[Asterisk-Dev] Re: Livevoip IAX DTMF troubles

Robert Webb asterisk at ropeguru.com
Fri Apr 8 04:45:47 MST 2005



<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






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