[asterisk-bugs] [Asterisk 0017953]: INVITE with Replaces: breaks Monitor() call recording.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17953 
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Reported By:                kkm
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17953
Category:                   Channels/chan_sip/Transfers
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.10 
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:             2010-09-05 02:55 CDT
Last Modified:              2010-09-05 16:11 CDT
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Summary:                    INVITE with Replaces: breaks Monitor() call
recording.
Description: 
An INVITE that Replaces: a dialog that had a channel recording with
Monitor() stops recording on that channel.

- A peer sends initial INVITE;
- (180 trying is sent back)
- my dialplan sets up recording of the top channel using Monitor();
- places the call into a queue;
- a queue agent answers the call;
- (here the calls is answered an a 200 is sent)
- (call bridged and recording begins)
- In 400-900 ms from our 200 Ok, another INVITE is sent, with a Replaces:
header pointing to the dialog established with the initial INVITE

And that causes Monitor to stop recording the call. 

Log file (https://issues.asterisk.org/view.php?id=156#c900 lines) with some
minor commentary added. The second invite
is near line 640.
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 (0126635) schmidts (reporter) - 2010-09-05 16:11
 https://issues.asterisk.org/view.php?id=17953#c126635 
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iam not sure cause atleast you should leave the ports in your log to differ
Caller A & Caller B. Are they both on IP 99.99.99.99 or is the sip debug
only from the origin caller?
But if i read the replace header right, the peer (queueagent) with ip
99.99.99.99 which answers the phone want to get the direct media stream:
so asterisk wont see the rtp stream anymore. if this is true, how could
asterisk still monitor the call? 
do you have directmedia enabled in your sip.conf? if yes try to disable
it. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-09-05 16:11 schmidts       Note Added: 0126635                          
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