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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