[Asterisk-Users] Issue with DTMF Tones - Codec Issues

[Aaron W]

Thanks I give give that a try.  One follow up question.  If the call
is coming in via the PSTN, and going through the NEAX (PBX) then to
the Cisco, can I control the way the PBX sends the DTMF, or is the
cisco some how able to split out the DTMF tones from everything else?

I was assuming that becuase I am going through the PBX, the cisco
would recieve the DTMF inband, and therefore it would have to send it
out also as inband.

Thanks again
Aaron

On 8/16/05, maka <icokan at gmail.com> wrote:
> just a suggestion, but why don't you try using RFC2833 dtmf relay
> between the cisco and the asterisk box.
> 
> use dtmfmode=rfc2833 in sip.conf, and you can also set the dtmf mode
> per peer in sip.conf
> also, if you use inband dtmf, this would only work with u-law and
> a-law, and not g729.
> 
> on the cisco, enter
> Router(config-dial-peer)# dtmf-relay rtp-nte
> in dial-peer configuration mode.
> 
> I recently had problems with a cisco gw forwarding pstn dtmf digits to
> my asterisk box, and rfc2833(which is what rtp-nte stands for in
> cisco's terms) solved it successfully.
> 
> 
> cheers
> 
> On 8/16/05, Aaron W <walsham at gmail.com> wrote:
> > Topology:
> > PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server
> >
> > When I make a call to a VoIP user from the PSTN, the call gets routed
> > through the PBX, and Cisco.  Because of that the DTMF tones are passed
> > inband, which I can hear on the VoIP end of the call. However, I have
> > one extension on asterisk set up so that I can check voice mail when
> > away from my phone.  When I call that number again via the PSTN, and I
> > am prompted to enter my extension number Asterisk never "hears" the
> > dtmf tones.  I have done some digging around, and my guess is that the
> > issue relates to the codec being used messing up the tones.
> >
> > Am I on the right track? Is there a ideal way to handle this?  what do
> > others do?
> >
> > I have posted my sip.conf below.
> >
> > Thanks,
> > Aaron
> >
> > [general]
> > port = 5060                 ; Port to bind to
> > bindaddr = 0.0.0.0          ; Address to bind to
> > context = default           ; Default for incoming calls (default
> > context has no routing for security purposes)
> > ;dtmfmode=rfc2833
> > dtmfmode=inband
> > srvlookup = yes
> > disallow=all                ; Disallow all codecs
> > ;allow=g729                  ; Codecs that we allow (in order of preference)
> > allow=ulaw
> > ;allow=alaw
> > allow=g729
> > ;allow=ulaw
> > ;allow=all
> >
> >
> > [3120]
> > callerid=Aaron Walsh <3120>
> > type=friend
> > host=dynamic
> > canreinvite=no
> > qualify=yes
> > nat=yes
> > setvar=LDPREFIX=1999999
> > context=XXXXXXX
> > secret=XXXXX
> > mailbox=3120 at XXXXX
> > _______________________________________________
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> 
> 
> --
> I'm sick and tired of being sick and tired...
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