[Asterisk-Users] Re: Livevoip still no DTMF?

[Eric Wieling aka ManxPower]

Brian Litzinger wrote:
> On Fri, Apr 01, 2005 at 12:12:57PM -0600, Eric Wieling aka ManxPower wrote:
> 
>>Brian Litzinger wrote:
>> > 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
>>>
>>>sip.conf:
>>>I have dtmfmode=inband for both sip.media.com and sip.broadvoice.com
>>>and both are limited to ulaw, alaw.
>>
>>Get rid of the bandwidth= statement.  In the [livevoip] put disallow=all 
>>and allow=ulaw (or the ONE codec you want to use).  Also comment out the 
>>tos=low just to see if that makes any difference.
> 
> 
> By your command...
> 
> Made the suggested changes.  Called in via SIP and Cell Phone.  Still
> no response to DTMF.
> 

It was worth a try. 8-)  Try allow=gsm instead, but I doubt it will make 
any difference.  Your other option is to just switch providers.