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

[maka]

I took a look at the NEAX brochures available from NEC's website. I
may be wrong but I don't think you could change the way dtmf tones are
sent from the PBX, but you should be able to send them out of band
(with RTP, as per RFC 2833) from the cisco to the asterisk box.

Generally, out of band dtmf is always better (when available) and more
reliable than inband dtmf. Bear in mind that certain phones, such as
grandstreams, do not work well with rfc2833 dtmf relaying, but work
well with dtmf sent in SIP INFO messages.

cheers

On 8/16/05, Aaron W <walsham at gmail.com> wrote:
> 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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> >
> >
> > --
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-- 
I'm sick and tired of being sick and tired...