a2billing 的星号 dtmf 问题？
asterisk dtmf issue from a2billing?
当我从 phone 打电话时，我遇到了星号和 a2billing 的问题，我在输入 2 位数字后听到忙音，关于如何解决问题的任何想法或某种设置可以应用于修复它
这是提升调试级别后来自 asterisk 的日志[添加了与所需调用场景相关的 asterisk/full 日志部分]
[2017-11-09 23:49:06] DEBUG[122210] chan_sip.c: Allocating new SIP dialog for 24033380-3719252948-513622@SBC01 - INVITE (No RTP)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel"
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Found SIP option: -timer-
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Matched SIP option: timer
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Found SIP option: -100rel-
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Matched SIP option: 100rel
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting 'XX.XXX.XX.XXX:5060' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host 'XX.XXX.XX.XXX' and port '5060'.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting 'NULL.invalid' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host 'NULL.invalid' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Allocated port 19348 for RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Creating ICE session 10.64.4.73:19348 (19348) for RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.8.71' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.8.71' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.8.73' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.8.73' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.4.71' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.4.71' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.4.73' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.4.73' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: RTP instance '0x7f972c0069c8' is setup and ready to go
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Setting NAT on RTP to Off
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP o=SBC01 1 0 IN IP4 XX.XXX.XX.XXX... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP s=sip call... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '82.213.25.241' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '82.213.25.241' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP c=IN IP4 82.213.25.241... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Setting payload 8 based on m type on 0x7f97980b5b70
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Setting payload 18 based on m type on 0x7f97980b5b70
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Setting payload 101 based on m type on 0x7f97980b5b70
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=ptime:10... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000/1... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15,16-255... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=X-AD-REALM:0... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Copying payload 8 from 0x7f97980b5b70 to 0x7f972c006b90
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Copying payload 18 from 0x7f97980b5b70 to 0x7f972c006b90
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Copying payload 101 from 0x7f97980b5b70 to 0x7f972c006b90
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: We're settling with these formats: (g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Checking SIP call limits for device
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Updating call counter for incoming call
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.4.73:5060' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.4.73' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting 'NULL.invalid' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host 'NULL.invalid' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Incoming INVITE with 'timer' option supported
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: INVITE also has "Session-Expires" header.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Session-Expires: 3600
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Refresher: UAC
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: INVITE also has "Min-SE" header.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Received Min-SE: 600
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** Our native formats are (g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** Joint capabilities are (g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** Our capabilities are (ulaw|g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** AST_CODEC_CHOOSE formats are g729
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: This channel will not be able to handle video.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: build_route: Contact hop: <sip:NULL@XX.XXX.XX.XXX:5060;tgrp=LABPROFILE>
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: SIP/1700_CS2K_In-00000010: New call is still down.... Trying...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for XX.XXX.XX.XXX:5060
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: No provider found, checking channel drivers for SIP - 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] chan_sip.c: Checking device state for peer 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: Changing state for SIP/1700_CS2K_In - state 1 (Not in use)
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: device 'SIP/1700_CS2K_In' state '1'
[2017-11-09 23:49:06] DEBUG[122246] app_queue.c: Device 'SIP/1700_CS2K_In' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'EXTEN' is '022983342'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Gosub'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] app_stack.c: Channel SIP/1700_CS2K_In-00000010 has no datastore, so we're allocating one.
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] db.c: Unable to find key 'NULL' in family 'blacklist'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] db.c: Unable to find key '' in family 'blacklist'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function BLACKLIST() result is '0'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'GotoIf'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Not taking any branch
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Return'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'FROM_DID' is '022983342'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function CALLERID(name) result is ''
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Expression result is '1'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function CALLERID(num) result is 'NULL'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'ExecIf'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function CALLERPRES() result is 'allowed_not_screened'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'EXTEN' is '022983342'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Goto'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'DeadAGI'
[2017-11-09 23:49:06] WARNING[11040][C-00000010] res_agi.c: DeadAGI has been deprecated, please use AGI in all cases!
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'IDCONF' is NULL
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'A2B_CUSTOM1' is NULL
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'A2B_CUSTOM2' is NULL
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: SIP answering channel: SIP/1700_CS2K_In-00000010
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Setting the marker bit due to a source update
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: No provider found, checking channel drivers for SIP - 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] chan_sip.c: Checking device state for peer 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: Changing state for SIP/1700_CS2K_In - state 1 (Not in use)
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: device 'SIP/1700_CS2K_In' state '1'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Setting framing from config on incoming call
[2017-11-09 23:49:06] DEBUG[122246] app_queue.c: Device 'SIP/1700_CS2K_In' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: ** Our capability: (g729) Video flag: True Text flag: True
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: ** Our prefcodec: (nothing)
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: -- Done with adding codecs to SDP
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Done building SDP. Settling with this capability: (g729)
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for XX.XXX.XX.XXX:5060
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Session timer started: 1 - 24033380-3719252948-513622@SBC01 1768000ms
[2017-11-09 23:49:06] DEBUG[122210] chan_sip.c: = Looking for Call ID: 24033380-3719252948-513622@SBC01 (Checking From) --From tag 3719252948-513625 --To-tag as39e484de
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Stopping retransmission on '24033380-3719252948-513622@SBC01' of Response 1: Match Found
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: 0x7f972c04c200 -- Probation learning mode pass with source address 82.213.25.241:44944
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Ooh, format changed from unknown to g729
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Created smoother: format: g729 ms: 20 len: 20
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 49 (1), at 82.213.25.241:44944
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 49 (1), at 82.213.25.241:44944
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:09] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:09] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:09] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:09] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:09] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:09] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 35 (#), at 82.213.25.241:44944
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 35 (#), at 82.213.25.241:44944
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Difference is 11832, ms is 1499
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:12] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (gsm)
[2017-11-09 23:49:12] WARNING[11040][C-00000010] file.c: Unable to open digits/hundred (format (g729)): No such file or directory
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 57 (9), at 82.213.25.241:44944
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 57 (9), at 82.213.25.241:44944
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:17] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:17] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:17] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:18] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:18] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:19] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:19] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:19] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:19] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:20] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:20] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 56 (8), at 82.213.25.241:44944
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 56 (8), at 82.213.25.241:44944
[2017-11-09 23:49:20] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:20] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:21] DEBUG[122210] chan_sip.c: = Looking for Call ID: 24033380-3719252948-513622@SBC01 (Checking From) --From tag 3719252948-513625 --To-tag as39e484de
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: **** Received BYE (8) - Command in SIP BYE
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] netsock2.c: Splitting 'XX.XXX.XX.XXX:5060' into...
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] netsock2.c: ...host 'XX.XXX.XX.XXX' and port '5060'.
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Setting SIP_ALREADYGONE on dialog 24033380-3719252948-513622@SBC01
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Session timer stopped: 1 - 24033380-3719252948-513622@SBC01
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Received bye, issuing owner hangup
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for XX.XXX.XX.XXX:5060
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] res_agi.c: SIP/1700_CS2K_In-00000010 hungup
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] pbx.c: Spawn extension (a2billing-ahlan,022983342,1) exited non-zero on 'SIP/1700_CS2K_In-00000010'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] channel.c: Soft-Hanging up channel 'SIP/1700_CS2K_In-00000010'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] channel.c: Hanging up channel 'SIP/1700_CS2K_In-00000010'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] chan_sip.c: Hangup call SIP/1700_CS2K_In-00000010, SIP callid 24033380-3719252948-513622@SBC01
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] cdr_mysql.c: Inserting a CDR record.
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`,`did`) VALUES ('2017-11-09 23:49:06','\"NULL\" <NULL>','NULL','022983342','a2billing-ahlan','SIP/1700_CS2K_In-00000010','AGI','GET','15','15','ANSWERED','3','1510264146.16','022983342')
此致，
亲爱的，
感谢您的努力，但问题不是来自 a2billing 或 asterisk 方面，而是音频开关的问题，rtp 超时太小
重新注册，
当我从 phone 打电话时，我遇到了星号和 a2billing 的问题，我在输入 2 位数字后听到忙音，关于如何解决问题的任何想法或某种设置可以应用于修复它
这是提升调试级别后来自 asterisk 的日志[添加了与所需调用场景相关的 asterisk/full 日志部分]
[2017-11-09 23:49:06] DEBUG[122210] chan_sip.c: Allocating new SIP dialog for 24033380-3719252948-513622@SBC01 - INVITE (No RTP)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Begin: parsing SIP "Supported: timer, 100rel"
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Found SIP option: -timer-
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Matched SIP option: timer
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Found SIP option: -100rel-
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] sip/reqresp_parser.c: Matched SIP option: 100rel
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting 'XX.XXX.XX.XXX:5060' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host 'XX.XXX.XX.XXX' and port '5060'.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting 'NULL.invalid' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host 'NULL.invalid' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Allocated port 19348 for RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Creating ICE session 10.64.4.73:19348 (19348) for RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.8.71' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.8.71' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.8.73' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.8.73' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.4.71' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.4.71' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.4.73' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.4.73' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: RTP instance '0x7f972c0069c8' is setup and ready to go
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Setting NAT on RTP to Off
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP o=SBC01 1 0 IN IP4 XX.XXX.XX.XXX... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP s=sip call... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '82.213.25.241' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '82.213.25.241' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP c=IN IP4 82.213.25.241... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Setting payload 8 based on m type on 0x7f97980b5b70
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Setting payload 18 based on m type on 0x7f97980b5b70
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Setting payload 101 based on m type on 0x7f97980b5b70
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=ptime:10... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000/1... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15,16-255... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Processing media-level (audio) SDP a=X-AD-REALM:0... UNSUPPORTED OR FAILED.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Copying payload 8 from 0x7f97980b5b70 to 0x7f972c006b90
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Copying payload 18 from 0x7f97980b5b70 to 0x7f972c006b90
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] rtp_engine.c: Copying payload 101 from 0x7f97980b5b70 to 0x7f972c006b90
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: We're settling with these formats: (g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Checking SIP call limits for device
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Updating call counter for incoming call
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting '10.64.4.73:5060' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host '10.64.4.73' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: Splitting 'NULL.invalid' into...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] netsock2.c: ...host 'NULL.invalid' and port ''.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Incoming INVITE with 'timer' option supported
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: INVITE also has "Session-Expires" header.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Session-Expires: 3600
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Refresher: UAC
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: INVITE also has "Min-SE" header.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Received Min-SE: 600
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** Our native formats are (g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** Joint capabilities are (g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** Our capabilities are (ulaw|g729)
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: *** AST_CODEC_CHOOSE formats are g729
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: This channel will not be able to handle video.
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: build_route: Contact hop: <sip:NULL@XX.XXX.XX.XXX:5060;tgrp=LABPROFILE>
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: SIP/1700_CS2K_In-00000010: New call is still down.... Trying...
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for XX.XXX.XX.XXX:5060
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: No provider found, checking channel drivers for SIP - 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] chan_sip.c: Checking device state for peer 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: Changing state for SIP/1700_CS2K_In - state 1 (Not in use)
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: device 'SIP/1700_CS2K_In' state '1'
[2017-11-09 23:49:06] DEBUG[122246] app_queue.c: Device 'SIP/1700_CS2K_In' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'EXTEN' is '022983342'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Gosub'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] app_stack.c: Channel SIP/1700_CS2K_In-00000010 has no datastore, so we're allocating one.
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] db.c: Unable to find key 'NULL' in family 'blacklist'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] db.c: Unable to find key '' in family 'blacklist'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function BLACKLIST() result is '0'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'GotoIf'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Not taking any branch
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Return'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'FROM_DID' is '022983342'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function CALLERID(name) result is ''
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Expression result is '1'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function CALLERID(num) result is 'NULL'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'ExecIf'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Function CALLERPRES() result is 'allowed_not_screened'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Set'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'EXTEN' is '022983342'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'Goto'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Launching 'DeadAGI'
[2017-11-09 23:49:06] WARNING[11040][C-00000010] res_agi.c: DeadAGI has been deprecated, please use AGI in all cases!
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'IDCONF' is NULL
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'A2B_CUSTOM1' is NULL
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] pbx.c: Result of 'A2B_CUSTOM2' is NULL
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: SIP answering channel: SIP/1700_CS2K_In-00000010
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Setting the marker bit due to a source update
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: No provider found, checking channel drivers for SIP - 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] chan_sip.c: Checking device state for peer 1700_CS2K_In
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: Changing state for SIP/1700_CS2K_In - state 1 (Not in use)
[2017-11-09 23:49:06] DEBUG[122200] devicestate.c: device 'SIP/1700_CS2K_In' state '1'
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Setting framing from config on incoming call
[2017-11-09 23:49:06] DEBUG[122246] app_queue.c: Device 'SIP/1700_CS2K_In' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: ** Our capability: (g729) Video flag: True Text flag: True
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: ** Our prefcodec: (nothing)
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: -- Done with adding codecs to SDP
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Done building SDP. Settling with this capability: (g729)
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for XX.XXX.XX.XXX:5060
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] chan_sip.c: Session timer started: 1 - 24033380-3719252948-513622@SBC01 1768000ms
[2017-11-09 23:49:06] DEBUG[122210] chan_sip.c: = Looking for Call ID: 24033380-3719252948-513622@SBC01 (Checking From) --From tag 3719252948-513625 --To-tag as39e484de
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[2017-11-09 23:49:06] DEBUG[122210][C-00000010] chan_sip.c: Stopping retransmission on '24033380-3719252948-513622@SBC01' of Response 1: Match Found
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: 0x7f972c04c200 -- Probation learning mode pass with source address 82.213.25.241:44944
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Ooh, format changed from unknown to g729
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Created smoother: format: g729 ms: 20 len: 20
[2017-11-09 23:49:06] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 49 (1), at 82.213.25.241:44944
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 49 (1), at 82.213.25.241:44944
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:09] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:09] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:09] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:09] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:09] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:09] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:09] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 35 (#), at 82.213.25.241:44944
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 35 (#), at 82.213.25.241:44944
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Difference is 11832, ms is 1499
[2017-11-09 23:49:10] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:11] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:12] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (gsm)
[2017-11-09 23:49:12] WARNING[11040][C-00000010] file.c: Unable to open digits/hundred (format (g729)): No such file or directory
[2017-11-09 23:49:12] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:13] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 57 (9), at 82.213.25.241:44944
[2017-11-09 23:49:15] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 57 (9), at 82.213.25.241:44944
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:16] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] DEBUG[11040][C-00000010] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[2017-11-09 23:49:17] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:17] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:17] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:17] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:18] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:18] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:18] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:19] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:19] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 48 (0), at 82.213.25.241:44944
[2017-11-09 23:49:19] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:19] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 50 (2), at 82.213.25.241:44944
[2017-11-09 23:49:20] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:20] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating BEGIN DTMF Frame: 56 (8), at 82.213.25.241:44944
[2017-11-09 23:49:20] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Creating END DTMF Frame: 56 (8), at 82.213.25.241:44944
[2017-11-09 23:49:20] WARNING[11040][C-00000010] channel.c: Unable to find a codec translation path from (g729) to (slin)
[2017-11-09 23:49:20] ERROR[11040][C-00000010] channel.c: Could not set write format to SLINEAR
[2017-11-09 23:49:21] DEBUG[122210] chan_sip.c: = Looking for Call ID: 24033380-3719252948-513622@SBC01 (Checking From) --From tag 3719252948-513625 --To-tag as39e484de
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: **** Received BYE (8) - Command in SIP BYE
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] netsock2.c: Splitting 'XX.XXX.XX.XXX:5060' into...
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] netsock2.c: ...host 'XX.XXX.XX.XXX' and port '5060'.
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Setting SIP_ALREADYGONE on dialog 24033380-3719252948-513622@SBC01
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Session timer stopped: 1 - 24033380-3719252948-513622@SBC01
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Received bye, issuing owner hangup
[2017-11-09 23:49:21] DEBUG[122210][C-00000010] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for XX.XXX.XX.XXX:5060
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] res_agi.c: SIP/1700_CS2K_In-00000010 hungup
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] pbx.c: Spawn extension (a2billing-ahlan,022983342,1) exited non-zero on 'SIP/1700_CS2K_In-00000010'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] channel.c: Soft-Hanging up channel 'SIP/1700_CS2K_In-00000010'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] channel.c: Hanging up channel 'SIP/1700_CS2K_In-00000010'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] chan_sip.c: Hangup call SIP/1700_CS2K_In-00000010, SIP callid 24033380-3719252948-513622@SBC01
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f972c0069c8'
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] cdr_mysql.c: Inserting a CDR record.
[2017-11-09 23:49:21] DEBUG[11040][C-00000010] cdr_mysql.c: SQL command as follows: INSERT INTO cdr (`calldate`,`clid`,`src`,`dst`,`dcontext`,`channel`,`lastapp`,`lastdata`,`duration`,`billsec`,`disposition`,`amaflags`,`uniqueid`,`did`) VALUES ('2017-11-09 23:49:06','\"NULL\" <NULL>','NULL','022983342','a2billing-ahlan','SIP/1700_CS2K_In-00000010','AGI','GET','15','15','ANSWERED','3','1510264146.16','022983342')
此致，
亲爱的，
感谢您的努力，但问题不是来自 a2billing 或 asterisk 方面，而是音频开关的问题，rtp 超时太小
重新注册，