[Asterisk-Users] SIP Asterisk Polycom Reinvite

[Sean Cook]

hahah... I have run into that dozens of times... you can even pull on 
the cord a bit and it seems tight... then you give it a little more push 
and <click>... you spend about 5 minutes thinking geez I am an idiot...

Sean

Matthew T. O'Connor wrote:
> 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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