[asterisk-bugs] [Asterisk 0018721]: UA2 not receiving re-INVITE after Asterisk redirect action

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18721 
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Reported By:                soliax
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18721
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.16.1 
JIRA:                        
Regression:                 No 
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:             2011-02-01 05:30 CST
Last Modified:              2011-02-08 04:25 CST
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Summary:                    UA2 not receiving re-INVITE after Asterisk redirect
action
Description: 
<Problem description>
When performing the following redirect action from AMI we came across a
problem where the re-INVITE was not sent to the receiver.

<Asterisk version>
1.6.2.16.1

<OS>
Linux (CentOS)

<Operation>

?UA-1's extension: 20000
?UA-2's extension: 20001
?
?1. Originate Action from AMI from SIP UA-1 to UA-2

action: originate
channel: SIP/20000
callerid: 20000
async: true
priority: 1
exten: 20001
?
?2.From Asterisk, SIP UA-1 receives initial-INVITE so it returns "180
RING" and "200 OK"
??
?3.From Asterisk, SIP UA-2 receives initial-INVITE so it returns "180
RING" and "200 OK"

? 4. Asterisk sends both SIP UA-1 and UA-2 re-INVITE and the RTP route is
established as a Peer2Peer between each UAs' local IP addresses.

?5. Using the following AMI redirect action, we redirect each channel to a
MeetMe number (or hold queue)

action: redirect
channel: SIP/20001-0000000f
priority: 1
exten: 9060000
extrachannel: SIP/20000-0000000e
extrapriority: 1
extraexten: 9060000

? 6. UA-1 receives re-INVITE but UA-2 does not. Because of this, UA-2's
media stream doesn't update. In the case of MeetMe, the sound for both UAs
is mute. In the case I've outlined in this document with the accompanying
logs we tried to redirect to the 9060000 hold queue and the problem was
UA-1 can hear the hold queue music and UA-2's voice but UA-2 can only hear
the hold music.

<Asterisk settings>
-user.conf
??
??[20000]
??type=friend
??secret=20000
??host=dynamic
??nat=no
??callerid=Sip<20000>
??context=default
??disallow=all
??allow=ulaw
??canreinvite=yes
??dtmfmode=rfc2833
?
-extensions.conf (extract)
?
??; Extension dial plan
??exten => _2XXXX,1,Dial(SIP/${EXTEN},60)
??exten => _2XXXX,n,Ringing()
??exten => _2XXXX,n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"]?102)
??exten => _2XXXX,102,Busy(3)
??exten => _2XXXX,103,HangUp()
??
??;MeetMe number dial plan
??exten => _912,1,MeetMe(${EXTEN},dq)
??
??; Hold queue dial plan
??exten => _906XXXX,n,Queue(${EXTEN})
??
-meetme.conf (extract)
??conf => 912
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---------------------------------------------------------------------- 
 (0131651) soliax (reporter) - 2011-02-08 04:25
 https://issues.asterisk.org/view.php?id=18721#c131651 
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Hi,

Sorry about the large ZIP files. Please ignore them.

We were able to reproduce the same problem on Asterisk 1.8 as well.

You can see the new log files + Wireshark logs (with the filters on for
RTCP and SIP protocols only so hence a lot smaller in size). All the log
files have the Asterisk version plus the "20110208" date in their
filenames.

Please let me know if there is any more info you need from me.

Regards,
Simon. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-08 04:25 soliax         Note Added: 0131651                          
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