[Asterisk-Users] Local Call Problems

[Seth Remington]

Set canreinvite=no in sip.conf to force the RTP voice traffic to pass
through asterisk so it can do the transcoding.

-Seth

On Tue, 2004-07-13 at 10:46, James Dutton wrote:
> Hi
>  
> Further to my previous email...
>  
> I have a Xten software phone connecting to a Grandstream 100 hardware
> phone. My first problem is that voice transmits in one direction only.
> Secondly, this only works if the codecs on both are identical. If the
> Xten uses GSM and the Grandstream uses ULAW then the phones connect,
> but no voice can be hears in either direction. I assumed (possibly
> wrongly) that Asterisk did the appropriate codec translation?
>  
> Regards
> James
-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559