[asterisk-users] One-Way Audio after Hold

[Mike Diehl]

Brent Torrenga wrote:
>
> I have an Asterisk 1.6.2 server on a public IP, Cisco 7940 on the 
> localnet, and a trunk to Sipphone/Gizmo/Google Voice.  The externhost 
> and localnet parameters are all set correctly in sip.conf.  An inbound 
> call from Sipphone works great until the local channel places the call 
> on hold.  During hold, the Sipphone user cannot hear music, only 
> silence.  The silence continues after the hold, though the local phone 
> can hear the Sipphone user.
>
>  
> Every possible combination of nat=yes, no, maybe, possibly or never 
> gives the same result.  Further, canreinvite=yes/no/nonat has no 
> result.  I suspect a possible reinvite issue with Asterisk being out 
> of the RTP stream, so I have tried all the usual variables in the 
> DialI() command as well to no avail. 
> Any thoughts on how to fix one-way-audio after a hold?
>
I have the same problem, only my customers report that it only happens 
occasionally.  Most of the time, they can transfer calls just fine.  
They can also put calls on hold and retrieve them as expected.  However, 
sometimes, about once a day, they try to recover a call and the caller 
can't hear them, but they can hear the caller.

I've seen this happen once, but I've been unable to reproduce it reliably.

Any ideas?

Mike Diehl.
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