Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can follow her. It sounds like the Sipura 3000 would be perfect, what else would do it? Chris Mason
I have looked for other FXO SIP Gateways and there are not many to choose from. I found another made by clipcom, but that was about it, other than a small asterisk server. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Mason (Lists) Sent: Thursday, May 05, 2005 8:07 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] FXO ATA? Is the Sipura 3000 the only way to interface a remote pstn line and connect incoming calls to Asterisk? I have a location connected by network that has a phone line, when the room is occupied I want the line ti ring there as normal, but when the employee is travelling I want the line to be conencted to a ATA that then feeds it as an incoming pstn line to the pbx located at my office so it can follow her. It sounds like the Sipura 3000 would be perfect, what else would do it? Chris Mason _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On 5/5/05, Chris Mason (Lists) <lists@masonc.com> wrote:> Is the Sipura 3000 the only way to interface a remote pstn line and connect > incoming calls to Asterisk? I have a location connected by network that has > a phone line, when the room is occupied I want the line ti ring there as > normal, but when the employee is travelling I want the line to be conencted > to a ATA that then feeds it as an incoming pstn line to the pbx located at > my office so it can follow her. > It sounds like the Sipura 3000 would be perfect, what else would do it? >Nothing really can touch the sipura's for features, ease of use, and their very good documentation. Even if there was another product for half the price I would probably still use the spa-3000. Chris
The Grandstream HandyTone 488 has an FXO port. I've never used it though. Cheers, Jon. On Thursday 05 May 2005 07:07 am, Chris Mason (Lists) wrote:> Is the Sipura 3000 the only way to interface a remote pstn line and connect > incoming calls to Asterisk? I have a location connected by network that has > a phone line, when the room is occupied I want the line ti ring there as > normal, but when the employee is travelling I want the line to be conencted > to a ATA that then feeds it as an incoming pstn line to the pbx located at > my office so it can follow her. > It sounds like the Sipura 3000 would be perfect, what else would do it? > > Chris Mason > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
I've been looking at something similar, but with more ports. Something to handle the incoming (FXO) analog lines, but without the investment in a channel bank and T1 card because we only need 4-8 FXOs and no FXS. I've looked at the AudioCodes MP104 which looks like it will take the FXOs and turn them into SIP channels. Anyone have experience with these? Maybe my lack of experience is causing incorrect expectations. While they are pricey (~$1,000US), they are still cheaper than a T1 card and a channel bank I think. On Thu, May 05, 2005 at 08:07:14AM -0400, Chris Mason (Lists) wrote:> Is the Sipura 3000 the only way to interface a remote pstn line and connect > incoming calls to Asterisk? I have a location connected by network that has > a phone line, when the room is occupied I want the line ti ring there as > normal, but when the employee is travelling I want the line to be conencted > to a ATA that then feeds it as an incoming pstn line to the pbx located at > my office so it can follow her. > It sounds like the Sipura 3000 would be perfect, what else would do it?-- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot.
Why not go with Multitech? They are expensive, but great units. Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael George Sent: Friday, May 06, 2005 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO ATA? I've been looking at something similar, but with more ports. Something to handle the incoming (FXO) analog lines, but without the investment in a channel bank and T1 card because we only need 4-8 FXOs and no FXS. I've looked at the AudioCodes MP104 which looks like it will take the FXOs and turn them into SIP channels. Anyone have experience with these? Maybe my lack of experience is causing incorrect expectations. While they are pricey (~$1,000US), they are still cheaper than a T1 card and a channel bank I think. On Thu, May 05, 2005 at 08:07:14AM -0400, Chris Mason (Lists) wrote:> Is the Sipura 3000 the only way to interface a remote pstn line and > connect incoming calls to Asterisk? I have a location connected by > network that has a phone line, when the room is occupied I want the > line ti ring there as normal, but when the employee is travelling I > want the line to be conencted to a ATA that then feeds it as an > incoming pstn line to the pbx located at my office so it can followher.> It sounds like the Sipura 3000 would be perfect, what else would doit? -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
> Why not go with Multitech? They are expensive, but great units.For the same cost I could get a T1 card and a channel bank on Ebay and have change left over. These are exepense units. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759>>
Folks! Let me clarify this for you all. ATCOM's ATA does not have an FXO port. The "Lifeline" port is not an FXO Port. It is an FXS Passthrough port. It does not have any of the FXO features that you are looking for. You cannot do a modprobe on this - nor can you pass your peer traffic to this port. Imagine this to be like an FXS Port with the Handset offhook and ready for you to dial a number to call out using your existing analog line. That's all it does. This helps you make calls using your existing analog line in case of a failure in your IP network. Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jon Gabrielson Sent: Thursday, May 05, 2005 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO ATA? The AG-168E has an FXO port? The only seller I can find seems to think it is just a single FXS port. http://www.iaxtalk.com/product_info.php?products_id=30 You wouldn't happen to have another link with more info would you? Thanks, Jon. On Thursday 05 May 2005 01:33 pm, Joseph wrote:> Indeed SPA-3000 as a lot of features, maybe too many :-). My asterisk> is controlling everything so most of these features just complicate > the setup. I've one SPA-3000 and have on order AG-168VE from ATCOM. > > The AG-168 supports IAX2 and the FXO port is "pass though" type. > The difference is that SPA-3000 answer the phone and rings asterisk > (the phone at this moment has been answered the ringing party is > incurring the charges before asterisk answered the phone), the AG-168 > is ringing the asterisk directly, so I think the "pass through port" > is a benefit in this case for asterisk users.-------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
On Thursday 05 May 2005 10:27 pm, Tim Connolly wrote:> Pass through has the same functionality as a modem with a "line" and a > "phone" connection. Line is where you plug in the dialtone, the dial passes > through the "phone" connection unless the card picks up (like a modem > does). > > I have a X100P clone that is setup as a passthrough. I've never seen a pass > through on a FXS, but then I've only messed with ATA-186's recently. >That is not correct or at best not completely correct. That is what I would have believed it to be, but some passthrus (like the handytones) allow you to dial *00 or some other combination to dial out of the PSTN directly. I guess it is possible that the *00 turns off the FXS to allow the straight passthru, but even if that is so, the X100p doesn't have an equivalent functionality.
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rusty Shackleford Sent: Friday, May 06, 2005 2:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FXO ATA?> Why not go with Multitech? They are expensive, but great units.Because they are ridiculously expensive. It is true that Multitech's VOIP gear is very good stuff. I've used it and it "just works". But apparently, their marketing people haven't been paying attention to the market and they are still using pricing that reflects the market 5 years ago. /End Quote/ I totall agree with that comment. Multitech is just a rip-off, when you compare the products with others existing in the market. Seshu -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
[snip]> I totall agree with that comment. Multitech is just a rip-off, when you > compare the products with others existing in the market. > > SeshuI just checked their pricing: Multitech: MVP2102-Port VOIP Gateway $899.00 MVP4104-Port VOIP Gateway $1499.00 All I can say WOW!!! (speechless) In comparison: ATCOM: Ag-268 $66.00 2x FXS Ag-468 $88.00 4x FXS Sipura units 2xFXS about 100 +/- whatever -- #Joseph
On Thu, 5 May 2005 08:07:14 -0400, Chris Mason (Lists) wrote:>Is the Sipura 3000 the only way to interface a remote pstn line and connect >incoming calls to Asterisk? I have a location connected by network that has >a phone line, when the room is occupied I want the line ti ring there as >normal, but when the employee is travelling I want the line to be conencted >to a ATA that then feeds it as an incoming pstn line to the pbx located at >my office so it can follow her. >It sounds like the Sipura 3000 would be perfect, what else would do it? > >Chris MasonJust go to and use a fxs-fxo adapter with an ata unit :-) Gary .