[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 noreply at bugs.digium.com
Tue Apr 7 13:58:04 CDT 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14843 
====================================================================== 
Reported By:                aragon
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-04-07 08:12 CDT
Last Modified:              2009-04-07 13:58 CDT
====================================================================== 
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
====================================================================== 

---------------------------------------------------------------------- 
 (0102853) jvandal (reporter) - 2009-04-07 13:58
 http://bugs.digium.com/view.php?id=14843#c102853 
---------------------------------------------------------------------- 
If this can help, I'm trying to do some analysis...

The error occur when the AGI do a: 'AGI Tx >> ... ', if I look on
res/res_agi.c, it call the ast_carefulwrite function, maybe the timeout of
100ms is too short ?

if (agidebug)
        ast_verbose("AGI Tx >> %s", stuff); /* \n provided by caller */
res = ast_carefulwrite(fd, stuff, strlen(stuff), 100); 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-07 13:58 jvandal        Note Added: 0102853                          
======================================================================




More information about the asterisk-bugs mailing list