[Asterisk-Users] Problem with intermittent one-way audio

[Barry Flanagan]

Peter Fern wrote:
> I've had the same problem with all boxen running the same version.  We
> ditched IAX2 for SIP and it has been working fine since.
> 

Well, upgrading my remote site to 1.2.5 appears to have fixed my issues.

-Barry


> Doug Lytle wrote:
> 
>> Barry Flanagan wrote:
>>
>>> Hi,
>>>
>>> I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
>>> connect to a 1.2.5 box for PSTN. There are 15 users on the remote
>>> server, all connecting via SIP softphones.
>>>
>>> For some reason, there is an increasing number of calls where the callee
>>>  does not get any audio although the caller can hear them perfectly.
>>>   
>>
>> I've had this problem in the past, when not running the same version
>> of Asterisk on both ends of the trunk.
>>
>> Doug
>>
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-- 

-Barry Flanagan