[Asterisk-Users] SIP Asterisk Polycom Reinvite

Damon Estep damon at suburbanbroadband.net
Thu Apr 6 13:07:32 MST 2006


Thanks,

I have experienced that as well, and we do carefully check the handset
cord now! This is a different issue in this case.

The IP501 handset cord clicks twice going into the handset, the first
click is not good enough! Small design issue on those phones...

Damon

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Matthew T. O'Connor
> Sent: Thursday, April 06, 2006 1:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP Asterisk Polycom Reinvite
> 
> I had a one way audio problem with my Polycom 501's and it turned out
> that the cord wasn't plugged in to the handset all the way.  It looked
> like it was in, but it wasn't in all the way till it clicked.
> 
> Matt
> 
> 
> 
> Damon Estep wrote:
> > Wondering if anyone has experienced an intermittent one way audio
> > (called party can not hear) problem in these conditions;
> >
> >
> >
> > Several IP501 phones local, same subnet.
> >
> > Remote asterisk
> >
> > No NAT anywhere
> >
> >
> >
> > Polycom IP501 ulaw only, canreinvite=yes
> >
> > Asterisk
> >
> > Call termination path is to a sonus GSX operated by the upstream
> > carrier, ulaw only, canreinvite=no
> >
> >
> >
> > The idea is that if the Polycoms are canreinvite=yes and the PSTN
> > termination path is canreinvite=no then calls between polycoms
should
> > not have asterisk in the media stream and wan link utilization is
> reduced.
> >
> >
> >
> > The problem looks like the Polycom keeps trying to reinvite the
sonus
> > and the call never sets up right, and not with all calls...
> >
> >
> >
> > Any experience with this? Maybe there is a totally different issue I
am
> > overlooking?
> >
> >
> >
> > About 3 to 5% of all Polycom to PSTN via asterisk>SIP peer calls are
> > impacted.
> >
> >
> >
> > I have not set the Polycom canreinvite=no yet, hoping to not have to
do
> > that as the wan link is a t1 that is also used for data.
> >
> >
> >
> > Thanks for any help!
> >
> >
> >
> > Damon
> >
> >
> >
> >
> >
> >
> >
------------------------------------------------------------------------
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list