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Unable To Write Frame To Channel Goautodial

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Could my system be using ztdummy instead of the card? Try adding /n to the local channel and see if the error message goes away. "Local/[email protected]_channels/n" Show Michael L. I'm trying to coorelate to see if these line up with the users who can't get in. Fax Faling On PJSip I Think This Is A Bug (video Call File) 11.25.1 And 13.13.1 Asterisk Installation Script On CentOS7 With Systemd [SOLVED] Asterisk Installation Script On CentOS7 With Systemd this contact form

regards rudra mflorell wrote:what zaptel timer are you using?are you using G729 at all? As you write I have seen this also with SIP in Meetme conferences sometimes when sip-channels is hung up. vicidial.org VICIDIAL astGUIclient discussion forum Skip to content Advanced search Vicidial.org Home Vicidial Forum Vicidial Wiki Vicidial Issue Tracker astGUIclient Project Page Board index Change font size regards rudra Jan 6 19:26:38 WARNING[30555]: app_meetme.c:1556 conf_run: Unable to write frame to channel: Success Jan 6 19:26:38 WARNING[30555]: app_meetme.c:1556 conf_run: Unable to write frame to channel: Success Jan 6 19:26:38

Unable To Write Frame To Channel Goautodial

Young, Rusty Newton Votes: 0 Vote for this issue Watchers: 5 Start watching this issue Dates Created: 04/Jun/12 2:17 AM Updated: 14/Jan/13 1:51 PM Resolved: 07/Nov/12 11:20 AM DevelopmentAgile View on Thanks. Try JIRA - bug tracking software for your team. This gives the Local channel a chance to optimize out as this is only done during processing of audio frames and Vicidial was written at a time when Asterisk had issues

Terms Privacy Opt Out Choices Advertise Get latest updates about Open Source Projects, Conferences and News. mflorell Site Admin Posts: 15344Joined: Wed Jun 07, 2006 2:45 pmLocation: Florida Website Top Reply with quote asterisk timing device by rudra_ach » Sat Jan 27, 2007 3:15 am Hi spinto Posts: 96Joined: Mon Jan 29, 2007 3:06 pm Top Reply with quote by mflorell » Tue Jan 30, 2007 12:02 pm post results of "lsmod" mflorell Site Admin It is acting like a test carrier.

Frequency of Occurrence: Occasional Description Basically it looks like in some instances MeetMe is still trying to pass frames to a channel that should have hung up. It looks like it is generating about 50 errors per second for one of these calls. How would I check that? http://forums.asterisk.org/viewtopic.php?f=1&t=81295 There are no sound problems for me, either, but when the caller hangs up and this error occurs, the trunk statuses are not updated properly and the phones still show them

Board index The team • Delete all board cookies • All times are UTC - 5 hours Powered by phpBB Forum Software © phpBB Group Linked ApplicationsLoading… DashboardsProjectsIssuesAgileSign a License Agreement People Assignee: Unassigned Reporter: Michael Cargile Issue Participants: David Brillert, Johan Wilfer, Jonathan Rose, Matt Jordan, Michael Cargile, Michael L. After I Hang Up the call I get this: [2013-03-16 20:19:48] VERBOSE24377 logger.c: [2013-03-16 20:19:48] Parsing '/etc/asterisk/manager.conf': [2013-03-16 20:19:48] WARNING[12844] app_meetme.c: Unable to write frame to channel Local/[email protected],2[2013-03-16 20:19:48] VERBOSE[24377] logger.c: The system is only being used for conferencing (app_MeetMe).

Asterisk App_meetme.c Unable To Write Frame To Channel

I'm not sure what logs I can provide to help troubleshoot and resolve. Is this a timing issue? Unable To Write Frame To Channel Goautodial According to Asterisk wiki, the preferred timing module is res_timing_timerfd.so (then res_timing_dahdi.so and the least preferred res_timing_pthread.so). From: mistral9999 at hotmail.com To: asterisk-users at lists.digium.com Date: Fri, 20 Jan 2012 14:09:21 -0500 Subject: [asterisk-users] meetme - Unable to write frame to channel Hi, Once in a while when

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 -------------- next part -------------- An HTML attachment was scrubbed... http://popupjammer.com/unable-to/unable-to-create-channel-of-type-39-sip-39-cause-20-subscriber-absent-freepbx.html Should I still apply the timer patch? Any thoughts? mflorell Site Admin Posts: 15344Joined: Wed Jun 07, 2006 2:45 pmLocation: Florida Website Top Display posts from previous: All posts1 day7 days2 weeks1 month3 months6 months1 year Sort by AuthorPost

Closed Activity Ascending order - Click to sort in descending order All Comments Work Log History Activity CI Builds Transitions Gerrit Reviews Source Reviews Builds Hide Permalink Rusty Newton added a rudra_ach Posts: 108Joined: Fri Jun 23, 2006 11:22 am Top Reply with quote by mflorell » Mon Jan 08, 2007 7:48 am That's most likely your problem, try getting an Briefly describe the problem (required): Upload screenshot of ad (required): Select a file, or drag & drop file here. ✔ ✘ Please provide the ad click URL, if possible: Home Browse navigate here mflorell Site Admin Posts: 15344Joined: Wed Jun 07, 2006 2:45 pmLocation: Florida Website Top Reply with quote thanks!

I have seen some posted bug fixes for somewhat related things in Asterisk 1.2.15, so I plan on trying that to see if there is a similar problem with the new Attend the live webc= ast > and join the prime developer group breaking into this new coding territor= y! > http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D110944&bid=3D241720&dat= =3D121642 > _______________________________________________ > Astguiclient-users mailing list > [email protected] > It's when people call in from external that it kind of goes crazy.

Also, this patch has a lot of similarity with https://issues.asterisk.org/jira/secure/attachment/43067/meetme-hangup-trunk-360459.patch which I've been using for about five months now and it has worked perfectly.

Atlassian Sign in Register Home Projects Help Search: GOautodial CE ISO Installer (Open Source Call Center Suite - web based predictive dialer + inbound IVR & ACD) Overview Activity Roadmap Issues What version of Asterisk are you using? Show Johan Wilfer added a comment - 09/Oct/12 3:39 PM Nope. Show Johan Wilfer added a comment - 11/Nov/12 11:47 AM Okay, I have now tested the new patch.

Same thing... As stated this error pops up occasionally. Please don't fill out this field. http://popupjammer.com/unable-to/unable-to-load-frame-39-s-content-invalid-or-nonexistent-document.html Hide Permalink Michael Cargile added a comment - 08/Nov/12 4:14 PM Sorry got side tracked by work.

Everything works great if we call it from internal extensions. Hide Permalink Jonathan Rose added a comment - 08/Nov/12 11:35 AM - edited Any recent release of 1.8 (last couple months or so) should be fine. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA. The Local channel is pointed at the 91999NXXXXXX extension depending on the "phone number" being dialed.

Shares Of This Company Will Triple By Christmas. are you doing any recording= ? I am not sure if this is a SIP issue, or a MeetMe issue. The patch in issue ASTERISK-19594 https://issues.asterisk.org/jira/secure/attachment/43067/meetme-hangup-trunk-360459.patch seems to work, but apparently only masks the problem.

Yes they both 8001/8002 can hear eachother well. [2013-03-17 20:32:03] VERBOSE10281 logger.c: [2013-03-17 20:32:03] Manager 'sendcron' logged off from 127.0.0.1[2013-03-17 20:32:05] VERBOSE[17671] logger.c: [2013-03-17 20:32:05] -- Registered SIP '8002' at 80.80.47.239 There is 1 extension configured that automatically forwards all calls to the custom destination that is the MeetMe setup. I run the setup described in the description in ASTERISK-19949.