Paul Rodan
2004-Nov-01 13:29 UTC
[Asterisk-Users] Unable to write frame to channel: Success - MeetMe problem
After I upgraded from the CVS Head to the latest CVS 1.0 Stable, my paging application doesn't work. Using the Wiki as a guidance, I made a line 2 on all phones with auto answer. When someone wants to page out, they dial an extension and it brings everyone into the conference, with everyone muted. This system used to work flawless, but now when I use the extension, it brings everybody into the conference without a problem, but it's silent, until the timeout kicks in. Everybody can't hear the person speaking, the who initiated the conference, it's broken. My log files show: Nov 1 15:15:22 NOTICE[8716311]: Call failed to go through, reason 3 Nov 1 15:15:22 NOTICE[8831023]: Call failed to go through, reason 3 Nov 1 15:15:23 NOTICE[8486946]: Call completed to SIP/rkrisel_page Nov 1 15:15:24 WARNING[8339481]: Unable to write frame to channel: Success Nov 1 15:15:24 WARNING[8519716]: Unable to write frame to channel: Success Nov 1 15:15:24 WARNING[8323096]: Unable to write frame to channel: Success Any ideas? My timing is provided by zaptelrtc (the zaprtc module and the rtcsetup binary). This is how it's been for some time. I've recompiled a and reinstalled Asterisk/Zaptel/LibPRI/ZaptelRTC to no avail.>From my extensions.conf file:--- [paging] exten => *,1,AbsoluteTimeout(15) exten => *,2,agi(pageall) exten => *,3,MeetMe(1111,xdqp) exten => *,4,Hangup [add-to-paging] exten => start,1,AbsoluteTimeout(15) exten => start,2,MeetMe(1111,dmqp) exten => start,3,Hangup exten => h,1,Hangup exten => t,1,Hangup exten => T,1,Hangup --- One of my .call files: --- Channel: SIP/rkrisel_page Context: add-to-wupaging Extension: start Priority: 1 CallerID: Office Pager <1111> WaitTime: 3 --- And what I have in meetme.conf --- conf => 1111 --- And what I have in my agi script pageall: --- #!/bin/sh /bin/cp /var/lib/asterisk/paging/*.call /var/spool/asterisk/outgoing --- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041101/a0a900fc/attachment.htm
Paul Rodan
2004-Nov-01 14:56 UTC
[Asterisk-Users] Unable to write frame to channel: Success - MeetMeproblem
Nov 1 16:54:05 NOTICE[12238881]: Call failed to go through, reason 0 Nov 1 16:54:05 NOTICE[12009491]: Unable to request channel SIP/lbarr_page Nov 1 16:54:05 NOTICE[12009491]: Call failed to go through, reason 0 Nov 1 16:54:05 NOTICE[12304421]: Unable to request channel SIP/noclobby1_page Nov 1 16:54:05 NOTICE[12304421]: Call failed to go through, reason 0 Nov 1 16:54:08 NOTICE[12042261]: Call failed to go through, reason 3 Nov 1 16:54:09 NOTICE[12353576]: Call failed to go through, reason 3 Nov 1 16:54:11 NOTICE[12222496]: Call completed to SIP/drodden_page Starting to get a little frustrated here; several users have called and screamed at me. This paging system used to work. When I page, everybody's phone answers but everybody gets dead silence, the conference room is completely broken. It's disappointing really. I'd like to think it's the zaprtc timing, but it's worked flawlessly in the past, nothing has changed except zaptel/libpri/asterisk I didn't change the kernel, or the way the module was loaded or the rtcsetup program running in the background. I may be forced to downgrade but I don't know to what version. Sigh. ________________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Rodan Sent: Monday, November 01, 2004 3:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Unable to write frame to channel: Success - MeetMeproblem After I upgraded from the CVS Head to the latest CVS 1.0 Stable, my paging application doesn?t work. Using the Wiki as a guidance, I made a line 2 on all phones with auto answer. When someone wants to page out, they dial an extension and it brings everyone into the conference, with everyone muted. This system used to work flawless, but now when I use the extension, it brings everybody into the conference without a problem, but it's silent, until the timeout kicks in. Everybody can't hear the person speaking, the who initiated the conference, it?s broken. My log files show: Nov? 1 15:15:22 NOTICE[8716311]: Call failed to go through, reason 3 Nov? 1 15:15:22 NOTICE[8831023]: Call failed to go through, reason 3 Nov? 1 15:15:23 NOTICE[8486946]: Call completed to SIP/rkrisel_page Nov? 1 15:15:24 WARNING[8339481]: Unable to write frame to channel: Success Nov? 1 15:15:24 WARNING[8519716]: Unable to write frame to channel: Success Nov? 1 15:15:24 WARNING[8323096]: Unable to write frame to channel: Success Any ideas? My timing is provided by zaptelrtc (the zaprtc module and the rtcsetup binary). This is how it?s been for some time. I?ve recompiled a and reinstalled Asterisk/Zaptel/LibPRI/ZaptelRTC to no avail.>From my extensions.conf file:--- [paging] exten => *,1,AbsoluteTimeout(15) exten => *,2,agi(pageall) exten => *,3,MeetMe(1111,xdqp) exten => *,4,Hangup [add-to-paging] exten => start,1,AbsoluteTimeout(15) exten => start,2,MeetMe(1111,dmqp) exten => start,3,Hangup exten => h,1,Hangup exten => t,1,Hangup exten => T,1,Hangup --- One of my .call files: --- Channel: SIP/rkrisel_page Context: add-to-wupaging Extension: start Priority: 1 CallerID: Office Pager <1111> WaitTime: 3 --- And what I have in meetme.conf --- conf => 1111 --- And what I have in my agi script pageall: --- #!/bin/sh /bin/cp /var/lib/asterisk/paging/*.call /var/spool/asterisk/outgoing ---