search for: ast_readaudio_callback

Displaying 20 results from an estimated 43 matches for "ast_readaudio_callback".

2014 Dec 12
1
Corrupt MixMonitor recordings - .gsm format
...t files. I noted warnings and errors in the CLI, apparently coinceding with corrupt MixMonitor recordings: format_gsm.c:65 gsm_read: Short read (13) (Resource temporarily unavailable)! WARNING[30727]: app_dial.c:1379 wait_for_answer: Unable to write frametype: 2 WARNING[14712]: file.c:766 ast_readaudio_callback: Failed to write frame WARNING[2612]: file.c:766 ast_readaudio_callback: Failed to write frame WARNING[25283]: format_gsm.c:65 gsm_read: Short read (32) (Resource temporarily unavailable)! WARNING[28804]: file.c:766 ast_readaudio_callback: Failed to write frame WARNING[28804]: file.c:766 ast_re...
2003 Oct 31
1
Problems with SIP
...ryone is busy at this time -- Executing VoiceMail("SIP/-0810f210", "u1234") in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- Playing 'vm/1234/unavail' (language 'en') WARNING[21518]: File file.c, Line 512 (ast_readaudio_callback): Failed to write frame WARNING[21518]: File file.c, Line 512 (ast_readaudio_callback): Failed to write frame -- Playing 'vm-intro' (language 'en') WARNING[21518]: File file.c, Line 512 (ast_readaudio_callback): Failed to write frame -- Playing 'beep' (language '...
2004 May 26
2
SPAM MESSAGE - [Asterisk-Dev] warning message (sound card) - when I run asterisk!!!
...rats") in new stack May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:408 soundcard_setinput: Unable to re-open DSP device: Device or resource busy May 26 00:40:55 WARNING[-1221268560]: chan_oss.c:567 oss_write: Unable to set device to input mode May 26 00:40:55 WARNING[-1221268560]: file.c:537 ast_readaudio_callback: Failed to write frame -- Playing 'demo-congrats' (language 'en') == Spawn extension (local, s, 5) exited non-zero on 'OSS/dsp' Is there anyone can give me any hints or help? Thanks, Neo -------------- next part -------------- An HTML attachment was scrubbed......
2009 Mar 29
2
h exten no getting run ...
...0620b28-0") in new stack [Mar 29 10:33:49] -- <Zap/1-1> Playing 'custom/Set1/a4e471fa-bd5d-859d-de11-a80900620b28-0' (language 'en') [Mar 29 10:33:53] -- Channel 0/1, span 1 got hangup request, cause 16 [Mar 29 10:33:53] WARNING[18721]: file.c:738 ast_readaudio_callback: Failed to write frame [Mar 29 10:33:53] == Spawn extension (questionnaire-menu, s, 3) exited non-zero on 'Zap/1-1' [Mar 29 10:33:53] -- Hungup 'Zap/1-1' [Mar 29 10:33:53] == End MixMonitor Recording Zap/1-1 ====================================================...
2010 Apr 29
2
Code in extensions.conf to leave a voice mail in another PBX ?!
...tdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [s-NOANSWER at macro-stdexten:1] VoiceMail("IAX2/pbx2-15464", "u8029") in new stack *[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback: Failed to write frame* -- <IAX2/pbx2-15464> Playing '/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'IAX2/pbx2-15464' in macro 'std...
2004 Sep 20
2
Voicemail Directory
...;) Sep 20 14:01:30 WARNING[118801]: app_directory.c:184 play_mailbox_owner: Can't find extension '6102' in context 'CompanyName'. Did you pass the wrong context to Directory? -- Playing 'dir-nomore' (language 'en') Sep 20 14:01:33 WARNING[118801]: file.c:548 ast_readaudio_callback: Failed to write frame == Spawn extension (CompanyName, 2, 1) exited non-zero on 'SIP/66.207.32.30-40700600' This indicates that I am passing the wrong context to Directory, however I am able to do mailbox lookups. It is just when i try to say 'yes this is the mailbox I want'...
2004 May 21
0
unable to use EXEC in AGI
...: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10 WARNING[1209214400]: file.c:537 ast_readaudio_callback: Failed t o write frame -- Playing 'vm-login' (language 'en') May 21 04:25:10 WARNING[1209214400]: app_voicemail.c:2748 vm_execmain: Couldn't read username == Spawn extension (demo, 4000, 2) exited non-zero on 'Phone/phone0' -- Hungup 'Phone/pho...
2004 Jun 02
1
oh323: Failed to create smoother
...323 voice communication should be started, following error occurs: -- Executing Playback("OH323/R1", "invalid") in new stack Jun 3 01:26:20 ERROR[294931]: chan_oh323.c:1933 oh323_write: OH323/R1: Failed to create smoother. Jun 3 01:26:20 WARNING[294931]: file.c:539 ast_readaudio_callback: Failed to write frame Asterisk is CVS Head, oh322+libs: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time librari...
2004 Oct 05
0
Just getting started with Asterisk
...ot;SIP/-08363000", "10") in new stack -- Set Response Timeout to 10 -- Executing BackGround("SIP/-08363000", "demo-congrats") in new stack -- Playing 'demo-congrats' (language 'en') Oct 5 09:58:17 WARNING[137733120]: file.c:537 ast_readaudio_callback: Failed to write frame -- Executing BackGround("SIP/-08363000", "demo-instruct") in new stack Oct 5 09:58:17 WARNING[137733120]: file.c:537 ast_readaudio_callback: Failed to write frame -- Playing 'demo-instruct' (language 'en') I have no clue as t...
2005 Jul 03
0
no sound. "Failed to write frame"
...connect to asterisk (which happens fine), I tried the echo service by dialing 600. Then I get: -- Executing Playback("SIP/julio-5ae7", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') Jul 4 14:35:42 WARNING[13521]: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 600, 1) exited non-zero on 'SIP/julio-5ae7' Then I dialed 1000, for the "congrats" message: That's what I get: -- Executing Goto("SIP/julio-f0ed", "default|s|1") in new stack -- Goto (d...
2005 Jul 04
0
no sound. "Failed to write frame" (2nd post)
...o connect to asterisk (which happens fine), I tried the echo service by dialing 600. Then I get: -- Executing Playback("SIP/julio-5ae7", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') Jul 4 14:35:42 WARNING[13521]: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 600, 1) exited non-zero on 'SIP/julio-5ae7' On my SJPhone I can see the sound is being sent, but nothing is coming back to me. Then I dialed 1000, for the "congrats" message: That's what I get: -- Executing Goto(&...
2005 Sep 21
1
Addendum to Problem with Queues question
...for '#' to acknowledge -- Started music on hold, class 'default', on Local/3044@local-4fee,2 -- Unable to find extension '' in context 'crystal-sip' -- Playing 'pbx-invalid' (language 'en') Sep 21 10:30:30 WARNING[52987]: file.c:550 ast_readaudio_callback: Failed to write frame -- Stopped music on hold on Local/3044@local-4fee,2 Sep 21 10:30:30 WARNING[52987]: res_features.c:450 ast_bridge_call: Bridge failed on channels Local/3044@local-4fee,2 and SIP/3044-ea92 == Spawn extension (macro-sipline, s, 1) exited non-zero Why doesn...
2007 Jul 05
2
sometimes calls drop during attended transfer
Hi, I'm testing attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every time I try to transfer from called phone to a third phone I get this message: -- SIP/5-082a9f78...
2007 Sep 16
0
Problem with asterisk 1.4.11 and playing files to meetme conference
...onf_run: Audio bytes: 320 Buffer size: 276 [Sep 16 06:20:44] NOTICE[18424]: app_meetme.c:1911 conf_run: Audio bytes: 18 Buffer size: 320 [Sep 16 06:20:44] WARNING[18424]: app_meetme.c:1599 conf_run: Unable to set buffering information: Invalid argument [Sep 16 06:20:44] WARNING[18423]: file.c:626 ast_readaudio_callback: Failed to write frame [Sep 16 06:20:44] WARNING[18423]: file.c:626 ast_readaudio_callback: Failed to write frame [Sep 16 06:20:44] NOTICE[18423]: pbx_spool.c:371 attempt_thread: Call completed to Local/11004 at confplay This setup worked without problems under asterisk 1.2. Any idea of what...
2009 Aug 14
1
play prompt after hanup
...up a call? I have tried below but failed. ... exten => s,n,Dial(SIP/1234) ... exten => h,1,Playback(demo-instruct) -- Executing [h at macro-safedial:2] Playback("SIP/3601-09856bf0", "demo-instruct") in new stack [Aug 14 17:24:03] WARNING[2496]: file.c:738 ast_readaudio_callback: Failed to write frame -- <SIP/3601-09856bf0> Playing 'demo-instruct' (language 'en')
2006 Jan 18
1
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds
...digits/2' (language 'es') -- Playing 'vm-isunavail' (language 'es') Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor: Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 331222, 3) exited non-zero on 'SIP/172.25.92.153-085340d0' The channels has RTP activity because I hear the voicemail message Someone has an idea to arrange this problem Thanks in advance! -------------- next part ------------...
2013 Feb 21
2
Playback on h exten
...,1") in new stack -- Goto (play,s,1) -- Executing [s at play:1] NoOp("SIP/300-00000045", "play") in new stack -- Executing [s at play:2] SayDigits("SIP/300-00000045", "123579") in new stack [Feb 21 10:35:00] WARNING[31945]: file.c:833 ast_readaudio_callback: Failed to write frame -- <SIP/300-00000045> Playing 'digits/1.ulaw' (language 'en') == Spawn extension (play, s, 2) exited non-zero on 'SIP/300-00000045' This is my dialplan: [from-test] exten => _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten =>...
2004 Dec 06
1
Console as extension problems
...6 10:08:29 WARNING[901141]: chan_oss.c:413 soundcard_setinput: -- Remote UNIX connection disconnected Unable to re-open DSP device: Device or resource busy Dec 6 10:08:29 WARNING[901141]: chan_oss.c:572 oss_write: Unable to set device to input mode Dec 6 10:08:29 WARNING[901141]: file.c:550 ast_readaudio_callback: Failed to write frame -- Playing 'new/whistle' (language 'en') << Hangup on console >> == Spawn extension (cmcm-internal, 6789, 2) exited non-zero on 'Zap/9-1' -- Hungup 'Zap/9-1' To make sure it wasn't a permissions issue, I...
2004 Sep 14
2
Spawn extension.....exited non-zero
...BackGround("IAX2/*********@*********/5", "demo-congrats") in new stack -- Playing 'demo-congrats' (language 'en') Sep 14 19:08:10 NOTICE[138140672]: chan_iax2.c:2375 iax2_read: I should never be called! Sep 14 19:08:10 WARNING[138140672]: file.c:548 ast_readaudio_callback: Failed to write frame == Spawn extension (inbound, *********, 2) exited non-zero on 'IAX2/*********@*********/5' -- Hungup 'IAX2/*********@*********/5' If this helps: I am running on FreeBSD (my previous asterisk server was also freebsd and with no probl...
2005 Sep 19
2
ztdummy configuration help
...19 13:51:22 WARNING[14066]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Sep 19 13:51:22 WARNING[14066]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') Sep 19 13:51:23 WARNING[14066]: file.c:554 ast_readaudio_callback: Failed to write frame -- Playing 'conf-getconfno' (language 'en') Any help is greatly appreciated. Kurt