[Jan 28 22:06:22] NOTICE[11844] cdr.c: CDR logging disabled, data will be lost. [Jan 28 22:06:22] NOTICE[11844] loader.c: 64 modules will be loaded. [Jan 28 22:06:22] NOTICE[11844] config.c: Registered Config Engine mysql [Jan 28 22:06:22] VERBOSE[11844] chan_sip.c: SIP channel loading... [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: <--- SIP read from UDP://78.105.1.131:5061 ---> INVITE sip:123@dev-sip.tele500.com SIP/2.0 Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKlgcskymb Max-Forwards: 70 To: From: "10000" ;tag=ylgrq Call-ID: avowlzfudtcoktt@78.105.1.131 CSeq: 508 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: 251 v=0 o=10000 2015682806 899660237 IN IP4 78.105.1.131 s=- c=IN IP4 78.105.1.131 t=0 0 m=audio 8002 RTP/AVP 8 0 3 101 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 28 22:06:34] DEBUG[11852] chan_sip.c: Header 0 [ 42]: INVITE sip:123@dev-sip.tele500.com SIP/2.0 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKlgcskymb [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 3 [ 33]: To: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 4 [ 55]: From: "10000" ;tag=ylgrq [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 5 [ 37]: Call-ID: avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 6 [ 16]: CSeq: 508 INVITE [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 7 [ 38]: Contact: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 8 [ 29]: Content-Type: application/sdp [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 9 [ 78]: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 10 [ 37]: Supported: replaces,norefersub,100rel [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 11 [ 23]: User-Agent: Twinkle/1.1 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 12 [ 19]: Content-Length: 251 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 13 [ 0]: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 0 [ 3]: v=0 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 1 [ 48]: o=10000 2015682806 899660237 IN IP4 78.105.1.131 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 2 [ 3]: s=- [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 3 [ 21]: c=IN IP4 78.105.1.131 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 5 [ 30]: m=audio 8002 RTP/AVP 8 0 3 101 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-15 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Body 11 [ 10]: a=ptime:20 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: --- (13 headers 12 lines) --- [Jan 28 22:06:34] DEBUG[11852] acl.c: Found IP address for this socket [Jan 28 22:06:34] VERBOSE[11852] netsock.c: == Using SIP RTP CoS mark 5 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Setting NAT on RTP to Off [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Allocating new SIP dialog for avowlzfudtcoktt@78.105.1.131 - INVITE (With RTP) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Begin: parsing SIP "Supported: replaces,norefersub,100rel" [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Found SIP option: -replaces- [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Matched SIP option: replaces [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Found SIP option: -norefersub- [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Matched SIP option: norefersub [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Found SIP option: -100rel- [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Matched SIP option: 100rel [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Sending to 78.105.1.131 : 5061 (NAT) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Initializing initreq for method INVITE - callid avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Using INVITE request as basis request - avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: No matching peer for '10000' from '78.105.1.131:5061' [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found RTP audio format 8 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found RTP audio format 0 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found RTP audio format 3 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found RTP audio format 101 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Peer audio RTP is at port 78.105.1.131:8002 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found audio description format PCMA for ID 8 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found audio description format PCMU for ID 0 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found audio description format GSM for ID 3 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Found audio description format telephone-event for ID 101 [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Got unsupported a:fmtp in SDP offer [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Capabilities: us - 0x140f (g723|gsm|ulaw|alaw|ilbc|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Peer audio RTP is at port 78.105.1.131:8002 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: We're settling with these formats: 0xe (gsm|ulaw|alaw) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Checking SIP call limits for device [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Updating call counter for incoming call [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: Looking for 123 in common (domain dev-sip.tele500.com) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: *** Our native formats are 0x8 (alaw) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: *** Joint capabilities are 0xe (gsm|ulaw|alaw) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: *** Our capabilities are 0x140f (g723|gsm|ulaw|alaw|ilbc|g722) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw) [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: This channel will not be able to handle video. [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: build_route: Contact hop: [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: list_route: hop: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Session timer started: 1 - avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: SIP/dev-sip.tele500.com-081faf48: New call is still down.... Trying... [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: <--- Transmitting (NAT) to 78.105.1.131:5061 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKlgcskymb;received=78.105.1.131;rport=5061 From: "10000" ;tag=ylgrq To: Call-ID: avowlzfudtcoktt@78.105.1.131 CSeq: 508 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKlgcskymb;received=78.105.1.131;rport=5061 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 2 [ 55]: From: "10000" ;tag=ylgrq [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 3 [ 33]: To: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 4 [ 37]: Call-ID: avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 5 [ 16]: CSeq: 508 INVITE [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 6 [ 18]: Server: Media GW 1 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 9 [ 31]: Contact: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 11 [ 0]: [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 78.105.1.131:5061 [Jan 28 22:06:34] DEBUG[11849] devicestate.c: No provider found, checking channel drivers for SIP - dev-sip.tele500.com [Jan 28 22:06:34] DEBUG[11849] chan_sip.c: Checking device state for peer dev-sip.tele500.com [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11856] pbx.c: Launching 'Answer' [Jan 28 22:06:34] VERBOSE[11856] pbx.c: -- Executing [123@common:1] Answer("SIP/dev-sip.tele500.com-081faf48", "") in new stack [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11849] channel.c: Avoiding initial deadlock for channel '0x8204de0' [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: SIP answering channel: SIP/dev-sip.tele500.com-081faf48 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Setting framing from config on incoming call [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: ** Our capability: 0xe (gsm|ulaw|alaw) Video flag: True Text flag: True [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [Jan 28 22:06:34] VERBOSE[11856] chan_sip.c: Audio is at 78.105.1.127 port 10534 [Jan 28 22:06:34] VERBOSE[11856] chan_sip.c: Adding codec 0x8 (alaw) to SDP [Jan 28 22:06:34] VERBOSE[11856] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [Jan 28 22:06:34] VERBOSE[11856] chan_sip.c: Adding codec 0x2 (gsm) to SDP [Jan 28 22:06:34] VERBOSE[11856] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: -- Done with adding codecs to SDP [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Done building SDP. Settling with this capability: 0xe (gsm|ulaw|alaw) [Jan 28 22:06:34] VERBOSE[11856] chan_sip.c: <--- Reliably Transmitting (NAT) to 78.105.1.131:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKlgcskymb;received=78.105.1.131;rport=5061 From: "10000" ;tag=ylgrq To: ;tag=as007cd34b Call-ID: avowlzfudtcoktt@78.105.1.131 CSeq: 508 INVITE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 300 v=0 o=root 1803537085 1803537085 IN IP4 78.105.1.127 s=Media GW 1 c=IN IP4 78.105.1.127 t=0 0 m=audio 10534 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKlgcskymb;received=78.105.1.131;rport=5061 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 2 [ 55]: From: "10000" ;tag=ylgrq [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 3 [ 48]: To: ;tag=as007cd34b [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 4 [ 37]: Call-ID: avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 5 [ 16]: CSeq: 508 INVITE [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 6 [ 18]: Server: Media GW 1 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 9 [ 31]: Contact: [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 10 [ 29]: Content-Type: application/sdp [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 11 [ 19]: Content-Length: 300 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Header 12 [ 0]: [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 0 [ 3]: v=0 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 1 [ 48]: o=root 1803537085 1803537085 IN IP4 78.105.1.127 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 2 [ 12]: s=Media GW 1 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 3 [ 21]: c=IN IP4 78.105.1.127 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 4 [ 5]: t=0 0 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 5 [ 31]: m=audio 10534 RTP/AVP 8 0 3 101 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 8 [ 19]: a=rtpmap:3 GSM/8000 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 9 [ 33]: a=rtpmap:101 telephone-event/8000 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 10 [ 15]: a=fmtp:101 0-16 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 11 [ 25]: a=silenceSupp:off - - - - [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 12 [ 10]: a=ptime:20 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Body 13 [ 10]: a=sendrecv [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #2 [Jan 28 22:06:34] DEBUG[11856] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 78.105.1.131:5061 [Jan 28 22:06:34] DEBUG[11849] devicestate.c: Changing state for SIP/dev-sip.tele500.com - state 2 (In use) [Jan 28 22:06:34] DEBUG[11849] devicestate.c: device 'SIP/dev-sip.tele500.com' state '2' [Jan 28 22:06:34] DEBUG[11849] devicestate.c: No provider found, checking channel drivers for SIP - dev-sip.tele500.com [Jan 28 22:06:34] DEBUG[11849] chan_sip.c: Checking device state for peer dev-sip.tele500.com [Jan 28 22:06:34] DEBUG[11849] devicestate.c: Changing state for SIP/dev-sip.tele500.com - state 2 (In use) [Jan 28 22:06:34] DEBUG[11849] devicestate.c: device 'SIP/dev-sip.tele500.com' state '2' [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Processing device state change for 'SIP/dev-sip.tele500.com' [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Adding per-server state of 'In use' for 'SIP/dev-sip.tele500.com' [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Aggregate devstate result is 2 [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Aggregate state for device 'SIP/dev-sip.tele500.com' has changed to 'In use' [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Processing device state change for 'SIP/dev-sip.tele500.com' [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Adding per-server state of 'In use' for 'SIP/dev-sip.tele500.com' [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Aggregate devstate result is 2 [Jan 28 22:06:34] DEBUG[11848] devicestate.c: Aggregate state for device 'SIP/dev-sip.tele500.com' has not changed from 'In use' [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: <--- SIP read from UDP://78.105.1.131:5061 ---> ACK sip:123@78.105.1.127 SIP/2.0 Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKbrzacahy Max-Forwards: 70 To: ;tag=as007cd34b From: "10000" ;tag=ylgrq Call-ID: avowlzfudtcoktt@78.105.1.131 CSeq: 508 ACK User-Agent: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 0 [ 32]: ACK sip:123@78.105.1.127 SIP/2.0 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKbrzacahy [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 3 [ 48]: To: ;tag=as007cd34b [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 4 [ 55]: From: "10000" ;tag=ylgrq [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 5 [ 37]: Call-ID: avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 6 [ 13]: CSeq: 508 ACK [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 7 [ 23]: User-Agent: Twinkle/1.1 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Header 9 [ 0]: [Jan 28 22:06:34] VERBOSE[11852] chan_sip.c: --- (9 headers 0 lines) --- [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #2 [Jan 28 22:06:34] DEBUG[11852] chan_sip.c: Stopping retransmission on 'avowlzfudtcoktt@78.105.1.131' of Response 508: Match Found [Jan 28 22:06:34] DEBUG[11856] pbx.c: Launching 'ExternalIVR' [Jan 28 22:06:34] VERBOSE[11856] pbx.c: -- Executing [123@common:2] ExternalIVR("SIP/dev-sip.tele500.com-081faf48", "/enc/etc/asterisk/ivr/test.sh") in new stack [Jan 28 22:06:34] WARNING[11856] app_externalivr.c: SIP/dev-sip.tele500.com-081faf48: Answering channel and starting generator [Jan 28 22:06:34] DEBUG[11856] app_externalivr.c: got command 'S,/var/lib/asterisk/sounds/demo-congrats' [Jan 28 22:06:34] DEBUG[11856] app_externalivr.c: sent 'T,1233180394' [Jan 28 22:06:34] DEBUG[11856] channel.c: Set channel SIP/dev-sip.tele500.com-081faf48 to write format gsm [Jan 28 22:06:34] DEBUG[11856] rtp.c: Ooh, format changed from unknown to alaw [Jan 28 22:06:34] DEBUG[11856] rtp.c: Created smoother: format: 8 ms: 20 len: 160 [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: Sending dtmf: 49 (1), at 78.105.1.131 [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: Sending dtmf: 49 (1), at 78.105.1.131 [Jan 28 22:06:36] DEBUG[11856] app_externalivr.c: sent '1,1233180396' [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000001 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: Difference is 896, ms is 132 [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000002 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: Sending dtmf: 50 (2), at 78.105.1.131 [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000002 (len = 4) [Jan 28 22:06:36] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000002 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000002 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000002 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: Sending dtmf: 50 (2), at 78.105.1.131 [Jan 28 22:06:37] DEBUG[11856] app_externalivr.c: sent '2,1233180397' [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000002 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: Difference is 1048, ms is 151 [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: Sending dtmf: 51 (3), at 78.105.1.131 [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: Sending dtmf: 51 (3), at 78.105.1.131 [Jan 28 22:06:37] DEBUG[11856] app_externalivr.c: sent '3,1233180397' [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: - RTP 2833 Event: 00000003 (len = 4) [Jan 28 22:06:37] DEBUG[11856] rtp.c: Difference is 1200, ms is 170 [Jan 28 22:06:40] VERBOSE[11852] chan_sip.c: <--- SIP read from UDP://78.105.1.131:5061 ---> BYE sip:123@78.105.1.127 SIP/2.0 Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKdcatnupn Max-Forwards: 70 To: ;tag=as007cd34b From: "10000" ;tag=ylgrq Call-ID: avowlzfudtcoktt@78.105.1.131 CSeq: 509 BYE User-Agent: Twinkle/1.1 Content-Length: 0 <-------------> [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 0 [ 32]: BYE sip:123@78.105.1.127 SIP/2.0 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 78.105.1.131:5061;rport;branch=z9hG4bKdcatnupn [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 3 [ 48]: To: ;tag=as007cd34b [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 4 [ 55]: From: "10000" ;tag=ylgrq [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 5 [ 37]: Call-ID: avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 6 [ 13]: CSeq: 509 BYE [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 7 [ 23]: User-Agent: Twinkle/1.1 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 8 [ 17]: Content-Length: 0 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 9 [ 0]: [Jan 28 22:06:40] VERBOSE[11852] chan_sip.c: --- (9 headers 0 lines) --- [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: **** Received BYE (8) - Command in SIP BYE [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Initializing initreq for method BYE - callid avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:40] VERBOSE[11852] chan_sip.c: Sending to 78.105.1.131 : 5061 (NAT) [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Setting SIP_ALREADYGONE on dialog avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Session timer stopped: -1 - avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Received bye, issuing owner hangup [Jan 28 22:06:40] VERBOSE[11852] chan_sip.c: <--- Transmitting (NAT) to 78.105.1.131:5061 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKdcatnupn;received=78.105.1.131;rport=5061 From: "10000" ;tag=ylgrq To: ;tag=as007cd34b Call-ID: avowlzfudtcoktt@78.105.1.131 CSeq: 509 BYE Server: Media GW 1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 1 [ 90]: Via: SIP/2.0/UDP 78.105.1.131:5061;branch=z9hG4bKdcatnupn;received=78.105.1.131;rport=5061 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 2 [ 55]: From: "10000" ;tag=ylgrq [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 3 [ 48]: To: ;tag=as007cd34b [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 4 [ 37]: Call-ID: avowlzfudtcoktt@78.105.1.131 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 5 [ 13]: CSeq: 509 BYE [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 6 [ 18]: Server: Media GW 1 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 7 [ 66]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 8 [ 26]: Supported: replaces, timer [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 9 [ 31]: Contact: [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 10 [ 17]: Content-Length: 0 [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Header 11 [ 0]: [Jan 28 22:06:40] DEBUG[11852] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 78.105.1.131:5061 [Jan 28 22:06:42] VERBOSE[11855] asterisk.c: -- Remote UNIX connection disconnected *********** test.sh still running !!! **************** ********** KILLING test.sh manualy now *************** [Jan 28 22:07:05] WARNING[11856] app_externalivr.c: SIP/dev-sip.tele500.com-081faf48: Wait failed (No child processes) [Jan 28 22:07:05] DEBUG[11856] pbx.c: Extension 123, priority 2 returned normally even though call was hung up [Jan 28 22:07:05] DEBUG[11856] channel.c: Soft-Hanging up channel 'SIP/dev-sip.tele500.com-081faf48' [Jan 28 22:07:05] DEBUG[11856] channel.c: Hanging up channel 'SIP/dev-sip.tele500.com-081faf48' [Jan 28 22:07:05] DEBUG[11856] chan_sip.c: Hangup call SIP/dev-sip.tele500.com-081faf48, SIP callid avowlzfudtcoktt@78.105.1.131 [Jan 28 22:07:05] DEBUG[11856] cdr.c: Dropping CDR ! [Jan 28 22:07:05] DEBUG[11856] devicestate.c: device 'SIP/dev-sip.tele500.com' state '1' [Jan 28 22:07:05] DEBUG[11848] devicestate.c: Processing device state change for 'SIP/dev-sip.tele500.com' [Jan 28 22:07:05] DEBUG[11848] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/dev-sip.tele500.com' [Jan 28 22:07:05] DEBUG[11848] devicestate.c: Aggregate devstate result is 1 [Jan 28 22:07:05] DEBUG[11848] devicestate.c: Aggregate state for device 'SIP/dev-sip.tele500.com' has changed to 'Not in use' [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: Destroying SIP dialog avowlzfudtcoktt@78.105.1.131 [Jan 28 22:07:05] VERBOSE[11852] chan_sip.c: Really destroying SIP dialog 'avowlzfudtcoktt@78.105.1.131' Method: BYE [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: ---------- SIP HISTORY for 'avowlzfudtcoktt@78.105.1.131' [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: * SIP Call [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 001. Rx INVITE / 508 INVITE / sip:123@dev-sip.tele500.com [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 002. NewChan Channel SIP/dev-sip.tele500.com-081faf48 - from avowlzfudtcoktt [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 003. TxResp SIP/2.0 / 508 INVITE - 100 Trying [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 004. TxRespRel SIP/2.0 / 508 INVITE - 200 OK [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 005. Rx ACK / 508 ACK / sip:123@78.105.1.127 [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 006. Rx BYE / 509 BYE / sip:123@78.105.1.127 [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 007. RTCPaudio Quality:ssrc=908181293;themssrc=3030204736;rxjitter=0.021304;rx [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 008. RTCPaudioJitter Quality:rxjitter=0.021304; [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 009. RTCPaudioLoss Quality:lost=2;expected=297; [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 010. RTCPaudioRTT Quality:Not available [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 011. TxResp SIP/2.0 / 509 BYE - 200 OK [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: 012. Hangup Cause Unknown [Jan 28 22:07:05] DEBUG[11852] chan_sip.c: ---------- END SIP HISTORY for 'avowlzfudtcoktt@78.105.1.131' ********** test.sh KILLED ***************