[asterisk-bugs] [Asterisk 0018952]: Receiving DTMF for Queue trough SIP trunk fails if not inband.

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18952 
====================================================================== 
Reported By:                syruscardoza
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18952
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.16.2 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-03-09 22:15 CST
Last Modified:              2011-03-31 12:21 CDT
====================================================================== 
Summary:                    Receiving DTMF for Queue trough SIP trunk fails if
not inband.
Description: 
I am connected with a SIP provider, wich send dtmf via sip info and rfc2833
using gsm, i want to a caller leave the queue by pressing SINGLE digit
extension wich is defined in context configured for the queue.

When callers press the digit nothing happens, if i set dtmfmode=inband,
sometimes works but i have to change my codec to G.711 because inband is
not supported with GSM, sometimes work and sometimes does not because of
the bandwith.

I have no problems with my queues with DAHDI incoming calls.

This is my configuration:

queues.conf
[ventas]
context=callbackventas
queuememberstatus=yes
eventmemberstatus=yes
eventwhencalled=yes
strategy=leastrecent
maxlen=10
timeout=15
retry=1
joinempty = strict
ringinuse = no
servicelevel=20
autopause=yes
setqueuevar=yes

extensions.conf
[testqueue]
exten => 4599,1,Set(CHANNEL(language)=lat)
exten => 4599,n,Set(CDR(userfield)=Entrante)
exten => 4599,n,Queue(ventas,,,,180,,,go-rec,)
exten =>
4599,n,QueueLog(ventas,${UNIQUEID},NONE,${QUEUESTATUS},${CALLERID(num)})
exten => 4599,n,Hangup



[callbackventas]
exten => 1,1,Noop(${QUEUEHOLDTIME})

====================================================================== 

---------------------------------------------------------------------- 
 (0133209) lmadsen (administrator) - 2011-03-31 12:21
 https://issues.asterisk.org/view.php?id=18952#c133209 
---------------------------------------------------------------------- 
A PCAP capture of the inbound audio and SIP trace will be the only way to
triage this issue. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-03-31 12:21 lmadsen        Note Added: 0133209                          
2011-03-31 12:21 lmadsen        Status                   new => feedback     
======================================================================