Hello Mike,
 
  As far as I know hold is handled natively by FreeSwitch and you do not have to do anything in your dialplan.
 
There are two ways for the MP to ask for hold: One is to set the media address to 0.0.0.0 and the other is to set it to send-only or receive-only.
As far as I recall (I am at home now) the MP supports both; try changing the method and see whether it helps,
 
                           __Yehavi:

2012/6/8 Mike Kallies <mike.kallies@gmail.com>
Hello Yehavi,

It was already set to not disconnect.  It doesn't drop the call, it just
doesn't unhold.  I can hold again and it will play music, then when I
unhold, the connection doesn't come back and the error message reappears
in the logs.

I'm not sure how 'hold' works in this system.  The phone sends a signal
to Freeswitch, then Freeswitch uses the dial plan to go to the hold
music?  Does that require negotiation of a new audio protocol or
something?  Then when we un-hold, it tries to negotiate back?

I'm a n00b with Freeswitch.  Is there a document I'm missing which
describes how hold works?

Thanks,

-Mike


On 07/06/2012 11:39 PM, Yehavi Bourvine wrote:
> Search for the parameter DisconnectOnBrokenConnection and set it to
> zero. I don't remember on which screen it is.
>
>                         __Yehavi:
>
> 2012/6/8 Mike Kallies <mike.kallies@gmail.com
> <mailto:mike.kallies@gmail.com>>
>
>     Hey Everyone,
>
>     I'm using an Audiocodes MP-118 VoIP Gateway.  When making a call through
>     the device to a Freeswitch gateway on the other side, it sends the call
>     to my desk phone just fine.
>
>     When I mute the call, we get the hold music as per the dial plan,
>     everything is good.
>
>     Then when I unmute... the hold music stops, and... nothing.  The call
>     doesn't drop, it just sits there.
>
>     In the Audiocodes log, we see:
>
>
>     5d:1h:25m:32s (   lgr_psbrdex)(28879     )  recv <--
>     acEV_BROKEN_CONNECTION, Ch:0
>     5d:1h:25m:32s (      lgr_flow)(28880     )  #0:RTP_BROKEN_CONNECTION_EV
>     5d:1h:25m:32s (      lgr_flow)(28881     )  |
>     #0:RTP_BROKEN_CONNECTION_EV
>
>
>     I know this isn't entirely a Freeswitch question, but I figure the crowd
>     here might know the answer or where I should look for one.
>
>     Calls through our Grandstream device work fine, so I think this can be
>     narrowed down to the Audiocodes device.  But maybe there's a different
>     codec or option or something I could try to see if I can make it happy?
>
>     Thanks,
>
>     -Mike Kallies
>
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_________________________________________________________________________
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http://www.cudatel.com

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http://wiki.freeswitch.org
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