[Asterisk-Users] SIP Asterisk Polycom Reinvite

[Damon Estep]

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

 

 

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