asterisk users - Jan 2005 - Cisco 7905/7912, SIP, g729 and DTMF setup

If anyone sees any mistakes in the following advice, please let me know.

I recently went through a bit of a configuration nightmare with the
Cisco 7905 phone using the g729 codec and Asterisk and I thought I share
it here for anyone who might be searching for help on this in the
future.

The setup is three, 7905 Cisco phones with the SIP firmware attached to
a Asterisk server remotely through a NAT firewall. The Asterisk is
connecting to the PSTN via a SIP gateway (a Mediatrix box) which only
uses g711.

This setup was actually fairly easy but the one nagging problem was the
DTMF tones. We tried numerous configurations with different results.

Sometimes DTMF tones would work on outbound calls but not on the
Asterisk voice mail system. Other times they would work for voicemail
but no tone would be heard on the outside call. Even more frustrating,
sometimes we could get DTMF if the call was placed outbound, but
incoming calls had no DTMF.

Anyhow, here is what I learned.

1. When using a Cisco phone with the g729 codec, your sip.conf should be
as follows (simplified):

[XXXXXXX]
type=friend
context=local
username=XXXXXXX
callerid=XXXXXXX
secret=XXXXXXX
host=dynamic
mailbox=XXXXXXX
nat=yes
qualify=yes
dtmfmode=rfc2833 ; * See note.
canreinvite=no
disallow=all
allow=g729

* Note: If you use g729 you can not use "inband". Documentation on the
voip-wiki seems to indicate that you should use "dtmfmode=info" with
the
Cisco phone but I found this does NOT work end-to-end with outbound,
inbound, and voicemail system.

The settings on the Cisco phone are also very important. They should be:

RxCodec:3 ;g729
TxCodec:3 ;g729
AudioMode  :0x00000020 ; DTMF signalling Always out-of-band

* Note: remember you have to buy g729 licenses for Asterisk from digium.

On the flip side, the gateway is set as follows:

[mediatrix]
type=peer
context=mediatrix
host=xxx.xxx.xxx.xxx
dtmfmode=inband ; inband works with g711 only
disallow=all
allow=ulaw
allow=alaw

I hope this helps someone.

-- 
John Lange

Hi John/forum
I saw in your config that you are using a Mediatrix box,
I have problems of delay routing for all my calls to the PSTN,
 
Im also using a Mediatrix box (1204 ) version  2.4.10.68

I was wondering if you are using the same Mediatrix box and if you have the same
problem?
Or maybe you can help me with this issue.

For OUTGOING

My Dialplan for the Mediatrix box is the following, here at Mexico we use 8
digits for local calls.

([1-9]xxxxxxx|01xxxxxxxxxx|1111|060|0xx)

I have verified that inmediatly after I dial from my IP phone, the in-use light
turns on in Mediatrix but the call is not pass until the 4 seconds timer
expires.

I have tried disabling the Dial plan but it didnt help

Thanks!




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