[cisco-voip] DTMF fun

[Madziarczyk, Jonathan]

Good stuff all, I think I'm getting a clearer picture.

Ryan - yes we are seeing DTMF tones in the h245 debug "signalType "8"
duration 100".  I also added the rtp-nte in " dtmf-relay rtp-nte
h245-alphanumeric" on the H323 GW and "mgcp dtmf-relay voip codec all
mode nse" on the mgcp gw so they're all using the RFC within the CUCM
domain.  That seems to have helped the consistency of the DTMF tones.
As I understand it, that means that both gateways are using the RFC
standard for out of band.

Mark - yeah, we're using G711ULAW on both of the voice gateways...i've
got the bandwidth, I figured why not?

Jason - Yeah, I think I'm getting close to having to troubleshoot the
PBX now.  What threw me was that the problems only started showing up
when the CUCM was introduced to the path.  Yes there is an ISDN PRI
running DMS-100 on the first PBX.  Sadly, debugging our old NEC means a
few hundred at least to get someone out here who still knows a 12yr old
OS.  But like I said above I think I might have fixed some flakyness by
setting everything to follow the RFC out of band on the CUCM side of
things.  That is to say that things are consistently working or not
working now after repeated calls.  Now to see why some things are not
working when they used to. :)

Jonathan


-----Original Message-----
From: Rhodium [mailto:rhodium_uk at yahoo.co.uk] 
Sent: Wednesday, January 20, 2010 2:56 PM
To: VOIP Group; Madziarczyk, Jonathan
Subject: RE: [cisco-voip] DTMF fun

Guess it's just me mis-interpreting your diagram so apologies:

PSTN --> MGCP GW --> CUCM --> H.323 GW --> PBX#1 --> PBX#2 --> VM

So you are seeing the digits get received on the GW which is good.

So now you have broken the fault down to between the GW and the VM box.

I presume you have an ISDN PRI interface connected to the first PABX? If
you are using ISDN, DTMF tones are carried in band so you won't see it
in the D-channel.

Next step is to debug the first PABX to see if it is getting everything
in band, even if you call an extension on PABX2 from the PSTN and check
whether there is noise or interference when the digits are pressed.

Also, you might want to look at your voice port configuration to see
whether the compand type is correct, the echo-cancellation coverage,
output attenuation, etc.

Might be worth to check those settings as well on the first PABX.

HTH

Jason