[Asterisk-Users] One way audio

[administrator tootai]

Wilson Pickett a écrit :

>>What I face is that a SIP call to our GW has from time to time the
>>behaviour to "loose" audio. Hanging up and retrying can work, but mostly
>>we wait or use an IAX GW and try again and then it work. Can also take
>>few hours before it work again.
>>    
>>
>
>What RTP ports are used in asterisk and do the match those of the phones?
>  
>
Asterisk:
rtpstart 6970
rtpend 7170

ATA186:
RTP 5004

Remember that I face this problem from time to time only
-- 
Daniel