[asterisk-bugs] [Asterisk 0017279]: MOH not playing when phone transfer softkey pressed

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17279 
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Reported By:                scharrua
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17279
Category:                   Channels/chan_sip/Transfers
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.30 
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:             2010-05-03 18:23 CDT
Last Modified:              2010-05-07 10:44 CDT
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Summary:                    MOH not playing when phone transfer softkey pressed
Description: 
Hi!
the client's setup is as follows: digium 1FX0 1FXs card on dahdi 2.2.0,
Asterisk 1.4.27, 1 Yealink T28P amd 5 Yealink T22P, 1 GSM Gateway 1SIM
card
the problem is: caller A (from mobile or fixed line phone) calls client. A
RingGroup is set and correctly passes call to the extensions. Operator
(T28P) extension B answers the call and puts caller on hold and transfers
call to extension C. While on hold, caller A should hear MOH, but, despite
being correctly set, no music is started, nothing is heard.
On CLI, when operator presses the Transfer softkey on the phone, nothing
appears, suggesting that there is no SIP request for transfer, whatsoever.
Strangely, the same setup but using a OpenVox B200P (2BRI card) and mISDN
drivers does everything correctly. This is the only diference between the 2
clients.
i've checked musiconhold.conf, features.conf, extensions.conf, everything
seems correctly set!
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 (0121553) scharrua (reporter) - 2010-05-07 10:44
 https://issues.asterisk.org/view.php?id=17279#c121553 
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sorry for the late answer.

i erroneously referenced the 1.4.30, it was in fact 1.4.27.

i followed your instructions, enabled Logger and Sip debug, etc... and
indeed, the log file doesn't show any request for Attended Transfer or Hold
(unless operator uses *8).

I did solved the problem by updating to 1.4.31. I only updated Asterisk
code, haven't touched ANY of the .conf files, and everything is now working
perfectly. So i suspect it might be something with chan_sip in 1.4.27 (just
a wild guess...)
I will attach the requested files ASAP.

Please note that i'm using Asterisk GUI lastest SVN version with many mods
coded by me, but none is relevant to MOH or to this particular case.

Thanks once again.

Sergio Charrua
www.voip.pt - Voice Over IP Portugal 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-07 10:44 scharrua       Note Added: 0121553                          
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