[Asterisk-Users] asterisk-grandstream call

[Paul Liew]

Hi Bill,

Your problem seems to be a codec negotiation issue, I think you need to
specify for each SIP peer:
disallow=all
allow=alaw
allow=ulaw    ; and any others that you might need

Paul

> ----- Original Message ----- 
> From: Bill Michaelson
> To: asterisk-users at lists.digium.com
> Sent: Tuesday, February 10, 2004 9:26 AM
> Subject: Re: [Asterisk-Users] asterisk-grandstream call
>
>
> Arg.. my posting was mangled by text-wrapping.  Sorry.
>
> Here again...
> sip.conf:
>
> [general]
> port = 5060       ; Port to bind to
> bindaddr = 0.0.0.0      ; Address to bind to
> context = default    ; Default for incoming calls
> [248379]
> username=billdesk
> type=friend
> host=dynamic
> canreinvite=no
> mailbox=1234
> context=demo