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- [Jul 18 16:12:06] VERBOSE[24157] config.c: == Parsing '/etc/asterisk/logger.conf': [Jul 18 16:12:06] DEBUG[24157] config.c: Parsing /etc/asterisk/logger.conf
- [Jul 18 16:12:06] VERBOSE[24157] config.c: == Found
- [Jul 18 16:12:06] VERBOSE[24157] logger.c: Asterisk Event Logger restarted
- [Jul 18 16:12:06] VERBOSE[24157] logger.c: Asterisk Queue Logger restarted
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c:
- <--- SIP read from UDP:192.168.20.1:5060 --->
- INVITE sip:300@192.168.1.40 SIP/2.0
- Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9
- From: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- To: <sip:300@192.168.1.40>
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 1798690992 INVITE
- Contact: "Mushtaq" <sip:254@192.168.20.1:5060;transport=udp>
- Max-Forwards: 70
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
- Content-Type: application/sdp
- Supported: timer
- Content-Length: 275
- v=0
- o=UserA 1459024281 4215621787 IN IP4 192.168.20.1
- s=Session SDP
- c=IN IP4 192.168.20.1
- t=0 0
- m=audio 49154 RTP/AVP 8 0 18 101
- a=rtpmap:8 PCMA/8000
- a=rtpmap:0 PCMU/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- <------------->
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 0 [ 35]: INVITE sip:300@192.168.1.40 SIP/2.0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 2 [ 59]: From: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 3 [ 26]: To: <sip:300@192.168.1.40>
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 5 [ 23]: CSeq: 1798690992 INVITE
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 6 [ 60]: Contact: "Mushtaq" <sip:254@192.168.20.1:5060;transport=udp>
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 7 [ 16]: Max-Forwards: 70
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 8 [ 69]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 9 [ 29]: Content-Type: application/sdp
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 10 [ 16]: Supported: timer
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 11 [ 19]: Content-Length: 275
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 12 [ 0]:
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 0 [ 3]: v=0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 1 [ 49]: o=UserA 1459024281 4215621787 IN IP4 192.168.20.1
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 2 [ 13]: s=Session SDP
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 3 [ 21]: c=IN IP4 192.168.20.1
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 4 [ 5]: t=0 0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 5 [ 32]: m=audio 49154 RTP/AVP 8 0 18 101
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 6 [ 20]: a=rtpmap:8 PCMA/8000
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 8 [ 21]: a=rtpmap:18 G729/8000
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 9 [ 19]: a=fmtp:18 annexb=no
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Body 11 [ 15]: a=fmtp:101 0-15
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: --- (12 headers 12 lines) ---
- [Jul 18 16:13:15] DEBUG[27072] acl.c: Found IP address for this socket
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.1.40:5060
- [Jul 18 16:13:15] VERBOSE[27072] netsock.c: == Using SIP RTP CoS mark 5
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Setting NAT on RTP to On
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Allocating new SIP dialog for 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1 - INVITE (With RTP)
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Begin: parsing SIP "Supported: timer"
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Found SIP option: -timer-
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Matched SIP option: timer
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Sending to 192.168.20.1 : 5060 (NAT)
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Initializing initreq for method INVITE - callid 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Using INVITE request as basis request - 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found peer 'avaya' for '254' from 192.168.20.1:5060
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Setting NAT on RTP to On
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP o=UserA 1459024281 4215621787 IN IP4 192.168.20.1... UNSUPPORTED.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP s=Session SDP... UNSUPPORTED.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.20.1... OK.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 8
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 0
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 18
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found RTP audio format 101
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format PCMA for ID 8
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format PCMU for ID 0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format G729 for ID 18
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... UNSUPPORTED.
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Found audio description format telephone-event for ID 101
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Peer audio RTP is at port 192.168.20.1:49154
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: We're settling with these formats: 0x10c (ulaw|alaw|g729)
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Checking SIP call limits for device
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Updating call counter for incoming call
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: Looking for 300 in access (domain 192.168.1.40)
- [Jul 18 16:13:15] DEBUG[27072] frame.c: Could not find preferred codec - Going for the best codec
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** Our native formats are 0x4 (ulaw)
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** Joint capabilities are 0x10c (ulaw|alaw|g729)
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** Our capabilities are 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14)
- [Jul 18 16:13:15] DEBUG[27072] frame.c: Could not find preferred codec - Going for the best codec
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw)
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: This channel will not be able to handle video.
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: build_route: Contact hop: "Mushtaq" <sip:254@192.168.20.1:5060;transport=udp>
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: list_route: hop: <sip:254@192.168.20.1:5060;transport=udp>
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: SIP/avaya-000000d5: New call is still down.... Trying...
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c:
- <--- Transmitting (NAT) to 192.168.20.1:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9;received=192.168.20.1;rport=5060
- From: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- To: <sip:300@192.168.1.40>
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 1798690992 INVITE
- Server: Asterisk PBX 1.6.2.22
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:300@192.168.1.40>
- Content-Length: 0
- <------------>
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.20.1:5060
- [Jul 18 16:13:15] DEBUG[27060] devicestate.c: No provider found, checking channel drivers for SIP - avaya
- [Jul 18 16:13:15] DEBUG[27060] chan_sip.c: Checking device state for peer avaya
- [Jul 18 16:13:15] DEBUG[27060] devicestate.c: Changing state for SIP/avaya - state 1 (Not in use)
- [Jul 18 16:13:15] DEBUG[27060] devicestate.c: device 'SIP/avaya' state '1'
- [Jul 18 16:13:15] DEBUG[27089] app_queue.c: Device 'SIP/avaya' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Jul 18 16:13:15] DEBUG[25493] pbx.c: Launching 'Answer'
- [Jul 18 16:13:15] VERBOSE[25493] pbx.c: -- Executing [300@access:1] Answer("SIP/avaya-000000d5", "") in new stack
- [Jul 18 16:13:15] DEBUG[27060] devicestate.c: No provider found, checking channel drivers for SIP - avaya
- [Jul 18 16:13:15] DEBUG[27060] chan_sip.c: Checking device state for peer avaya
- [Jul 18 16:13:15] DEBUG[27060] devicestate.c: Changing state for SIP/avaya - state 1 (Not in use)
- [Jul 18 16:13:15] DEBUG[27060] devicestate.c: device 'SIP/avaya' state '1'
- [Jul 18 16:13:15] DEBUG[27089] app_queue.c: Device 'SIP/avaya' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: SIP answering channel: SIP/avaya-000000d5
- [Jul 18 16:13:15] DEBUG[25493] rtp.c: Setting the marker bit due to a source update
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Setting framing from config on incoming call
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: ** Our capability: 0x10c (ulaw|alaw|g729) Video flag: True Text flag: True
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
- [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Audio is at 192.168.1.40 port 11258
- [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
- [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding codec 0x8 (alaw) to SDP
- [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding codec 0x100 (g729) to SDP
- [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: -- Done with adding codecs to SDP
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Done building SDP. Settling with this capability: 0x10c (ulaw|alaw|g729)
- [Jul 18 16:13:15] VERBOSE[25493] chan_sip.c:
- <--- Reliably Transmitting (NAT) to 192.168.20.1:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 192.168.20.1:5060;branch=z9hG4bK88046fb1ce64815dcca8ce34281014b9;received=192.168.20.1;rport=5060
- From: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- To: <sip:300@192.168.1.40>;tag=as09decdab
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 1798690992 INVITE
- Server: Asterisk PBX 1.6.2.22
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:300@192.168.1.40>
- Content-Type: application/sdp
- Content-Length: 306
- v=0
- o=root 805920992 805920992 IN IP4 192.168.1.40
- s=Asterisk PBX 1.6.2.22
- c=IN IP4 192.168.1.40
- t=0 0
- m=audio 11258 RTP/AVP 0 8 18 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:18 G729/8000
- a=fmtp:18 annexb=no
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------>
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25179
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.20.1:5060
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c:
- <--- SIP read from UDP:192.168.20.1:5060 --->
- ACK sip:300@192.168.1.40 SIP/2.0
- Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK2295d3be4d069b089fe3bba9eb3606b3
- From: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- To: <sip:300@192.168.1.40>;tag=as09decdab
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 1798690992 ACK
- Max-Forwards: 70
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
- Content-Length: 0
- <------------->
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 0 [ 32]: ACK sip:300@192.168.1.40 SIP/2.0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 1 [ 87]: Via: SIP/2.0/UDP 192.168.20.1:5060;rport;branch=z9hG4bK2295d3be4d069b089fe3bba9eb3606b3
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 2 [ 59]: From: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 3 [ 41]: To: <sip:300@192.168.1.40>;tag=as09decdab
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 5 [ 20]: CSeq: 1798690992 ACK
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 6 [ 16]: Max-Forwards: 70
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 7 [ 69]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 8 [ 17]: Content-Length: 0
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Header 9 [ 0]:
- [Jul 18 16:13:15] VERBOSE[27072] chan_sip.c: --- (9 headers 0 lines) ---
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25179
- [Jul 18 16:13:15] DEBUG[27072] chan_sip.c: Stopping retransmission on '6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1' of Response 1798690992: Match Found
- [Jul 18 16:13:15] DEBUG[25493] chan_sip.c: Oooh, format changed to 8 alaw
- [Jul 18 16:13:15] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to read format ulaw
- [Jul 18 16:13:15] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
- [Jul 18 16:13:15] DEBUG[25493] pbx.c: Launching 'Set'
- [Jul 18 16:13:15] VERBOSE[25493] pbx.c: -- Executing [300@access:2] Set("SIP/avaya-000000d5", "CallId=6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1") in new stack
- [Jul 18 16:13:15] DEBUG[25493] pbx.c: Launching 'AGI'
- [Jul 18 16:13:15] VERBOSE[25493] pbx.c: -- Executing [300@access:3] AGI("SIP/avaya-000000d5", "agi://192.168.1.41/testtreatment,inbound") in new stack
- [Jul 18 16:13:15] DEBUG[25493] res_agi.c: Wow, connected!
- [Jul 18 16:13:15] NOTICE[25493] channel.c: Dropping incompatible voice frame on SIP/avaya-000000d5 of format ulaw since our native format has changed to 0x8 (alaw)
- [Jul 18 16:13:16] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
- [Jul 18 16:13:16] DEBUG[25493] format_wav.c: Skipping unknown block 'fact'
- [Jul 18 16:13:16] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/Welcome' (escape_digits=) (sample_offset 0)
- [Jul 18 16:13:16] DEBUG[25493] rtp.c: Ooh, format changed from unknown to alaw
- [Jul 18 16:13:16] DEBUG[25493] rtp.c: Created smoother: format: 8 ms: 20 len: 160
- [Jul 18 16:13:16] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
- [Jul 18 16:13:19] DEBUG[25493] rtp.c: Got RTCP report of 88 bytes
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (73 requested / 73 actual) timer ticks per second
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
- [Jul 18 16:13:20] DEBUG[25493] format_wav.c: Skipping unknown block 'fact'
- [Jul 18 16:13:20] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/LanguageSelection' (escape_digits=0123456789*#) (sample_offset 0)
- [Jul 18 16:13:20] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
- [Jul 18 16:13:23] DEBUG[25493] rtp.c: Got RTCP report of 88 bytes
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: Sending dtmf: 49 (1), at 192.168.20.1
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: Sending dtmf: 49 (1), at 192.168.20.1
- [Jul 18 16:13:26] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:26] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:26] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000001 (len = 4)
- [Jul 18 16:13:26] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
- [Jul 18 16:13:26] DEBUG[25493] format_wav.c: Skipping unknown block 'fact'
- [Jul 18 16:13:26] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/E_MainMenu' (escape_digits=0123456789*#) (sample_offset 0)
- [Jul 18 16:13:26] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: Sending dtmf: 48 (0), at 192.168.20.1
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: Sending dtmf: 48 (0), at 192.168.20.1
- [Jul 18 16:13:29] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:29] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:29] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] rtp.c: - RTP 2833 Event: 00000000 (len = 4)
- [Jul 18 16:13:29] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format slin
- [Jul 18 16:13:29] DEBUG[25493] format_wav.c: Skipping unknown block 'LIST'
- [Jul 18 16:13:29] VERBOSE[25493] res_agi.c: -- Playing '/var/lib/asterisk/sounds/AHI/E_TransferToAgent' (escape_digits=) (sample_offset 0)
- [Jul 18 16:13:29] DEBUG[25493] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (52 requested / 52 actual) timer ticks per second
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Set channel SIP/avaya-000000d5 to write format ulaw
- [Jul 18 16:13:34] VERBOSE[25493] res_agi.c: -- AGI Script Executing Application: (Transfer) Options: (SIP/avaya/6000)
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: SIP transfer of 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1 to avaya/6000
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Strict routing enforced for session 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: Parsing <sip:254@192.168.20.1:5060;transport=udp> for address/port to send to
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: set destination to 192.168.20.1, port 5060
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: Reliably Transmitting (NAT) to 192.168.20.1:5060:
- REFER sip:254@192.168.20.1:5060;transport=udp SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK6c4d8931;rport
- Max-Forwards: 70
- From: <sip:300@192.168.1.40>;tag=as09decdab
- To: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- Contact: <sip:300@192.168.1.40>
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 102 REFER
- User-Agent: Asterisk PBX 1.6.2.22
- Refer-To: <sip:avaya/6000@192.168.1.40>
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Referred-By: <sip:300@192.168.1.40>
- Content-Length: 0
- ---
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25181
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Trying to put 'REFER sip:2' onto UDP socket destined for 192.168.20.1:5060
- [Jul 18 16:13:34] VERBOSE[25493] res_agi.c: -- AGI Script Executing Application: (NoOp) Options: (${TRANSFERSTATUS})
- [Jul 18 16:13:34] VERBOSE[25493] res_agi.c: -- <SIP/avaya-000000d5>AGI Script agi://192.168.1.41/testtreatment completed, returning 0
- [Jul 18 16:13:34] VERBOSE[25493] pbx.c: -- Auto fallthrough, channel 'SIP/avaya-000000d5' status is 'UNKNOWN'
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Soft-Hanging up channel 'SIP/avaya-000000d5'
- [Jul 18 16:13:34] DEBUG[25493] channel.c: Hanging up channel 'SIP/avaya-000000d5'
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Hangup call SIP/avaya-000000d5, SIP callid 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: Scheduling destruction of SIP dialog '6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1' in 32000 ms (Method: ACK)
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Strict routing enforced for session 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: Parsing <sip:254@192.168.20.1:5060;transport=udp> for address/port to send to
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: set_destination: set destination to 192.168.20.1, port 5060
- [Jul 18 16:13:34] VERBOSE[25493] chan_sip.c: Reliably Transmitting (NAT) to 192.168.20.1:5060:
- BYE sip:254@192.168.20.1:5060;transport=udp SIP/2.0
- Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK42599c30;rport
- Max-Forwards: 70
- From: <sip:300@192.168.1.40>;tag=as09decdab
- To: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 103 BYE
- User-Agent: Asterisk PBX 1.6.2.22
- X-Asterisk-HangupCause: Unknown
- X-Asterisk-HangupCauseCode: 0
- Content-Length: 0
- ---
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #25183
- [Jul 18 16:13:34] DEBUG[25493] chan_sip.c: Trying to put 'BYE sip:254' onto UDP socket destined for 192.168.20.1:5060
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c:
- <--- SIP read from UDP:192.168.20.1:5060 --->
- SIP/2.0 481 Dialog/Transaction Does Not Exist
- Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK6c4d8931;rport
- From: <sip:300@192.168.1.40>;tag=as09decdab
- To: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 102 REFER
- Supported: timer
- Content-Length: 0
- <------------->
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 0 [ 45]: SIP/2.0 481 Dialog/Transaction Does Not Exist
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK6c4d8931;rport
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 2 [ 43]: From: <sip:300@192.168.1.40>;tag=as09decdab
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 3 [ 57]: To: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 5 [ 15]: CSeq: 102 REFER
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 6 [ 16]: Supported: timer
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 7 [ 17]: Content-Length: 0
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 8 [ 0]:
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: --- (8 headers 0 lines) ---
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25181
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Stopping retransmission on '6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1' of Request 102: Match Found
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: SIP Response message for INCOMING dialog REFER arrived
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Remote host can't match request REFER to call '6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1'. Giving up
- [Jul 18 16:13:34] DEBUG[27060] devicestate.c: No provider found, checking channel drivers for SIP - avaya
- [Jul 18 16:13:34] DEBUG[27060] chan_sip.c: Checking device state for peer avaya
- [Jul 18 16:13:34] DEBUG[27060] devicestate.c: Changing state for SIP/avaya - state 1 (Not in use)
- [Jul 18 16:13:34] DEBUG[27060] devicestate.c: device 'SIP/avaya' state '1'
- [Jul 18 16:13:34] DEBUG[27089] app_queue.c: Device 'SIP/avaya' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c:
- <--- SIP read from UDP:192.168.20.1:5060 --->
- SIP/2.0 200 Ok
- Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK42599c30;rport
- From: <sip:300@192.168.1.40>;tag=as09decdab
- To: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- CSeq: 103 BYE
- Supported: timer
- Content-Length: 0
- <------------->
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 Ok
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.168.1.40:5060;branch=z9hG4bK42599c30;rport
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 2 [ 43]: From: <sip:300@192.168.1.40>;tag=as09decdab
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 3 [ 57]: To: "Mushtaq" <sip:254@192.168.1.40>;tag=2d5dbb8afb9c01c0
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 4 [ 54]: Call-ID: 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 5 [ 13]: CSeq: 103 BYE
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 6 [ 16]: Supported: timer
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 7 [ 17]: Content-Length: 0
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Header 8 [ 0]:
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: --- (8 headers 0 lines) ---
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #25183
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Stopping retransmission on '6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1' of Request 103: Match Found
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
- [Jul 18 16:13:34] DEBUG[27072] chan_sip.c: Destroying SIP dialog 6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1
- [Jul 18 16:13:34] VERBOSE[27072] chan_sip.c: Really destroying SIP dialog '6e21ed2131bbaf2271af7ba7abe65cf8@192.168.20.1' Method: ACK
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