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

Peter Fern pete at keypoint.com.au
Mon Mar 20 19:19:19 MST 2006


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.

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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