asterisk users - Oct 2006 - Grandstream ATA 286 tdm400 and Asterisk 1.2-13

Hi people,

I would like to read your suggestions as to where the issue might be.
ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port.
TDM04B= 4 FXO signal fxls
There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
will not make mention of it.

PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13

Asterisk is being used as a meetme server for 8 more calls.

Everything works fine in terms of the asterisk/meetme. The issue
arises when the calls comes in via the ATA286 box and in any part of
the meeting the CALLER hangs up but the ata286 does not realize the
caller hung up so the channels remains open and everyone in the room
hears a "busy" signal. After 30 seconds the ATA286 hangs up (I guess
due to timeout) and then the tdm04b hungs the channel and then the
meetme room gets back to normal.

This is an ATA286 issue right? nothing to do with the TDM or the asterisk box?
Since I do not own the ATA286 (the voip provider does) would you
recommend something to be asked/changed to the provider of the ATA?

Thanks,

-- 
------------------------------------------------------------
Erick Perez
------------------------------------------------------------

Erick Perez wrote:
> PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13
>
> Asterisk is being used as a meetme server for 8 more calls.
>
> Everything works fine in terms of the asterisk/meetme. The issue
> arises when the calls comes in via the ATA286 box and in any part of
> the meeting the CALLER hangs up but the ata286 does not realize the
> caller hung up so the channels remains open and everyone in the room
> hears a "busy" signal. After 30 seconds the ATA286 hangs up (I
guess
> due to timeout) and then the tdm04b hungs the channel and then the
> meetme room gets back to normal.
The ATA will be getting the hangup - it'll be what's generating the busy
tone you hear when the SIP session between the ATA and your VoIP 
provider is terminated.

If you can get your provider to enable the P205 "Polarity Reversal" 
setting on the ATA, the ATA will reverse the polarity of the voltage on 
it's FXS port when and outgoing call is answered (outbound calls), and 
when the remote end hangs up (for calls in either direction).

You'll then be able to set hanguponpolarityswitch=yes in zapata.conf, 
and hangups should then be detected almost immediately (with luck, 
before any tones are heard).

HTH,
    Nic.

-- 
Nic Bellamy,
Head Of Engineering, Vadacom Ltd - http://www.vadacom.co.nz/

On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez
wrote:
> Hi people,
> 
> I would like to read your suggestions as to where the issue might be.
> ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS 
> port.
> TDM04B= 4 FXO signal fxls
> There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
> will not make mention of it.
> 
> PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13
What exactly is the point is such settings? Why not connect directly to
the provider over SIP? Or to the ATA over SIP?

-- 
               Tzafrir Cohen       
icq#16849755                    jabber:tzafrir@jabber.org
+972-50-7952406           mailto:tzafrir.cohen@xorcom.com       
http://www.xorcom.com  iax:guest@local.xorcom.com/tzafrir

I forgot to mention that the Carrier that owns the ATA box was not
willing to let me connect directly over IP, I was only allowed to use
the FXS port. He already ack that he has a problem with
disconnections.


On 10/31/06, Tzafrir Cohen <tzafrir.cohen@xorcom.com>
wrote:
> On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote:
> > Hi people,
> >
> > I would like to read your suggestions as to where the issue might be.
> > ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one
FXS
> > port.
> > TDM04B= 4 FXO signal fxls
> > There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
> > will not make mention of it.
> >
> > PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13
>
> What exactly is the point is such settings? Why not connect directly to
> the provider over SIP? Or to the ATA over SIP?
>
> --
>               Tzafrir Cohen
> icq#16849755                    jabber:tzafrir@jabber.org
> +972-50-7952406           mailto:tzafrir.cohen@xorcom.com
> http://www.xorcom.com  iax:guest@local.xorcom.com/tzafrir
> _______________________________________________
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-- 
------------------------------------------------------------
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
------------------------------------------------------------