[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:                     new
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:              2011-02-01 05:35 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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---------------------------------------------------------------------- 
 (0131332) soliax (reporter) - 2011-02-01 05:35
 https://issues.asterisk.org/view.php?id=18721#c131332 
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Update:

1. Please excuse the tabs seem to be have mistranslated into ?.

2. Log files I've attached are:
2.1 asterisk_cli.log - Asterisk console log with sip history and debug
turned on
2.2 CSP20000_reinvite_asterisk2_10.zip - ZIP of Wireshark log from PC
running extension UA-1's 20000
2.3 CSP20001_reinvite_asterisk2_10.zip - ZIP of Wireshark log from PC
running extension UA-1's 20001 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-01 05:35 soliax         Note Added: 0131332                          
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