[asterisk-users] SIP_CODEC variable

[Mike]

Thanks, I took it for granted that the phones did support gsm...silly me.

Mike

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Jared Smith
> Sent: Wednesday, February 25, 2009 9:15
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP_CODEC variable
> 
> On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
> > I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
> > appropriate Page command call. But I get this in th CLI:
> 
> > NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
> > ${SIP_CODEC} variable because it is not shared by both ends.
> 
> This is a wild guess (and I don't currently have the time to check it
> out properly), but if my memory serves me the Polycom phones don't
> support the GSM codec.  You might try ulaw instead.
> 
> 
> 
> --
> Jared Smith
> Digium, Inc. | Training Manager
> 
> 
> 
> 
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