[Asterisk-Users] RE: [Asterisk-Dev] Re: Livevoip IAX DTMF troubles

[Rich Adamson]

> > > 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??

I believe they are two separate issues. The reason for saying that is
my livevoip 800 number suffers from the no ringback issue (but dtmf is
passed to * correctly), and the OP's issue was no dtmf passed via iax2
control packets. It is entirely possible the "no dtmf" might incure
the "no ringback", but testing it wasn't possible since we couldn't
get past the IVR prompts to know whether ringback was present or not.

Since this thread really has nothing to do with -dev anymore, any
additional followup postings should be moved to the -user list.