[Asterisk-Users] VoicePulse Connect & DTMF Tones

[Brian Mulligan]

Interesting this. With SIP this usually a mismatch between how DTMF is
propagated between endpoints, i.e. RTP, SIPinfo or RFC22833. Whichever
method is used both endpoints need to use the same. But you are using IAX so
I assume this is not an issue. Perhaps your provider of the incoming calls
need to set some variable or other. Maybe its a codec thing, not helpful I
know....

Brian

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Sam Bacsa
> Sent: 31 March 2004 11:31
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] VoicePulse Connect & DTMF Tones
>
>
> I am using VoicePulse Connect! to connect to my Asterisk server through
> IAX2.
>
> It appears that when anyone dials my DID and proceeds to dial an
> extension when prompted, the DTMF tones are not received by the server
> (as in, the server continues to play the incoming message in a loop, not
> realizing an extension has been entered).  I have even tried
> automatically transferring to a VM box and seeing if I can log in -- I
> cannot.
>
> Is there any way to fix this?  What are the configuration parameters to
> get DTMF to work on incoming calls?
>
> I am 100% sure that my extensions.conf configuration is in order, so it
> is not an Asterisk problem.
>
> Please Help!
>
>
> Thanks,
> Sam Bacsa
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