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

[Krishna Sumanth Chava]

Hi Shaun and Robb,

I tried the Avaya IP small Office with Lucent Analog phones. it worked fine
on hang ups - i think it is my old analog phone is the root cause.

I have only one major issue now.

I am not getting the Caller ID Name and Caller ID number from avaya to
asterisk.

Can you provide me your valuable input.

This is what i have.
When i configured the SIP trunk,

Under SIP line - i had
Primary authentication name = avayanew
 Primary authentication Password = avayanew

and also under the SIP URI:
Local URI, Contact and Display name - i had selected "use authentication
name" for successful calls, but as said caller ID is not passed through
asterisk.

when i try the "use user data" in there, i get "TTel:" the problem i had
before and cannot make/receive calls.

Please advise

Thanks as always

Regards
Krishna
On Mon, Nov 10, 2008 at 6:57 PM, Shaun Ewing <sewing at gmail.com> wrote:

> On Tue, Nov 11, 2008 at 4:56 AM, Krishna Sumanth Chava
> <kschava at gmail.com> wrote:
> > 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.
>
> Glad to hear it.
>
> > 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.
>
> What does SSA show when one end has hung up? If it still shows the
> call as active, then a disconnect signal has gone missing.
>
> I've never experienced this problem, but then again the only thing we
> use the POTS ports for is faxing and this is forced to use our PRI
> circuits. All of our handsets including conference room phones are IP.
>
> -Shaun
>
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