[asterisk-bugs] [Asterisk 0014843]: 1 in 3 incoming zap PRI calls do no hear audio (are not bridged) when call is answered with agi script

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14843 
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Reported By:                aragon
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14843
Category:                   Resources/res_agi
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.24 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-04-07 08:12 CDT
Last Modified:              2009-04-07 09:58 CDT
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Summary:                    1 in 3 incoming zap PRI calls do no hear audio (are
not bridged) when call is answered with agi script
Description: 
Incoming zap PRI call answers with DNIS and agi script
Some calls are answered properly with audio (bridged)
Sometimes caller hears no audio (not bridged)
Roughly 1 in 3 callers hears no audio.
No relation found to specific B channel

I'm opening as major because lots of calls are not being bridged.

Started with Asterisk 1.4.20 problem still in 1.4.24.1

zaptel-1.4.12
asterisk-1.4.24.1

The last line in the CLI keeps repeating until the caller hangs up

[Apr 6 19:35:18] NOTICE[26970]: utils.c:938 ast_carefulwrite: Timed out
trying to write
[Apr 6 19:35:18] NOTICE[26970]: utils.c:938 ast_carefulwrite: Timed out
trying to write

or 

[Apr 6 22:37:10] DEBUG[20731]: chan_sip.c:2226 __sip_ack: Stopping
retransmission on '7cd1530c5be721a24327004c5d520b3e at 10.0.50.254' of Request
102: Match Found
[Apr 6 22:37:10] DEBUG[20731]: chan_sip.c:4653 sip_alloc: Allocating new
SIP dialog for (No Call-ID) - OPTIONS
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 (0102840) aragon (reporter) - 2009-04-07 09:58
 http://bugs.digium.com/view.php?id=14843#c102840 
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At least downgrading to 1.4.21 proves that this is an issue with Asterisk
and not agi script...

I think jvandal maybe correct in his analysis of root cause.
The workaround to increase the writetimeout I think means the caller is
never bridged...

0013546: Partial writes on Manager API
Description 	Sometimes writing manager events with ast_carefulwrite in
manager.c only results in a partial write i.e. not all data is written to
the socket. There is no check for this and no code to send the missing
data. As a workaround increasing the writetimeout in manager.conf mitigates
this issue.
See
http://jira.reucon.org/browse/AJ-174?focusedCommentId=10522#action_10522
[^] for details. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-07 09:58 aragon         Note Added: 0102840                          
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