[asterisk-users] Voicemail, Asterisk and Grandstream BT200
Daniel Bareiro
daniel-listas at gmx.net
Mon Mar 22 04:25:56 CDT 2010
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Hi, Alyed.
On Mon, 22 Mar 2010, Alyed wrote:
>> I was with the following situation: if I call from a cell phone, my
>> Asterisk take the call, it presents to the caller the possibility to
>> dialing an extension number and, in case of not doing it, it
>> transfers this call to a specific extension.
>>
>> Then, if in this extension nobody takes the call, the service of
>> voicemail is triggered so that the caller leaves its message from the
>> cell phone. But if it hangs after to let the message without have
>> pressed previously the pound key, the channel is taken and no longer
>> any other call enters the PBX from the PSTN. This does not happen if
>> the caller presses the pound key after to have left his message.
>>
>> As I have a box at which the cable arrives from the PSTN in which
>> there are two ports of derivation and in one of them it leaves the
>> cable for the Asterisk PBX (connected only then), after to have
>> detected this problem I tried connecting in the other port an analog
>> telephone and, indeed, it did not have tone as if never it had been
>> hung. In addition this was confirmed because the MWI light never
>> blinked on the telephone.
>>
>> After restarting the Asterisk server, yes the MWI light blinks and in
>> addition I could corob the time in which the channel was "taken"
>> seeing that the message lasted more than nine minutes.
>>
>> To what this problem can be due? It has to do the call is made
>> specifically from cell phone through the PSTN (because if I leave a
>> message hanging directly without pressing the pound key from an local
>> extension, this does not happen)? There is some form to avoid it?
> Make sure you have
> busydetect=yes
> busycount=3
>
> somewhere below your [general] context in chan_dahdi.conf (or
> zapata.conf depending on your asterisk version) and restart the the
> service.
>
> This should be enoough to do the magic.
It didn't have configured these two parameters so I added now them but
in the [channels] context since I don't have a [general] context (It
does not sound to me that in the file by default generated by Asterisk
there would not be it either, although I can be mistaken).
Beyond that, with these two parameters, I no longer have the problem
mentioned before. Thanks!
However, the following doubt arises to me: it would also have had this
problem for some originating call from a telephone that is not a cell
phone?
Thanks for your reply.
Regards,
Daniel
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