asterisk users - Apr 2005 - Re: Livevoip IAX DTMF troubles

> Well I've boiled this problem down to the basics and it is utterly
> repeatable and looks to be an asterisk problem from my ignorant
> point of view.
> 
> Basically my asterisk box seems oblivious to incoming DTMF tones
> coming via calls from livevoip via IAX, though it works for
> other SIP providers.
> 
> I reduced the problem to a simple demonstratable and repeatable
> case set.
> 
> Installed a completely fresh asterisk from CVS HEAD.
> Asterisk CVS-HEAD-04/04/05-13:28:37
> 
> Made the following changes from in the "factory" configuration:
> 
> iax.conf:
> 
> added to the [general] section:
> 
> register => xxxxxxx:xxxxxxx@217.160.244.186
> 
> added to the end of the file
> 
> [livevoip]
> type=friend
> secret=xxxxxxx
> deny=0.0.0.0/0.0.0.0
> permit=217.160.244.186/255.255.255.0
> context=from-livevoip
> 
> 
> extensions.conf:
> 
> added to the end of the file
> 
> [from-livevoip]
> exten => 6505572300,1,Goto,demo|s|1
> 
> 
> Above is the sum total of all the changes to a "factory" install.
> 
> Dial 16505572300 (feel free to do this yourself) and the demo
> shows up and does not respond to DTMF tones.
> 
> If you look at a 'iax2 debug' log you will see things like:
> 
> Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF    Subclass: 6
>    Timestamp: 15832ms  SCall: 00002  DCall: 00167 [217.160.244.186:4569]
> 
> which seem to indicate the codes are making to my local asterisk box,
> or at least are not making it to the IVR system.
> (I pressed a six)
> 
> If I change to sipmedia or broadvoice (adding them above) and then
> dial in via them (both SIP rather than IAX) it all works correctly.
> 
> thoughts?
Cross posted on purpose (since this was posted to -dev and some folks
on -users may have an interest).

To bring some level of closure to the above and document the actual
findings that resulted from my analysis of the OP's problem, the
issue with the above is:
 - LiveVoip (Level3) was not sending the dtmf in iax2 packets, rather
   the tones were arriving inband. (I used both Ethereal and iax2 debug
   to verify.)
 - Since the OP was using iax2 with g711 to LiveVoIP, the tones were
   arriving at his * box via inband audio, and given the debug shown
   above (Tx-Frame), * interpreted the inband dtmf and actually sent
   the tone "back" to LiveVoip in an outbound iax2 control packet.

LiveVoip has acknowledged the problem and is working to resolve it.
Its not an asterisk issue.

Since LiveVoip indicated the problem exists for about 5% of their
DID's, the user could probably ask for a different DID, possibly
change to an 800 number, possibly change protocol from iax to sip 
where dtmf inband is supported, wait for a livevoip fix, etc, etc.

Rich

<SNIP>

> > If you look at a 'iax2 debug' log you will see things like:
> >
> > Tx-Frame Retry[000] -- OSeqno: 006 ISeqno: 007 Type: DTMF
Subclass: 6
> >    Timestamp: 15832ms  SCall: 00002  DCall: 00167
[217.160.244.186:4569]
> >
> > which seem to indicate the codes are making to my local asterisk
box,
> > or at least are not making it to the IVR system.
> > (I pressed a six)
> >
> > If I change to sipmedia or broadvoice (adding them above) and then
> > dial in via them (both SIP rather than IAX) it all works correctly.
> >
> > thoughts?
> 
> Cross posted on purpose (since this was posted to -dev and some folks
> on -users may have an interest).
> 
> To bring some level of closure to the above and document the actual
> findings that resulted from my analysis of the OP's problem, the
> issue with the above is:
>  - LiveVoip (Level3) was not sending the dtmf in iax2 packets, rather
>    the tones were arriving inband. (I used both Ethereal and iax2
debug
>    to verify.)
>  - Since the OP was using iax2 with g711 to LiveVoIP, the tones were
>    arriving at his * box via inband audio, and given the debug shown
>    above (Tx-Frame), * interpreted the inband dtmf and actually sent
>    the tone "back" to LiveVoip in an outbound iax2 control
packet.
> 
> LiveVoip has acknowledged the problem and is working to resolve it.
> Its not an asterisk issue.
> 
> Since LiveVoip indicated the problem exists for about 5% of their
> DID's, the user could probably ask for a different DID, possibly
> change to an 800 number, possibly change protocol from iax to sip
> where dtmf inband is supported, wait for a livevoip fix, etc, etc.
> 
> Rich
> 
Not meaning to be completely off topic here, as I am not completely up
to speed on all the protocols, but could this issue that LiveVoIP has
acknowledged also be related to the ringback issue with IAX everyone has
had??

Robert