[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. 
====================================================================== 
https://issues.asterisk.org/view.php?id=14244 
====================================================================== 
Reported By:                mbnwa
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2009-01-14 18:13 CST
Last Modified:              2009-09-25 07:08 CDT
====================================================================== 
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.)
====================================================================== 

---------------------------------------------------------------------- 
 (0111364) jehanzeb (reporter) - 2009-09-25 07:08
 https://issues.asterisk.org/view.php?id=14244#c111364 
---------------------------------------------------------------------- 
HI,
I am having a similar issue with my asterisk setup
I am on asterisk version 1.4.21.2 
My sinerio is almost exactly the same
Sip.cfg is configured for reinvites
i.e. canreinvite= yes
the server layout is as under
handsetA<--->ServerA<--->SeverB(Asterisk)<--->ServerC<--->handsetC,handsetD
and  media server
when a call is made from handset A destined for handset C, asterisk
initially keeps hold of the media while the call is being established and
once this is done server B (asterisk) sends re-invites to both servers  A
and C with the media ip addresses of hanset A and C respectively so that
now the RTP is flowing directly between the handsets.
The problem starts when handset C puts the caller (handset A ) on hold to
transfer call to handset D. this is a blind transfer at this point server C
sends an invite to asterisk server to re-route the media from handset A to
the media server and later it provides the ip adderss for handset D to
route media to in a separate invite. On both occasion the asterisk server
sends an OK back to server C but doesn’t send an invite back to server A
to reroute the media to the new phone. The result is even though handset A
can hear the RTP from Handset D , Handset D is not receiving any RTP from
Handset A I am attaching a trace file by the mane of "transfernortp.cap"
taken at the asterisk server to illustrate my point. Please can you advice
if there is a fix out for this.
Thanks,
Jehanzeb 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-25 07:08 jehanzeb       Note Added: 0111364                          
======================================================================