asterisk users - Sep 2006 - SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble

I am using the following setup:

Linksys SPA941 -> Asterisk -> NuFone -> PSTN -> ShoreTel system

The system works great for the most part.  Most people I call say it
sounds good.  However, every time I call a certain company that uses a
ShoreTel system, they claim the sound is garbled (understandable, but
not pleasant to listen to).  Everything sounds fine at my end.  If I
make a call w/ the following setup, it sounds fine:

Analog phone -> Asterisk:TDM400 -> NuFone -> PSTN -> ShoreTel system

So, it seems there is some type of weird interaction between my system
and the ShoreTel system if I use the SPA941 IP phone.

Does anyone have suggestions as to how I can start debugging this?

Thanks,
Cliff

-- 
======================Cliff Brake
http://bec-systems.com

Cliff Brake wrote:
> I am using the following setup:
> 
> Linksys SPA941 -> Asterisk -> NuFone -> PSTN -> ShoreTel system
> 
> The system works great for the most part.  Most people I call say it
> sounds good.  However, every time I call a certain company that uses a
> ShoreTel system, they claim the sound is garbled (understandable, but
> not pleasant to listen to).  Everything sounds fine at my end.  If I
> make a call w/ the following setup, it sounds fine:
> 
> Analog phone -> Asterisk:TDM400 -> NuFone -> PSTN -> ShoreTel
system
> 
> So, it seems there is some type of weird interaction between my system
> and the ShoreTel system if I use the SPA941 IP phone.
> 
> Does anyone have suggestions as to how I can start debugging this?
Check the RTP Packet Size (under the Sip tab). Set it to .020 (20 
milliseconds) and place another test call. For whatever reason, the 
Linksys/Sipura products default to 30 milliseconds and has impacted the 
quality of audio on some systems.

On 9/22/06, Rich Adamson <radamson@routers.com>
wrote:
> > So, it seems there is some type of weird interaction between my system
> > and the ShoreTel system if I use the SPA941 IP phone.
> >
> > Does anyone have suggestions as to how I can start debugging this?
>
> Check the RTP Packet Size (under the Sip tab). Set it to .020 (20
> milliseconds) and place another test call. For whatever reason, the
> Linksys/Sipura products default to 30 milliseconds and has impacted the
> quality of audio on some systems.
Setting the RTP packet size to 20ms seems to have fixed it.  Thanks
for the suggestion.

Cliff

-- 
======================Cliff Brake
http://bec-systems.com