[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 04:35:23 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 04:35 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)
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0016116 [patch] Fix/improve transaction/dialog-...
====================================================================== 

---------------------------------------------------------------------- 
 (0118269) MikaelF (reporter) - 2010-02-19 04:35
 https://issues.asterisk.org/view.php?id=15784#c118269 
---------------------------------------------------------------------- 
I am sorry if this is not the place to post this but I think I have the
same problem, asterisk version 1.6.2.0, pedantic is set to yes.

Two incomming calls to the asterisk from the same caller is sent, second
invite gets: "Ignoring this INVITE request":

INVITE sip:firstnumber at asteriskserver SIP/2.0
Record-Route:
<sip:serversendinginvites;ftag=a08d60977ceb7bd4d98241e67dff284eo;lr>
Via: SIP/2.0/UDP
serversendinginvites;branch=z9hG4bKf80b.94fdac8c3f8f9ad3283d7dbe8ac39a90.0
Via: SIP/2.0/UDP
serversendinginvites:5061;branch=z9hG4bK7be758dae3de7a79369bbd549678c890;rport=5061
Max-Forwards: 16
From:
<sip:originatingcaller at serversendinginvites>;tag=a08d60977ceb7bd4d98241e67dff284eo
To: <sip:firstnumber at serversendinginvites>
Call-ID: EA0785B5-1C7211DF-99B3A4EC-DCEB9CE0 at serverip
CSeq: 200 INVITE
Contact: Anonymous <sip:serversendinginvites:5061>
Expires: 300
User-Agent: serversendinginvites
cisco-GUID: 544724832-511663812-3825585790-2681720812
h323-conf-id: 544724832-511663812-3825585790-2681720812
Content-disposition: session
Content-Length: 415
Content-Type: application/sdp


INVITE sip:secondnumber at asteriskserver SIP/2.0
Record-Route:
<sip:serversendinginvites;ftag=67ec0ff3c23d9b35b737206e677cc802o;lr>
Via: SIP/2.0/UDP
serversendinginvites;branch=z9hG4bKf80b.90528f5579fe29e42359a222ba8074ea.0
Via: SIP/2.0/UDP
serversendinginvites:5061;branch=z9hG4bKb1cb9f7680bc7f3ed846dd3c2d9dbac9;rport=5061
Max-Forwards: 16
From:
<sip:originatingcaller at serversendinginvites>;tag=67ec0ff3c23d9b35b737206e677cc802o
To: <sip:secondnumber at serversendinginvites>
Call-ID: EA0785B5-1C7211DF-99B3A4EC-DCEB9CE0 at serverip
CSeq: 200 INVITE
Contact: Anonymous <sip:serversendinginvites:5061>
Expires: 300
User-Agent: serversendinginvites
cisco-GUID: 1293890127-1580750916-857664346-4274469766
h323-conf-id: 1293890127-1580750916-857664346-4274469766
Content-disposition: session
Content-Length: 415
Content-Type: application/sdp

Ignoring this INVITE request 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-19 04:35 MikaelF        Note Added: 0118269                          
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