[asterisk-users] Help with asterisk and avaya SIP trunking

[Shaun Ewing]

On Mon, Nov 10, 2008 at 2:28 PM, Krishna Sumanth Chava
<kschava at gmail.com> wrote:
> Hi Guys,
>
> Thanks that did help to resolve my issue. i tried the ."@10.10.8.1" and it
> worked and i had a successful call but i have the following 2 concerns.
>
> 1. We have voice communication from avaya to asterisk now but avaya
> is forcing asterisk to use only codec G723. if i disable G723, it says no
> compatible codecs. While the calls from asterisk to avaya are being accepted
> as "alaw"

Make sure you have Compression Mode in your SIP line config on the IP
Office set to your desired codec. You'll run into this problem if you
have it set to "Automatic Select".

Make sure you also reduce the number of codecs on the Asterisk side.
For example, our sip.conf entry looks like:

[ipo-cbr2]
type=friend
username=ipo-cbr2
secret=xxxxxx
host=172.31.2.1
nat=never
context=from-ipo-cbr2
insecure=port,invite
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=no
dtmfmode=auto

To reduce VCM usage, also make sure your IP handsets are using the
same codec. If they are, you won't use any VCM channels for a call.

> 2. I am having issues with DTMF. DTMF is not being recognized or being sent
> from avaya to asterisk.

It should work. Make sure the Asterisk side has dtmfmode=auto like above.

-Shaun