Matt bucasia 2008-08-14 17:44:58 UTC #4 Just a quick update, not downgraded DISA yet but wondered what error Asterisk was actually reporting so ran it with high verbose and this is Then I get: -- Executing Playback("SIP/julio-5ae7", "demo-echotest") in new stack -- Playing 'demo-echotest' (language 'en') Jul 4 14:35:42 WARNING: file.c:550 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 600, 1) Chehab"
After upgrading today if I hang up the call while a prompt is being read to me "Welcome to the phone book ..." then I get the error - [Aug 14 com> Date: 2011-06-20 21:40:15 Message-ID: 019101cc2f92$a46dc610$ed495230$ () xplorium ! So i guess i shouldn't worry about these messages, it's normal. Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _____________________________________________________________________
Retrieved from "http://wiki.kolmisoft.com/index.php/Failed_to_write_frame" Views Page Discussion Edit History Personal tools Log in Navigation Main Page MOR MANUAL MOR Addons MOR API MOR HGC MOR X6 MOR X7 MOR X8 MOR X9 Last edited by svetur on Sat May 05, 2007 2:29 pm, edited 1 time in total. A reboot seems to have stopped it.
Thanks, Mark Code: [Apr 4 04:02:18] VERBOSE logger.c: Asterisk Event Logger restarted [Apr 4 04:02:18] VERBOSE logger.c: Asterisk Queue Logger restarted [Apr 4 04:02:18] VERBOSE logger.c: -- Playing 'dir-nomatch' (escape_digits=) (sample_offset Top xrg Post subject: Posted: Sat May 05, 2007 7:53 am Joined: Thu Oct 19, 2006 9:56 amPosts: 300Location: Athens, Greece Did you Answer() before waiting for audio? No, create an account now. Skip to content Wiki Blog Forums Mailing Lists Contact Us Advanced search Forums have moved to https://community.asterisk.org Board index RSS RSS Change font size FAQ Information The requested topic does not
A2B is trying to read DTMF but none is entered before timeout period. I was actually running Asterisk v220.127.116.11 when this started happening and, hoping an upgrade of Asterisk would fix it, I upgraded to v18.104.22.168. Learn More. Board index The team • Delete all board cookies • All times are UTC - 6 hours Powered by phpBB © 2000, 2002, 2005, 2007 phpBB Group Support A2Billing : provided
Support A2Billing : Login Register FAQ Search It is currently Thu Dec 29, 2016 5:15 am View unanswered posts | View active topics Board index » V1.2.X Chameleon Interestingly all that appears on the * console now is pbdirectory: loop = 0 -- Playing 'pbdirectory/first-three-letters-entry' (language 'en') So I guess it's something to do with loop=1 in the new Imran Ahmed 2006-04-30 13:27:38 UTC PermalinkRaw Message Post by Hatami NugrahaHi all,I always get this error message after I hangup a call, what does it mean ?WARNING: file.c:583 ast_readaudio_callback: Failed to It would also be worth downgrading to a different version of asterisk (or upgrading ...) to see if that has any effect.
Thanks, Matt p_lindheimer 2008-08-14 17:34:53 UTC #2 hmm - it may be worth manually installing the 2.4 disa over the current one and doing a reload to see if the diaplan I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. Not sure if this constitutes a bug or is just peculiar to my install. Obviously the error is being caused by Asterisk and I'm not sure why a FreePBX upgrade would cause this but they coincided hence my reason for posting here.
Board index The team • Delete all board cookies • All times are UTC - 5 hours Powered by phpBB Forum Software © phpBB Group Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users [prev in list] [next in list] [prev in thread] [next in Top svetur Post subject: Posted: Sat May 05, 2007 2:29 pm Joined: Mon May 29, 2006 7:07 pmPosts: 287Location: Denver Hi, thanks for your help. his comment is here Forum content is licensed under a Creative Commons Attribution-ShareAlike 4.0 International License. [Asterisk-Users] no sound. "Failed to write frame" Julio Cesar Ody julioody at gmail.com Sun Jul 3 21:38:29 MST 2005
Important changes Recent changes Random page Search Toolbox What links here Related changes Special pages Printable version Permanent link This page was last modified on 17 April 2009, at 15:15. Daniel_Pesserl 2009-01-28 09:26:24 UTC #7 I run trixbox and found those huge log files as well. Alerts Alert Preferences Show All...
Downgrading to version 22.214.171.124 of DISA solved my issue. FreePBX® is a registered trademark of Sangoma Technologies, Inc. http://www.fantinibakery.com/fbc-wiki/FreePbx on 2010-11-13 phone system logs were hourly rotating. Fleming"
Without any details, any >> reply you get would be just a guess, nothing more. >> >>> >>> >>> >>> >>> >>> Regards >>> >>> >>> >>> >>> >>> >>> >>> Please ensure that the file 'manager_custom.conf' exists, even if it is empty.[Sep 15 13:54:08] ERROR config.c: *********** YOU SHOULD REALLY READ THIS ERROR ***********[Sep 15 13:54:08] ERROR config.c: ********************************************************* I will I copied and pasted from our wiki, so sorry if format is not clear. weblink I installed it on a Slackware 10.1, by using no more than "make && make install && make samples".
this was in log, 10,000+ [Jan 13 14:13:15] VERBOSE logger.c: -- Playing 'dir-nomatch' (escape_digits=) (sample_offset 0) [Jan 13 14:13:16] WARNING file.c: Failed to write frame [Jan 13 14:13:16] WARNING file.c: File This week I received an alert from Spiceworks that disk space was low. (I didn't even know Spiceworks was monitoring my Linux machine, but I'm glad it is!) It turns out This has been working for several months. By using, accessing, or advertising on this site, you agree to waive all legal claims against the following entities and members: PBX in a Flash Development Team, Incredible PBX Development Team,
com [Download message RAW] The problem remains even when I add to /etc/init.d/asterisk ulimit -n 65536 [[email protected] ~]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) I'll have a go at installing v2.4 DISA and let you know what happens. Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ Anyone know what caused this and how to prevent it?
Fleming >> Digium, Inc. | Director of Software Technologies >> Jabber: [email protected] | SIP: [email protected] | Skype: >> kpfleming >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Oh, and I used a different install prefix for *all* forementioned packages. (all the same, of course) The thing is, soon as I get SJPhone to connect to asterisk (which happens Restarting Asterisk stops the errors. Newer Than: Search this thread only Search this forum only Display results as threads More...
I would investigate further as there may be other issues on your system and what ever you have changed just masked the behavior. If it works - disable unnecessary ones.