[asterisk-bugs] [Asterisk 0015266]: Calling over TAPI hangs up the line if trunk responses with "SIP/2.0 403 Forbidden, no minutes left"

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jun 4 07:20:12 CDT 2009


The following issue has been RESOLVED. 
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https://issues.asterisk.org/view.php?id=15266 
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Reported By:                smps
Assigned To:                russell
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Project:                    Asterisk
Issue ID:                   15266
Category:                   Core/ManagerInterface
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     resolved
Asterisk Version:           1.4.25 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 suspended
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-06-03 20:14 CDT
Last Modified:              2009-06-04 07:20 CDT
====================================================================== 
Summary:                    Calling over TAPI hangs up the line if trunk
responses with "SIP/2.0 403 Forbidden, no minutes left"
Description: 
I use activa tsp to interface with ami from windows. If I call from
microsoft dialer.exe over ami and first trunk responses with "SIP/2.0 403
Forbidden, no minutes left" then ami hangs up the line und would not
failover thru other trunks. But if i call from sip telephone it will
failover thru another trunks. You can check execution output from asterisk
on pastebin (its a bit big and i use freepbx webinterface):
calling over tapi: http://pastebin.com/f747d9b4c
calling with sip phone: http://pastebin.com/f7b7a913d

In this example i use 2 SIP trunks from 2 different providers for calling
out. 


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---------------------------------------------------------------------- 
 (0105971) russell (administrator) - 2009-06-04 07:20
 https://issues.asterisk.org/view.php?id=15266#c105971 
---------------------------------------------------------------------- 
I would suggest pursuing some support through your windows dialer vendor or
the FreePBX community first for some assistance with debugging to see if
you have a problem actually in Asterisk or not. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-06-04 07:20 russell        Note Added: 0105971                          
2009-06-04 07:20 russell        Status                   new => resolved     
2009-06-04 07:20 russell        Resolution               open => suspended   
2009-06-04 07:20 russell        Assigned To               => russell         
======================================================================




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