[asterisk-bugs] [Asterisk 0014244]: No Audio on Call Transfer (Invite not being forwarded to Provider via Asterisk)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=14244 
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Reported By:                mbnwa
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14244
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-01-14 18:13 CST
Last Modified:              2009-09-25 08:48 CDT
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Summary:                    No Audio on Call Transfer (Invite not being
forwarded to Provider via Asterisk)
Description: 
Notes:
Asterisk 1.4.18 & Asterisk 1.6.x effected
Running directrtpsetup=yes
OS Debian
Kernel Version 2.6.26-1-amd64
Called number 13605551212
Caller's number 13605551211
Extension to get transfer: 13605551210
Caller z.z.z.z
Asterisk Server x.x.x.x
Carrier y.y.y.y

Call Flow
13605551211 calls 13605551212 makes the transfer to 13605551210 at this
point call is direct between 13605551212 and 13605551210 but no audio

Issue:
Phone 1 makes an outbound call then transfers to another extension invite
is sent from phone 1 to asterisk, Asterisk ack's however it  never sends
the invite to the carrier to update the audio path resulting in no audio

SIP Trace

(lmadsen: I have removed the inline call trace, and placed it in a text
file as 'original-call-trace.txt' and attached it to this issue. Long
inline traces make open issues difficult to work with.)
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---------------------------------------------------------------------- 
 (0111369) davidw (reporter) - 2009-09-25 08:48
 https://issues.asterisk.org/view.php?id=14244#c111369 
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You need to provide sip set debug output, not, I guess, Wireshark.

It is somewhat of a hassle for me to view it, even if it turns out to be
tcpdump compatible, but I wonder if you actually have a variation of
https://issues.asterisk.org/view.php?id=14385.

PS don't ask if something has been fixed.  You should be reasonably
confident that it hasn't, or should use the forum, irc, or mailing list. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-09-25 08:48 davidw         Note Added: 0111369                          
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