[asterisk-bugs] [Asterisk 0015784]: [regression] Simultaneous calls from same Call-ID silently ignored by asterisk

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jun 3 04:41:39 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=15784 
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Reported By:                m0bius
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15784
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                       SWP-221 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-08-27 05:40 CDT
Last Modified:              2010-06-03 04:41 CDT
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Summary:                    [regression] Simultaneous calls from same Call-ID
silently ignored by asterisk
Description: 
Hello everyone,

We have a follow-me system which terminates calls to an Asterisk server
which holds registrations for our VoIP users. In our follow-me system we
give the capability to the users to perform simultaneous follow-me to the
Asterisk Server (thus ringing two different voip accounts).

However I've noticed that on asterisk 1.6.1.1 and 1.6.1.4 when two calls
are sent simultaneously to different dialled numbers with the same Call-ID,
the second call does not enter the context. In a trace I did, I've seen
that asterisk responds to the SIP INVITE with Trying; however, that calls
stays there until it times out from the remote peer. 

The same thing has been tested on Asterisk 1.6.0.7 and 1.6.0.13 and it
works properly. I will attaching two traces (one from asterisk 1.6.0.7 and
one from asterisk 1.6.1.4)
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Relationships       ID      Summary
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related to          0016116 [patch] Fix/improve transaction/dialog-...
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 (0122868) MikaelF (reporter) - 2010-06-03 04:41
 https://issues.asterisk.org/view.php?id=15784#c122868 
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Look for these lines and change them to this:

7312: found = ast_strlen_zero(arg->tag) || ast_strlen_zero(p->theirtag) ||
!strcmp(p->theirtag, arg->tag);

7375: arg.tag = (req->method == SIP_RESPONSE) ? totag : fromtag;

They may not be at just those lines but that's all you need to change to
fix it, they should at least be close to the line numbers above. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-06-03 04:41 MikaelF        Note Added: 0122868                          
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