[Asterisk-Users] Unable to write frame to channel: Success - MeetMeproblem

[Paul Rodan]

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 at lists.digium.com
[mailto:asterisk-users-bounces at 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
---