[asterisk-bugs] [Asterisk 0015784]: [regression] Simultaneous calls from same Call-ID silently ignored by asterisk
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Feb 19 08:27:42 CST 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: 2010-02-19 08:27 CST
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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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(0118274) klaus3000 (reporter) - 2010-02-19 08:27
https://issues.asterisk.org/view.php?id=15784#c118274
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If there is one INVITE from the client to SER, and SER make parallel
forking, then, why:
- are from tags different?
- Via branch ids different?
There must be somebody mangling the tags.
INVITE sip:firstnumber at asteriskserver SIP/2.0
Via: SIP/2.0/UDP
serversendinginvites;branch=z9hG4bKf80b.94fdac8c3f8f9ad3283d7dbe8ac39a90.0
Via: SIP/2.0/UDP
serversendinginvites:5061;branch=z9hG4bK7be758dae3de7a79369bbd549678c890;rport=5061
From:
<sip:originatingcaller at serversendinginvites>;tag=a08d60977ceb7bd4d98241e67dff284eo
INVITE sip:secondnumber at asteriskserver SIP/2.0
Via: SIP/2.0/UDP
serversendinginvites;branch=z9hG4bKf80b.90528f5579fe29e42359a222ba8074ea.0
Via: SIP/2.0/UDP
serversendinginvites:5061;branch=z9hG4bKb1cb9f7680bc7f3ed846dd3c2d9dbac9;rport=5061
From:
<sip:originatingcaller at serversendinginvites>;tag=67ec0ff3c23d9b35b737206e677cc802o
Anyway, if these are really the message, then there is a bug in Asterisk.
Issue History
Date Modified Username Field Change
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2010-02-19 08:27 klaus3000 Note Added: 0118274
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