[Asterisk-Users] SIP Asterisk Polycom Reinvite

[Matthew T. O'Connor]

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
> 
>  
> 
>  
> 
> 
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