[asterisk-users] Issues with Asterisk 1.4 Beta

[Erick Perez]

My 2 cents but im still playing with 1.4
Issue 5: on the phones disable "silence supression" or set to "yes"
the "transmit silence" option. I am not sure if that is the nameof the
option in Swiss phones but the whole idea is to *not* save bandwidth
when the line goes silent (because both sides stop talking).
Make sure ALL SIP phones have disabled silence suppression
you may as well take a look at: bug 5374, which allows Asterisk to
communicate with
devices that support silence suppresion; bug 5409, comfort noise
generation in Asterisk; and bug 1234.

cheers,

On 10/12/06, Jason Walker <jason at jasonsolves.com> wrote:
> I thought I would list my issues so all of you that know more than me
> might be able to help.
>
> 1. I have 6 Swissphone ip10 they disconnect calls at either 70 seconds,
> 120 seconds or 180 seconds I have polycom Phones that go forever
> 2. When I try and transfer calls I have a LONG delay before the seconds
> "line" is usable.  Call1 on hold then make second call and 1 minute
> passes before it attempts a connect
> 3. I have many Polycom 501s and I cannot seem to get the tick server to
> work. I change settings but it does nto fetch the time
> 4.I get "   -- Got SIP response 500 "Internal Server Error" back from
> 192.168.0.XXX" from all my Polycom 501 phone every 2 mintues or so
> 5. I get "[Oct 12 08:49:56] NOTICE[29165]: rtp.c:708 process_rfc3389:
> Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off
> on client if possible. Client IP: 192.168.0.141" on my Swiss phones
>
> Any help would be great.  I am a little new to asterisk and so if I
> posted this incorrectly please let me know
>
> Jason Walker
>
>
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-- 
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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