<div dir="ltr"><div>Hello Mike,</div><div> </div><div> As far as I know hold is handled natively by FreeSwitch and you do not have to do anything in your dialplan.</div><div> </div><div>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.</div>
<div>As far as I recall (I am at home now) the MP supports both; try changing the method and see whether it helps,</div><div> </div><div> __Yehavi:<br><br></div><div class="gmail_quote">2012/6/8 Mike Kallies <span dir="ltr"><<a href="mailto:mike.kallies@gmail.com" target="_blank">mike.kallies@gmail.com</a>></span><br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">Hello Yehavi,<br>
<br>
It was already set to not disconnect. It doesn't drop the call, it just<br>
doesn't unhold. I can hold again and it will play music, then when I<br>
unhold, the connection doesn't come back and the error message reappears<br>
in the logs.<br>
<br>
I'm not sure how 'hold' works in this system. The phone sends a signal<br>
to Freeswitch, then Freeswitch uses the dial plan to go to the hold<br>
music? Does that require negotiation of a new audio protocol or<br>
something? Then when we un-hold, it tries to negotiate back?<br>
<br>
I'm a n00b with Freeswitch. Is there a document I'm missing which<br>
describes how hold works?<br>
<br>
Thanks,<br>
<br>
-Mike<br>
<div class="im"><br>
<br>
On 07/06/2012 11:39 PM, Yehavi Bourvine wrote:<br>
> Search for the parameter DisconnectOnBrokenConnection and set it to<br>
> zero. I don't remember on which screen it is.<br>
><br>
> __Yehavi:<br>
><br>
> 2012/6/8 Mike Kallies <<a href="mailto:mike.kallies@gmail.com">mike.kallies@gmail.com</a><br>
</div>> <mailto:<a href="mailto:mike.kallies@gmail.com">mike.kallies@gmail.com</a>>><br>
<div class="im">><br>
> Hey Everyone,<br>
><br>
> I'm using an Audiocodes MP-118 VoIP Gateway. When making a call through<br>
> the device to a Freeswitch gateway on the other side, it sends the call<br>
> to my desk phone just fine.<br>
><br>
> When I mute the call, we get the hold music as per the dial plan,<br>
> everything is good.<br>
><br>
> Then when I unmute... the hold music stops, and... nothing. The call<br>
> doesn't drop, it just sits there.<br>
><br>
> In the Audiocodes log, we see:<br>
><br>
><br>
> 5d:1h:25m:32s ( lgr_psbrdex)(28879 ) recv <--<br>
> acEV_BROKEN_CONNECTION, Ch:0<br>
> 5d:1h:25m:32s ( lgr_flow)(28880 ) #0:RTP_BROKEN_CONNECTION_EV<br>
> 5d:1h:25m:32s ( lgr_flow)(28881 ) |<br>
> #0:RTP_BROKEN_CONNECTION_EV<br>
><br>
><br>
> I know this isn't entirely a Freeswitch question, but I figure the crowd<br>
> here might know the answer or where I should look for one.<br>
><br>
> Calls through our Grandstream device work fine, so I think this can be<br>
> narrowed down to the Audiocodes device. But maybe there's a different<br>
> codec or option or something I could try to see if I can make it happy?<br>
><br>
> Thanks,<br>
><br>
> -Mike Kallies<br>
><br>
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</div></div></blockquote></div></div>