Erick Perez
2006-Oct-30 09:17 UTC
[asterisk-users] 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 ------------------------------------------------------------
Nic Bellamy
2006-Oct-30 20:49 UTC
[asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
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/
Tzafrir Cohen
2006-Oct-31 12:58 UTC
[asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
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.-13What 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
Erick Perez
2006-Oct-31 15:00 UTC
[asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
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 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------