xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 6 [ 13]: CSeq: 286 ACK [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 7 [ 23]: User-Agent: Twinkle/1.1 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Header 9 [ 0]: [Jan 9 16:49:38] VERBOSE[16653] logger.c: --- (9 headers 0 lines) --- [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #3 [Jan 9 16:49:38] DEBUG[16653] chan_sip.c: Stopping retransmission on 'xrfovqbpqnpadhc@192.168.1.58' of Response 286: Match Found [Jan 9 16:49:39] DEBUG[16653] rtp.c: RTCP NAT: Got RTCP from other end. Now sending to address 222.111.235.18:8001 [Jan 9 16:49:39] DEBUG[16653] rtp.c: Got RTCP report of 40 bytes [Jan 9 16:49:39] DEBUG[16653] rtp.c: RTP NAT: Got audio from other end. Now sending to address 222.111.235.18:8000 [Jan 9 16:49:39] DEBUG[16653] chan_sip.c: Oooh, format changed to 2 gsm [Jan 9 16:49:39] DEBUG[16653] channel.c: Set channel SIP/iam_peer-081fed38 to read format ulaw [Jan 9 16:49:39] DEBUG[16653] channel.c: Set channel SIP/iam_peer-081fed38 to write format ulaw [Jan 9 16:49:39] DEBUG[16653] channel.c: Set channel SIP/iam_peer-081fed38 to write format alaw [Jan 9 16:49:39] DEBUG[16653] rtp.c: Ooh, format changed from unknown to gsm [Jan 9 16:49:39] DEBUG[16653] rtp.c: Created smoother: format: 2 ms: 20 len: 33 [Jan 9 16:49:39] VERBOSE[16653] logger.c: -- Playing 'demo-thanks.alaw' (language 'es') [Jan 9 16:49:39] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer-081fed38 - state 4 (Invalid) [Jan 9 16:49:39] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer [Jan 9 16:49:39] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer [Jan 9 16:49:39] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer - state 1 (Not in use) [Jan 9 16:49:39] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer-081fed38 [Jan 9 16:49:39] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer-081fed38 [Jan 9 16:49:39] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer-081fed38' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:49:39] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:49:39] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer-081fed38 - state 4 (Invalid) [Jan 9 16:49:39] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer [Jan 9 16:49:39] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer [Jan 9 16:49:39] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer - state 1 (Not in use) [Jan 9 16:49:39] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer-081fed38' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:49:39] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:49:42] DEBUG[16653] channel.c: Set channel SIP/iam_peer-081fed38 to write format ulaw [Jan 9 16:49:42] VERBOSE[16653] logger.c: -- Auto fallthrough, channel 'SIP/iam_peer-081fed38' status is 'UNKNOWN' [Jan 9 16:49:42] DEBUG[16653] channel.c: Soft-Hanging up channel 'SIP/iam_peer-081fed38' [Jan 9 16:49:42] DEBUG[16653] channel.c: Hanging up channel 'SIP/iam_peer-081fed38' [Jan 9 16:49:42] DEBUG[16653] chan_sip.c: Hangup call SIP/iam_peer-081fed38, SIP callid xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:42] VERBOSE[16653] logger.c: Scheduling destruction of SIP dialog 'xrfovqbpqnpadhc@192.168.1.58' in 32000 ms (Method: ACK) [Jan 9 16:49:42] DEBUG[16653] chan_sip.c: Strict routing enforced for session xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:42] VERBOSE[16653] logger.c: set_destination: Parsing for address/port to send to [Jan 9 16:49:42] VERBOSE[16653] logger.c: set_destination: set destination to 192.168.1.58, port 5060 [Jan 9 16:49:42] VERBOSE[16653] logger.c: Reliably Transmitting (NAT) to 222.111.235.18:1033: BYE sip:anonimo@192.168.1.58 SIP/2.0 Via: SIP/2.0/UDP 88.99.0.111:5060;branch=z9hG4bK66a6ec43;rport Max-Forwards: 70 From: ;tag=as2f255ac5 To: "Anonimo" ;tag=ixiwr Call-ID: xrfovqbpqnpadhc@192.168.1.58 CSeq: 102 BYE User-Agent: Asterisk SVN trunk Content-Length: 0 --- [Jan 9 16:49:42] DEBUG[16653] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Jan 9 16:49:42] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer-081fed38 [Jan 9 16:49:42] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer [Jan 9 16:49:42] DEBUG[16653] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 9 16:49:42] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer-081fed38 [Jan 9 16:49:42] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer-081fed38 [Jan 9 16:49:42] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer-081fed38 - state 4 (Invalid) [Jan 9 16:49:42] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer [Jan 9 16:49:42] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer [Jan 9 16:49:42] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer - state 1 (Not in use) [Jan 9 16:49:42] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer-081fed38' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:49:42] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:49:43] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 88.99.0.111:5060;rport=5060;branch=z9hG4bK66a6ec43 To: "Anonimo" ;tag=ixiwr From: ;tag=as2f255ac5 Call-ID: xrfovqbpqnpadhc@192.168.1.58 CSeq: 102 BYE Server: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.99.0.111:5060;rport=5060;branch=z9hG4bK66a6ec43 [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 2 [ 53]: To: "Anonimo" ;tag=ixiwr [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 3 [ 43]: From: ;tag=as2f255ac5 [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 4 [ 37]: Call-ID: xrfovqbpqnpadhc@192.168.1.58 [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 6 [ 19]: Server: Twinkle/1.1 [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Header 8 [ 0]: [Jan 9 16:49:43] VERBOSE[16653] logger.c: --- (8 headers 0 lines) --- [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: Stopping retransmission on 'xrfovqbpqnpadhc@192.168.1.58' of Request 102: Match Found [Jan 9 16:49:43] VERBOSE[16653] logger.c: SIP Response message for INCOMING dialog BYE arrived [Jan 9 16:49:43] VERBOSE[16653] logger.c: Really destroying SIP dialog 'xrfovqbpqnpadhc@192.168.1.58' Method: ACK [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: ---------- SIP HISTORY for 'xrfovqbpqnpadhc@192.168.1.58' [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: * SIP Call [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: 001. Hangup Cause Unknown [Jan 9 16:49:43] DEBUG[16653] chan_sip.c: ---------- END SIP HISTORY for 'xrfovqbpqnpadhc@192.168.1.58' [Jan 9 16:50:02] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> INVITE sip:test@88.99.0.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKphdbzkxg Max-Forwards: 70 To: From: "Anonimo" ;tag=dggqa Call-ID: lyrmscfuortfsxl@192.168.1.58 CSeq: 180 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.1 Content-Length: 312 v=0 o=iam_friend 1800929682 1889284542 IN IP4 192.168.1.58 s=- c=IN IP4 192.168.1.58 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 0 [ 35]: INVITE sip:test@88.99.0.111 SIP/2.0 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKphdbzkxg [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 3 [ 26]: To: [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 4 [ 58]: From: "Anonimo" ;tag=dggqa [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 5 [ 37]: Call-ID: lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 6 [ 16]: CSeq: 180 INVITE [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 7 [ 38]: Contact: [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 9 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 10 [ 37]: Supported: replaces,norefersub,100rel [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 11 [ 23]: User-Agent: Twinkle/1.1 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 12 [ 19]: Content-Length: 312 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 13 [ 0]: [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 0 [ 3]: v=0 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 1 [ 54]: o=iam_friend 1800929682 1889284542 IN IP4 192.168.1.58 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 2 [ 3]: s=- [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.58 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 5 [ 36]: m=audio 8000 RTP/AVP 98 97 8 0 3 101 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 6 [ 23]: a=rtpmap:98 speex/16000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 7 [ 22]: a=rtpmap:97 speex/8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Body 13 [ 10]: a=ptime:20 [Jan 9 16:50:02] VERBOSE[16653] logger.c: --- (13 headers 14 lines) --- [Jan 9 16:50:02] DEBUG[16653] acl.c: Found IP address for this socket [Jan 9 16:50:02] VERBOSE[16653] logger.c: == Using SIP RTP CoS mark 5 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Setting NAT on RTP to Off [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Allocating new SIP dialog for lyrmscfuortfsxl@192.168.1.58 - INVITE (With RTP) [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel" [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Found SIP option: -replaces- [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Matched SIP option: replaces [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Found SIP option: -norefersub- [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Matched SIP option: norefersub [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Found SIP option: -100rel- [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Matched SIP option: 100rel [Jan 9 16:50:02] VERBOSE[16653] logger.c: Sending to 222.111.235.18 : 1033 (NAT) [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Initializing initreq for method INVITE - callid lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Using INVITE request as basis request - lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found user 'iam_friend' for 'iam_friend' [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Setting NAT on RTP to On [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found RTP audio format 98 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found RTP audio format 97 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found RTP audio format 8 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found RTP audio format 0 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found RTP audio format 3 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found RTP audio format 101 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Peer audio RTP is at port 192.168.1.58:8000 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found audio description format speex for ID 98 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found audio description format speex for ID 97 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found audio description format PCMA for ID 8 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found audio description format PCMU for ID 0 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found audio description format GSM for ID 3 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Found audio description format telephone-event for ID 101 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Got unsupported a:fmtp in SDP offer [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: T38 state changed to 0 on channel [Jan 9 16:50:02] VERBOSE[16653] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 9 16:50:02] VERBOSE[16653] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 9 16:50:02] VERBOSE[16653] logger.c: Peer audio RTP is at port 192.168.1.58:8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Checking SIP call limits for device iam_friend [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Updating call counter for incoming call [Jan 9 16:50:02] VERBOSE[16653] logger.c: Looking for test in test (domain 88.99.0.111) [Jan 9 16:50:02] DEBUG[16653] frame.c: Could not find preferred codec - Going for the best codec [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Jan 9 16:50:02] DEBUG[16653] frame.c: Could not find preferred codec - Going for the best codec [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: This channel will not be able to handle video. [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: build_route: Contact hop: [Jan 9 16:50:02] VERBOSE[16653] logger.c: list_route: hop: [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: SIP/iam_friend-08206828: New call is still down.... Trying... [Jan 9 16:50:02] VERBOSE[16653] logger.c: <--- Transmitting (NAT) to 222.111.235.18:1033 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKphdbzkxg;received=222.111.235.18;rport=1033 From: "Anonimo" ;tag=dggqa To: Call-ID: lyrmscfuortfsxl@192.168.1.58 CSeq: 180 INVITE User-Agent: Asterisk SVN trunk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_friend [Jan 9 16:50:02] DEBUG[16653] pbx.c: Launching 'Playback' [Jan 9 16:50:02] VERBOSE[16653] logger.c: -- Executing [test@test:1] Playback("SIP/iam_friend-08206828", "demo-thanks") in new stack [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_friend [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: SIP answering channel: SIP/iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Setting framing from config on incoming call [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 9 16:50:02] VERBOSE[16653] logger.c: Audio is at 88.99.0.111 port 10852 [Jan 9 16:50:02] VERBOSE[16653] logger.c: Adding codec 0x2 (gsm) to SDP [Jan 9 16:50:02] VERBOSE[16653] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 9 16:50:02] VERBOSE[16653] logger.c: Adding codec 0x8 (alaw) to SDP [Jan 9 16:50:02] VERBOSE[16653] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: -- Done with adding codecs to SDP [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 9 16:50:02] VERBOSE[16653] logger.c: <--- Reliably Transmitting (NAT) to 222.111.235.18:1033 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKphdbzkxg;received=222.111.235.18;rport=1033 From: "Anonimo" ;tag=dggqa To: ;tag=as6f197a66 Call-ID: lyrmscfuortfsxl@192.168.1.58 CSeq: 180 INVITE User-Agent: Asterisk SVN trunk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 315 v=0 o=root 333960278 333960278 IN IP4 88.99.0.111 s=Asterisk PBX SVN-trunk-r96988 c=IN IP4 88.99.0.111 t=0 0 m=audio 10852 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #7 [Jan 9 16:50:02] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Checking device state for peer iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Changing state for SIP/iam_friend-08206828 - state 4 (Invalid) [Jan 9 16:50:02] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_friend [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Checking device state for peer iam_friend [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Changing state for SIP/iam_friend - state 1 (Not in use) [Jan 9 16:50:02] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Checking device state for peer iam_friend-08206828 [Jan 9 16:50:02] DEBUG[16653] app_queue.c: Device 'SIP/iam_friend-08206828' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:50:02] DEBUG[16653] app_queue.c: Device 'SIP/iam_friend' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Changing state for SIP/iam_friend-08206828 - state 4 (Invalid) [Jan 9 16:50:02] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_friend [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Checking device state for peer iam_friend [Jan 9 16:50:02] DEBUG[16653] devicestate.c: Changing state for SIP/iam_friend - state 1 (Not in use) [Jan 9 16:50:02] DEBUG[16653] app_queue.c: Device 'SIP/iam_friend-08206828' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:50:02] DEBUG[16653] app_queue.c: Device 'SIP/iam_friend' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:50:02] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> ACK sip:test@88.99.0.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKmdrmwffy Max-Forwards: 70 To: ;tag=as6f197a66 From: "Anonimo" ;tag=dggqa Call-ID: lyrmscfuortfsxl@192.168.1.58 CSeq: 180 ACK User-Agent: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 0 [ 32]: ACK sip:test@88.99.0.111 SIP/2.0 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKmdrmwffy [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 3 [ 41]: To: ;tag=as6f197a66 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 4 [ 58]: From: "Anonimo" ;tag=dggqa [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 5 [ 37]: Call-ID: lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 6 [ 13]: CSeq: 180 ACK [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 7 [ 23]: User-Agent: Twinkle/1.1 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Header 9 [ 0]: [Jan 9 16:50:02] VERBOSE[16653] logger.c: --- (9 headers 0 lines) --- [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #7 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Stopping retransmission on 'lyrmscfuortfsxl@192.168.1.58' of Response 180: Match Found [Jan 9 16:50:02] DEBUG[16653] rtp.c: RTCP NAT: Got RTCP from other end. Now sending to address 222.111.235.18:8001 [Jan 9 16:50:02] DEBUG[16653] rtp.c: Got RTCP report of 40 bytes [Jan 9 16:50:02] DEBUG[16653] rtp.c: RTP NAT: Got audio from other end. Now sending to address 222.111.235.18:8000 [Jan 9 16:50:02] DEBUG[16653] chan_sip.c: Oooh, format changed to 2 gsm [Jan 9 16:50:02] DEBUG[16653] channel.c: Set channel SIP/iam_friend-08206828 to read format ulaw [Jan 9 16:50:02] DEBUG[16653] channel.c: Set channel SIP/iam_friend-08206828 to write format ulaw [Jan 9 16:50:02] DEBUG[16653] channel.c: Set channel SIP/iam_friend-08206828 to write format alaw [Jan 9 16:50:02] DEBUG[16653] rtp.c: Ooh, format changed from unknown to gsm [Jan 9 16:50:02] DEBUG[16653] rtp.c: Created smoother: format: 2 ms: 20 len: 33 [Jan 9 16:50:02] VERBOSE[16653] logger.c: -- Playing 'demo-thanks.alaw' (language 'es') [Jan 9 16:50:06] DEBUG[16653] channel.c: Set channel SIP/iam_friend-08206828 to write format ulaw [Jan 9 16:50:06] VERBOSE[16653] logger.c: -- Auto fallthrough, channel 'SIP/iam_friend-08206828' status is 'UNKNOWN' [Jan 9 16:50:06] DEBUG[16653] channel.c: Soft-Hanging up channel 'SIP/iam_friend-08206828' [Jan 9 16:50:06] DEBUG[16653] channel.c: Hanging up channel 'SIP/iam_friend-08206828' [Jan 9 16:50:06] DEBUG[16653] chan_sip.c: Hangup call SIP/iam_friend-08206828, SIP callid lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:06] VERBOSE[16653] logger.c: Scheduling destruction of SIP dialog 'lyrmscfuortfsxl@192.168.1.58' in 32000 ms (Method: ACK) [Jan 9 16:50:06] DEBUG[16653] chan_sip.c: Strict routing enforced for session lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:06] VERBOSE[16653] logger.c: set_destination: Parsing for address/port to send to [Jan 9 16:50:06] VERBOSE[16653] logger.c: set_destination: set destination to 192.168.1.58, port 5060 [Jan 9 16:50:06] VERBOSE[16653] logger.c: Reliably Transmitting (NAT) to 222.111.235.18:1033: BYE sip:iam_friend@192.168.1.58 SIP/2.0 Via: SIP/2.0/UDP 88.99.0.111:5060;branch=z9hG4bK08588a46;rport Max-Forwards: 70 From: ;tag=as6f197a66 To: "Anonimo" ;tag=dggqa Call-ID: lyrmscfuortfsxl@192.168.1.58 CSeq: 102 BYE User-Agent: Asterisk SVN trunk Content-Length: 0 --- [Jan 9 16:50:06] DEBUG[16653] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #10 [Jan 9 16:50:06] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_friend-08206828 [Jan 9 16:50:06] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_friend [Jan 9 16:50:06] DEBUG[16653] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 9 16:50:06] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_friend-08206828 [Jan 9 16:50:06] DEBUG[16653] chan_sip.c: Checking device state for peer iam_friend-08206828 [Jan 9 16:50:06] DEBUG[16653] devicestate.c: Changing state for SIP/iam_friend-08206828 - state 4 (Invalid) [Jan 9 16:50:06] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_friend [Jan 9 16:50:06] DEBUG[16653] chan_sip.c: Checking device state for peer iam_friend [Jan 9 16:50:06] DEBUG[16653] devicestate.c: Changing state for SIP/iam_friend - state 1 (Not in use) [Jan 9 16:50:06] DEBUG[16653] app_queue.c: Device 'SIP/iam_friend-08206828' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:50:06] DEBUG[16653] app_queue.c: Device 'SIP/iam_friend' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:50:07] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 88.99.0.111:5060;rport=5060;branch=z9hG4bK08588a46 To: "Anonimo" ;tag=dggqa From: ;tag=as6f197a66 Call-ID: lyrmscfuortfsxl@192.168.1.58 CSeq: 102 BYE Server: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.99.0.111:5060;rport=5060;branch=z9hG4bK08588a46 [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 2 [ 56]: To: "Anonimo" ;tag=dggqa [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 3 [ 43]: From: ;tag=as6f197a66 [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 4 [ 37]: Call-ID: lyrmscfuortfsxl@192.168.1.58 [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 6 [ 19]: Server: Twinkle/1.1 [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Header 8 [ 0]: [Jan 9 16:50:07] VERBOSE[16653] logger.c: --- (8 headers 0 lines) --- [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #10 [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: Stopping retransmission on 'lyrmscfuortfsxl@192.168.1.58' of Request 102: Match Found [Jan 9 16:50:07] VERBOSE[16653] logger.c: SIP Response message for INCOMING dialog BYE arrived [Jan 9 16:50:07] VERBOSE[16653] logger.c: Really destroying SIP dialog 'lyrmscfuortfsxl@192.168.1.58' Method: ACK [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: ---------- SIP HISTORY for 'lyrmscfuortfsxl@192.168.1.58' [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: * SIP Call [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: 001. Hangup Cause Unknown [Jan 9 16:50:07] DEBUG[16653] chan_sip.c: ---------- END SIP HISTORY for 'lyrmscfuortfsxl@192.168.1.58' [Jan 9 16:50:13] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> INVITE sip:test@88.99.0.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKlxyxxlne Max-Forwards: 70 To: From: "Anonimo" ;tag=koavl Call-ID: gbgdxgxdjxrczam@192.168.1.58 CSeq: 757 INVITE Contact: Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.1 Content-Length: 307 v=0 o=iam_peer 616947736 53766452 IN IP4 192.168.1.58 s=- c=IN IP4 192.168.1.58 t=0 0 m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 <-------------> [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 0 [ 35]: INVITE sip:test@88.99.0.111 SIP/2.0 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKlxyxxlne [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 3 [ 26]: To: [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 4 [ 56]: From: "Anonimo" ;tag=koavl [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 5 [ 37]: Call-ID: gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 6 [ 16]: CSeq: 757 INVITE [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 7 [ 36]: Contact: [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 9 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 10 [ 37]: Supported: replaces,norefersub,100rel [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 11 [ 23]: User-Agent: Twinkle/1.1 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 12 [ 19]: Content-Length: 307 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 13 [ 0]: [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 0 [ 3]: v=0 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 1 [ 49]: o=iam_peer 616947736 53766452 IN IP4 192.168.1.58 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 2 [ 3]: s=- [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.1.58 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 5 [ 36]: m=audio 8000 RTP/AVP 98 97 8 0 3 101 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 6 [ 23]: a=rtpmap:98 speex/16000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 7 [ 22]: a=rtpmap:97 speex/8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 8 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 9 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 10 [ 19]: a=rtpmap:3 GSM/8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 11 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 12 [ 15]: a=fmtp:101 0-15 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Body 13 [ 10]: a=ptime:20 [Jan 9 16:50:13] VERBOSE[16653] logger.c: --- (13 headers 14 lines) --- [Jan 9 16:50:13] DEBUG[16653] acl.c: Found IP address for this socket [Jan 9 16:50:13] VERBOSE[16653] logger.c: == Using SIP RTP CoS mark 5 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Setting NAT on RTP to Off [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Allocating new SIP dialog for gbgdxgxdjxrczam@192.168.1.58 - INVITE (With RTP) [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel" [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Found SIP option: -replaces- [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Matched SIP option: replaces [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Found SIP option: -norefersub- [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Matched SIP option: norefersub [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Found SIP option: -100rel- [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Matched SIP option: 100rel [Jan 9 16:50:13] VERBOSE[16653] logger.c: Sending to 222.111.235.18 : 1033 (NAT) [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Initializing initreq for method INVITE - callid gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Using INVITE request as basis request - gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:13] VERBOSE[16653] logger.c: No user 'iam_peer' in SIP users list [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found peer 'iam_peer' for 'iam_peer' from 222.111.235.18:1033 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Setting NAT on RTP to On [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found RTP audio format 98 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found RTP audio format 97 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found RTP audio format 8 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found RTP audio format 0 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found RTP audio format 3 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found RTP audio format 101 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Peer audio RTP is at port 192.168.1.58:8000 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found audio description format speex for ID 98 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found audio description format speex for ID 97 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found audio description format PCMA for ID 8 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found audio description format PCMU for ID 0 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found audio description format GSM for ID 3 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Found audio description format telephone-event for ID 101 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Got unsupported a:fmtp in SDP offer [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: T38 state changed to 0 on channel [Jan 9 16:50:13] VERBOSE[16653] logger.c: Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 9 16:50:13] VERBOSE[16653] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 9 16:50:13] VERBOSE[16653] logger.c: Peer audio RTP is at port 192.168.1.58:8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Checking SIP call limits for device iam_peer [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Updating call counter for incoming call [Jan 9 16:50:13] VERBOSE[16653] logger.c: Looking for test in test (domain 88.99.0.111) [Jan 9 16:50:13] DEBUG[16653] frame.c: Could not find preferred codec - Going for the best codec [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: *** Our capabilities are 0x8000e (gsm|ulaw|alaw|h263) [Jan 9 16:50:13] DEBUG[16653] frame.c: Could not find preferred codec - Going for the best codec [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: This channel will not be able to handle video. [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: build_route: Contact hop: [Jan 9 16:50:13] VERBOSE[16653] logger.c: list_route: hop: [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: SIP/iam_peer-08206970: New call is still down.... Trying... [Jan 9 16:50:13] VERBOSE[16653] logger.c: <--- Transmitting (NAT) to 222.111.235.18:1033 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKlxyxxlne;received=222.111.235.18;rport=1033 From: "Anonimo" ;tag=koavl To: Call-ID: gbgdxgxdjxrczam@192.168.1.58 CSeq: 757 INVITE User-Agent: Asterisk SVN trunk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Jan 9 16:50:13] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer-08206970 [Jan 9 16:50:13] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer [Jan 9 16:50:13] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer-08206970 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer-08206970 [Jan 9 16:50:13] DEBUG[16653] pbx.c: Launching 'Playback' [Jan 9 16:50:13] VERBOSE[16653] logger.c: -- Executing [test@test:1] Playback("SIP/iam_peer-08206970", "demo-thanks") in new stack [Jan 9 16:50:13] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer-08206970 [Jan 9 16:50:13] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: SIP answering channel: SIP/iam_peer-08206970 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Setting framing from config on incoming call [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 9 16:50:13] VERBOSE[16653] logger.c: Audio is at 88.99.0.111 port 10244 [Jan 9 16:50:13] VERBOSE[16653] logger.c: Adding codec 0x2 (gsm) to SDP [Jan 9 16:50:13] VERBOSE[16653] logger.c: Adding codec 0x4 (ulaw) to SDP [Jan 9 16:50:13] VERBOSE[16653] logger.c: Adding codec 0x8 (alaw) to SDP [Jan 9 16:50:13] VERBOSE[16653] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: -- Done with adding codecs to SDP [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 9 16:50:13] VERBOSE[16653] logger.c: <--- Reliably Transmitting (NAT) to 222.111.235.18:1033 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.58;branch=z9hG4bKlxyxxlne;received=222.111.235.18;rport=1033 From: "Anonimo" ;tag=koavl To: ;tag=as587a2c54 Call-ID: gbgdxgxdjxrczam@192.168.1.58 CSeq: 757 INVITE User-Agent: Asterisk SVN trunk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 315 v=0 o=root 105751118 105751118 IN IP4 88.99.0.111 s=Asterisk PBX SVN-trunk-r96988 c=IN IP4 88.99.0.111 t=0 0 m=audio 10244 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #11 [Jan 9 16:50:13] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> ACK sip:test@88.99.0.111 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKxsmyaqwg Max-Forwards: 70 To: ;tag=as587a2c54 From: "Anonimo" ;tag=koavl Call-ID: gbgdxgxdjxrczam@192.168.1.58 CSeq: 757 ACK User-Agent: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 0 [ 32]: ACK sip:test@88.99.0.111 SIP/2.0 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 1 [ 58]: Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKxsmyaqwg [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 3 [ 41]: To: ;tag=as587a2c54 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 4 [ 56]: From: "Anonimo" ;tag=koavl [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 5 [ 37]: Call-ID: gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 6 [ 13]: CSeq: 757 ACK [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 7 [ 23]: User-Agent: Twinkle/1.1 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Header 9 [ 0]: [Jan 9 16:50:13] VERBOSE[16653] logger.c: --- (9 headers 0 lines) --- [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #11 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Stopping retransmission on 'gbgdxgxdjxrczam@192.168.1.58' of Response 757: Match Found [Jan 9 16:50:13] DEBUG[16653] rtp.c: RTCP NAT: Got RTCP from other end. Now sending to address 222.111.235.18:8001 [Jan 9 16:50:13] DEBUG[16653] rtp.c: Got RTCP report of 40 bytes [Jan 9 16:50:13] DEBUG[16653] rtp.c: RTP NAT: Got audio from other end. Now sending to address 222.111.235.18:8000 [Jan 9 16:50:13] DEBUG[16653] chan_sip.c: Oooh, format changed to 2 gsm [Jan 9 16:50:13] DEBUG[16653] channel.c: Set channel SIP/iam_peer-08206970 to read format ulaw [Jan 9 16:50:13] DEBUG[16653] channel.c: Set channel SIP/iam_peer-08206970 to write format ulaw [Jan 9 16:50:13] DEBUG[16653] channel.c: Set channel SIP/iam_peer-08206970 to write format alaw [Jan 9 16:50:13] DEBUG[16653] rtp.c: Ooh, format changed from unknown to gsm [Jan 9 16:50:13] DEBUG[16653] rtp.c: Created smoother: format: 2 ms: 20 len: 33 [Jan 9 16:50:13] VERBOSE[16653] logger.c: -- Playing 'demo-thanks.alaw' (language 'es') [Jan 9 16:50:14] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer-08206970 - state 4 (Invalid) [Jan 9 16:50:14] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer [Jan 9 16:50:14] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer [Jan 9 16:50:14] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer - state 1 (Not in use) [Jan 9 16:50:14] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer-08206970 [Jan 9 16:50:14] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer-08206970 [Jan 9 16:50:14] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer-08206970' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:50:14] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:50:14] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer-08206970 - state 4 (Invalid) [Jan 9 16:50:14] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer [Jan 9 16:50:14] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer [Jan 9 16:50:14] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer - state 1 (Not in use) [Jan 9 16:50:14] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer-08206970' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:50:14] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:50:17] DEBUG[16653] channel.c: Set channel SIP/iam_peer-08206970 to write format ulaw [Jan 9 16:50:17] VERBOSE[16653] logger.c: -- Auto fallthrough, channel 'SIP/iam_peer-08206970' status is 'UNKNOWN' [Jan 9 16:50:17] DEBUG[16653] channel.c: Soft-Hanging up channel 'SIP/iam_peer-08206970' [Jan 9 16:50:17] DEBUG[16653] channel.c: Hanging up channel 'SIP/iam_peer-08206970' [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Hangup call SIP/iam_peer-08206970, SIP callid gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:17] VERBOSE[16653] logger.c: Scheduling destruction of SIP dialog 'gbgdxgxdjxrczam@192.168.1.58' in 32000 ms (Method: ACK) [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Strict routing enforced for session gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:17] VERBOSE[16653] logger.c: set_destination: Parsing for address/port to send to [Jan 9 16:50:17] VERBOSE[16653] logger.c: set_destination: set destination to 192.168.1.58, port 5060 [Jan 9 16:50:17] VERBOSE[16653] logger.c: Reliably Transmitting (NAT) to 222.111.235.18:1033: BYE sip:iam_peer@192.168.1.58 SIP/2.0 Via: SIP/2.0/UDP 88.99.0.111:5060;branch=z9hG4bK2e454b07;rport Max-Forwards: 70 From: ;tag=as587a2c54 To: "Anonimo" ;tag=koavl Call-ID: gbgdxgxdjxrczam@192.168.1.58 CSeq: 102 BYE User-Agent: Asterisk SVN trunk Content-Length: 0 --- [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Jan 9 16:50:17] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer-08206970 [Jan 9 16:50:17] DEBUG[16653] devicestate.c: Notification of state change to be queued on device/channel SIP/iam_peer [Jan 9 16:50:17] DEBUG[16653] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Jan 9 16:50:17] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer-08206970 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer-08206970 [Jan 9 16:50:17] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer-08206970 - state 4 (Invalid) [Jan 9 16:50:17] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer-08206970' changed to state '4' (Invalid) but we don't care because they're not a member of any queue. [Jan 9 16:50:17] DEBUG[16653] devicestate.c: No provider found, checking channel drivers for SIP - iam_peer [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Checking device state for peer iam_peer [Jan 9 16:50:17] DEBUG[16653] devicestate.c: Changing state for SIP/iam_peer - state 1 (Not in use) [Jan 9 16:50:17] DEBUG[16653] app_queue.c: Device 'SIP/iam_peer' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Jan 9 16:50:17] VERBOSE[16653] logger.c: <--- SIP read from 222.111.235.18:1033 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 88.99.0.111:5060;rport=5060;branch=z9hG4bK2e454b07 To: "Anonimo" ;tag=koavl From: ;tag=as587a2c54 Call-ID: gbgdxgxdjxrczam@192.168.1.58 CSeq: 102 BYE Server: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 1 [ 67]: Via: SIP/2.0/UDP 88.99.0.111:5060;rport=5060;branch=z9hG4bK2e454b07 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 2 [ 54]: To: "Anonimo" ;tag=koavl [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 3 [ 43]: From: ;tag=as587a2c54 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 4 [ 37]: Call-ID: gbgdxgxdjxrczam@192.168.1.58 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 6 [ 19]: Server: Twinkle/1.1 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 7 [ 17]: Content-Length: 0 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Header 8 [ 0]: [Jan 9 16:50:17] VERBOSE[16653] logger.c: --- (8 headers 0 lines) --- [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: Stopping retransmission on 'gbgdxgxdjxrczam@192.168.1.58' of Request 102: Match Found [Jan 9 16:50:17] VERBOSE[16653] logger.c: SIP Response message for INCOMING dialog BYE arrived [Jan 9 16:50:17] VERBOSE[16653] logger.c: Really destroying SIP dialog 'gbgdxgxdjxrczam@192.168.1.58' Method: ACK [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: ---------- SIP HISTORY for 'gbgdxgxdjxrczam@192.168.1.58' [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: * SIP Call [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: 001. Hangup Cause Unknown [Jan 9 16:50:17] DEBUG[16653] chan_sip.c: ---------- END SIP HISTORY for 'gbgdxgxdjxrczam@192.168.1.58'