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

[Krishna Sumanth Chava]

HI Shaun and Robb,

Thanks for the assistance.

I was able to force the codecs and had avaya talk in the right way. Also
addressed the DTMF issues.

I seem to be having issues with asterisk and avaya not detecting Hang ups.
i am using the Analog phones connected to the POTS ports on the IP Office. I
will try connecting the avaya Analog and Avaya IP Phone to IP Office and see
if that makes any difference.

Thanks again.

Regards
Krishna

On Mon, Nov 10, 2008 at 12:04 AM, Shaun Ewing <sewing at gmail.com> wrote:

> 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
>
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