[Asterisk-Users] Livevoip still no DTMF?

[Rich Adamson]

> > I read in the archives a number of discussions about livevoip, DID,
> > and DTMF not working.
> > 
> > However, no resolutions.
> > 
> > I just setup a livevoip DID and indeed the DTMF does not work.
> > 
> > I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
> 
> Thanks to all who responded.
> 
> DTMF still doesn't work.   Basically people reported that their
> 800 DID IAX or SIP all worked fine.
> 
> extensions.conf:
> 
> [from-broadvoice]
> exten => s,1,Goto,IVR|s|1
> 
> [from-livevoip]
> exten => 1234567890,1,Goto,IVR|s|1
> 
> [from-sipmedia]
> exten => s,1,Goto,IVR|s|1
> 
> [IVR]
> exten => s,1,Wait,1
> exten => s,2,Answer
> exten => s,3,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
> exten => s,4,ResponseTimeout,10         ; Set Response Timeout to 10 seconds
> exten => s,5,BackGround(welcome)
> 
> 
> Dialing in from broadvoice, sipmedia, and sip directly to the server
> all work fine.  (all are SIP).  The IAX connection from livevoip does
> not work from whatever source you might dial in.
> 
> 
> Perhaps it has something to do with the recent changes that occured
> at livevoip, or perhaps I am just unlucky.
> 
> iax.conf:
> 
> [general]
> bandwidth=high
> allow=all
> jitterbuffer=no
> tos=low
> register => 1234567:01234567890 at 217.160.244.186
> 
> [livevoip]
> type=friend
> secret=1234567890
> deny=0.0.0.0/0.0.0.0
> permit=217.160.244.186/255.255.255.0
> context=from-livevoip

As noted in my previous post, I'm configured just like you have
shown above with the exception that I also have:
 type=user
 disallow=all
 allow=gsm
 trunk=no
in the above context. Limiting the session to gsm _might_ influence
the form of dtmf that is allowed, but that's a pure guess.

Also, I'm using current CVS-Head (not v1.x) code. If I recall
correctly, seems others have complained about various dtmf
problems somewhere around v1.06/v1.07.

If the allow=gsm doesn't impact your problem, then I'd try
the cvs-head code to see what impact it has.