[Freeswitch-users] Audiocodes MP-118 VoIP Gateway can't un-hold

Yehavi Bourvine yehavi.bourvine at gmail.com
Fri Jun 8 22:33:38 MSD 2012


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 at 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 at gmail.com
> > <mailto:mike.kallies at 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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