[Asterisk-Users] unable to configure my Grandstream phone

[Balaji NJL]

i hv also added the alaw

> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = bogon-calls
> ;context = default
> disallow=all
> allow=g729
> allow=gsm
> allow=ulaw
allow=alaw

now i am able to call from my MSN -> * ->GS but the
other way is not
working. i am getting lot of noise when i try to place
any call from my GS.
i am no longer getting codec not compatible error
message anymore. i am
still unable to place any calls using my GS (to my
internal MSN extensions
or to external PSTN).

thanks for ur help,
-B
>

----- Original Message ----- 
From: "Balaji NJL" <bajjeen at yahoo.com>
To: <asterisk-users at lists.digium.com>
Cc: <pliew at atp.org.au>
Sent: Sunday, December 14, 2003 3:40 PM
Subject: Re: [Asterisk-Users] unable to configure my
Grandstream phone


> Hi Paul,
>
> thanks for the quick response. i tried the following
> configuration /
> combination still no luck
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = bogon-calls
> ;context = default
> disallow=all
> allow=g729
> allow=gsm
> allow=ulaw
>
> when i tried g711 i am getting an error in * that
> codec not found.
>
> When i specify only g729 my MSN doesnt work.
>
> thanks,
> -B
>
> ----- Original Message ----- 
> From: "Paul Liew" <pliew at atp.org.au>
> To: <asterisk-users at lists.digium.com>
> Sent: Sunday, December 14, 2003 2:14 PM
> Subject: Re: [Asterisk-Users] unable to configure my
> Grandstream phone
>
>
> >
> > ----- Original Message ----- 
> > From: Balaji NJL
> > To: asterisk-users at lists.digium.com
> > Sent: Monday, December 15, 2003 8:47 AM
> > Subject: [Asterisk-Users] unable to configure my
> Grandstream phone
> >
> >
> > <snip>
> > > Attempting native bridge of SIP/2003-b895 and
> SIP/2000-53e2
> > > WARNING[5126]: File chan_sip.c, Line 1954
> (process_sdp): No compatible
> > codecs!
> > >     -- Got SIP response 481 "Call
Leg/Transaction
> Does Not Exist" back
> > from 192.168.0.58
> > >
> > > and then the call drops. When i am making a call
> using Grandstream ph,
> it
> > rings the other side when they pick up the phone
the
> call then drops. then
> i
> > get the
> > > above error message.
> > >
> > > the follwoing us sip and Grandstream conf
> > >
> > >
> > > [general]
> > > port = 5060
> > > bindaddr = 0.0.0.0
> > > context = bogon-calls
> > > ;context = default
> > > disallow=all
> > > allow=gsm
> >
> > Balaji,
> >
> > Grandstreams do not support GSM. Options available
> can be seen on the GS
> > config page. Unless you purchase G729s (for low
> bandwidth), your only
> choice
> > is G711.
> >
> > Paul
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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