[Freeswitch-users] polycom one-way audio problem

[Matthew Kaufman]

Updated to current a couple days ago, default dialplan, extensions are 
registered as 1001 and 1002.

tcpdump on the freeswitch host shows RTP going both ways to/from both 
phones.

This is running on CentOS, so your log and debug instructions don't make 
much sense, but I do have SIP tracing enabled... see 
http://pastebin.freeswitch.org/6694

The phones are on the 10.10.155.0/24 subnet, the switch is on 
198.202.199.1, but there's no NAT between, just a router... from the POV 
of the switch, it can see all the phones on the 10.10.155.0/24 network 
directly.

Matthew Kaufman

Michael S Collins wrote:
> What FS revision and is this a default Dialplan? Please pastebin the  
> output of the CLI while making test calls. Be sure to press F8 to  
> enable debug messages. Also, if you can do so turn on SIP messages by  
> launching FreeSWITCH with TPORT_LOG=1.
>
> -MC
>
> Sent from my iPhone
>
> On Jan 6, 2009, at 10:22 PM, Matthew Kaufman <matthew at matthew.at> wrote:
>
>   
>> I have two Polycom phones (one 550 and one 650) successfully  
>> registered
>> to the switch. If I call from either extension to the other and  
>> answer,
>> audio flows from the calling party to the called party, but audio does
>> not flow from the called party back to the calling party. Even more
>> strange, the called party does not answer, then when the call is  
>> sent to
>> voicemail, the calling party *also* does not hear any of the voicemail
>> greeting, though they are recorded successfully. Calling the voicemail
>> box directly from either phone *does* work, and the calling party can
>> hear the prompts just fine in that case.
>>
>> There is no NAT between the phones and FreeSWITCH.
>>
>> Suggestions?
>>
>> Matthew Kaufman
>> matthew at matthew.at
>>
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