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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/2169bee3/attachment.htm
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
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@lists.digium.com [mailto:asterisk-users- > bounces@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 polycomsshould> > not have asterisk in the media stream and wan link utilization is > reduced. > > > > > > > > The problem looks like the Polycom keeps trying to reinvite thesonus> > and the call never sets up right, and not with all calls... > > > > > > > > Any experience with this? Maybe there is a totally different issue Iam> > 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 todo> > 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