Re: [asterisk-users] FXO ATA Options?
Conall O'Brien wrote: Hello, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. Does anyone have any experience using the Linksys Sipura 3201 as an FXO device for Asterisk? I use one at home and can recommend it as functional and reliable. It has an unbelievable number of configuration options. Linksys docs are a bit sparse, try the Sipura site under SPA3000. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO ATA Options?
Hi, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. i've had some problems setting the disconnect tone correctly to my country. As a matter of fact, i still do, as the calculated values does not always hang up the phone. Other than this i have a small issue which i did not understand completely. Sometimes the SPA webpage starts to load, and in the middle i get a connection reset page by firefox. Sometimes i can only load the page by refreshing 10-15 times or even more. This only happens in advanced admin mode, when using any other modes everything works fine. This refresh-error only occures from remote networks, not from a PC that is within the same subnet as the SPA (subnets are connected via site2site vpn tunnel). I haven't had time to correctly debug this issue (tcpdump etc) but it's so annoying that i will go and debug this once. It may be an MTU issue, an SPA performance issue, a firefox issue... This is an SPA3k which i'm using (actually not one but four, all involved in this problem). regards Adam ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO ATA Options?
Hello, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. Does anyone have any experience using the Linksys Sipura 3201 as an FXO device for Asterisk? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO ATA Options?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Conall O'Brien wrote: Hello, I'm currently looking at FXO options to provide a POTS line to Asterisk to trunk calls with. Does anyone have any experience using the Linksys Sipura 3201 as an FXO device for Asterisk? I just did this and it works great. Google for SPA3000 to find quick-and-easy instructions for configuring it and Asterisk for use. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD8DBQFHImRvCFu3bIiwtTARAu2aAKCU/Gzy/M+N351uQYvdvfvF3nTOQwCgkuI+ 6FkwsG4jg2RaFnxCYRgoKuY= =HmfR -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO ATA?
Jon Gabrielson wrote: On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the difference between a passthrough type and a regular FXO. What can you do with one that you can't do with the other? I noticed that the wiki says that the handytone 486's lifeline FXO port is not usable via SIP, only used as a fallback for power failure. Is this considered a passthrough or are there 3 types, pass-through, lifeline, and full FXO. I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. ___ 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
RE: [Asterisk-Users] FXO ATA?
Why not go with Multitech? They are expensive, but great units. Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 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. ___ 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 ___ 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
RE: [Asterisk-Users] FXO ATA?
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 ___ 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
Re: [Asterisk-Users] FXO ATA?
On Fri, May 06, 2005 at 04:24:32AM -0500, Eric Wieling aka ManxPower said: Jon Gabrielson wrote: On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. Not quite. A pure life-line FXO that is not voip accessable is useless to *. Usually this means that an extension on the FXS port uses the PSTN on the FXO during powerfailure / 911 calls. Some ATA's have this kind of port. The SPA-3000's FXO CAN pass through in life-line mode automatically for power faliures and if it is configured to do so via the dial-plan. The dial plan on the 3000 allows lots of flexibility here. From a VoIP standpoint, the FXS and FXO ports can be configured to be totally separate devices, where if you want to make a call via the PSTN, the call is looped through *. Pass through can also be used in terms of how the FXO interfaces with *. The standard config of the SPA-3000 for example answers the call and THEN forwards to * - acting more like a full gateway than a dumb FXO. It can also be configured (kludged) to pass through call info to * BEFORE the call is answered (which is frequently more desirable in many situations.) Hope this helps. ___ 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
Re: [Asterisk-Users] FXO ATA?
On Fri, 2005-05-06 at 04:24 -0500, Eric Wieling aka ManxPower wrote: I just checked my dictionary and it defines pass-thru as meaning totally useless for most people. Pass-thru and lifeline seem to be different terms for the same thing. i.e. The FXO port is connected to the FXS port in the event of a power outage, but other than that it is not useful. Well, I didn't check any dictionary :-) but I've checked what other people saying/posting about the AG-168 on their board at: http://en.atcom.com.cn/bbs/ AG-168V Series From their conversation/questions it seems to me that that unit can receive and make call to/from PSTN line. -- #Joseph ___ 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
RE: [Asterisk-Users] FXO ATA?
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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. ___ 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
Re: [Asterisk-Users] FXO ATA?
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. ___ 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
Re: [Asterisk-Users] FXO ATA?
On Fri, May 06, 2005 at 11:49:42AM -0700, Rusty Shackleford said: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Friday, May 06, 2005 3:54 AM 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. Multi-tech has always been this way across their entire product line. They sell enough units to stay in business, but are priced in a way that ensures that they will never be a market leader (in terms of unit sales.) It's too bad, because technically they are awesome. ___ 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
RE: [Asterisk-Users] FXO ATA?
[snip] I totall agree with that comment. Multitech is just a rip-off, when you compare the products with others existing in the market. Seshu I 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 ___ 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
Re: [Asterisk-Users] FXO ATA?
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 Mason Just go to and use a fxs-fxo adapter with an ata unit :-) Gary . ___ 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
[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
RE: [Asterisk-Users] FXO ATA?
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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 ___ 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
Re: [Asterisk-Users] FXO ATA?
On 5/5/05, Chris Mason (Lists) [EMAIL PROTECTED] 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 ___ 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
Re: [Asterisk-Users] FXO ATA?
[snip] 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 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. -- #Joseph ___ 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
Re: [Asterisk-Users] FXO ATA?
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 ___ 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
Re: [Asterisk-Users] FXO ATA?
On Thu, 05 May 2005 12:33:50 -0600, Joseph wrote: [snip] 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 Does anyone here have experience with the new Vantage FXO adapter from Aastra? I see it offered with four 480i phones as a complete small office package providing 4 FXOs, MOH and a lifeline function. Could be a nice freestanding alternative to my TDM card. My SPA-3000 will be listed on Ebay shortly. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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
Re: [Asterisk-Users] FXO ATA?
On Thu, May 05, 2005 at 12:33:50PM -0600, Joseph said: 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. See here on how to pass through with the Sipura: http://voxilla.com/forum-viewtopic-t-1335-sid-c3365f7a694970ed5b7fa0fce2618636.html Yes, I've tested it and it works. ___ 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
Re: [Asterisk-Users] FXO ATA?
Joseph wrote: snip 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. It is possible to pass through an incoming call to Asterisk without the Sipura answering it, although it does take some contortions. Basically, you have the FXO port add a character to the beginning of the CIDNumber (I picked Z). Then, for the FXS port, have it conditionally forward a call to * if it has that character at the beginning of the CIDNumber. Since all calls coming in from the FXO port would have that character, but no other calls would, it effectively makes the call pass through to * without answering it. See: http://www.voip-info.org/wiki-Sipura+3000 It took me a few tries to get the settings right, but in the end it works well. Also, in addressing the post about the Handytone 488... I had one for a week. Either I had a bad one, or the item needs a bit more work to be marketed as something anyone would want to rely on. I ended up returning mine. But it did seem to only pass the call to Asterisk after about 4 rings. - Dan ___ 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
Re: [Asterisk-Users] FXO ATA?
- Original Message - The Grandstream HandyTone 488 has an FXO port. I've never used it though. I could be wrong, but I seem to remember reading up on the HandyTone and deciding that it doesn't really act like a true FXO, as in calls come in and go straight to Asterisk like an FXO, and calls can dial out like a true FXO. I think it operated more like an 'ability to dial in number' and as a pass-through in case of power outage. Someone correct me if I'm wrong on that Paul 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 ___ 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 ___ 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
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. ___ 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
Re: [Asterisk-Users] FXO ATA?
On Thu, 2005-05-05 at 16:51 -0500, Jon Gabrielson wrote: 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. Have a look at the specification sheet: http://en.atcom.com.cn/En_AG-168V.html It has 1-FXS and one 1-Life Line (it is pass through type) -- #Joseph ___ 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
Re: [Asterisk-Users] FXO ATA?
On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the difference between a passthrough type and a regular FXO. What can you do with one that you can't do with the other? I noticed that the wiki says that the handytone 486's lifeline FXO port is not usable via SIP, only used as a fallback for power failure. Is this considered a passthrough or are there 3 types, pass-through, lifeline, and full FXO. Thanks, Jon. ___ 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
RE: [Asterisk-Users] FXO ATA?
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. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Gabrielson Sent: Thursday, May 05, 2005 9:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FXO ATA? On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the difference between a passthrough type and a regular FXO. What can you do with one that you can't do with the other? I noticed that the wiki says that the handytone 486's lifeline FXO port is not usable via SIP, only used as a fallback for power failure. Is this considered a passthrough or are there 3 types, pass-through, lifeline, and full FXO. Thanks, Jon. ___ 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 ___ 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
Re: [Asterisk-Users] FXO ATA?
On Thu, 2005-05-05 at 21:56 -0500, Jon Gabrielson wrote: On Thursday 05 May 2005 05:28 pm, Joseph wrote: It has 1-FXS and one 1-Life Line (it is pass through type) I've seen the pass-through term used alot and I'm not quite for sure what that means. What is the difference between a passthrough type and a regular FXO. What can you do with one that you can't do with the other? I noticed that the wiki says that the handytone 486's lifeline FXO port is not usable via SIP, only used as a fallback for power failure. Is this considered a passthrough or are there 3 types, pass-through, lifeline, and full FXO. Here is what I was able to find on: http://www.grandstream.com/y-faq.htm#gen16 PSTN Pass through port: What it can do: - Local manual switching between PSTN and IP mode on a per call basis. - User can switch to PSTN line by pressing *00 (or the configured strings) for each call before they are placed. The device will revert back to the default IP mode once the phone is hung up. - It can allow a PSTN call to ring/call the phone connected to the FXS port. - It also serves as a life line in case of power outage. What it CANNOT do: - Terminate a VoIP call into the PSTN port - Allow a call from PSTN to route other VoIP devices (different from the FXS phone) over the IP network - Automatically route calls made by the local user to PSTN line Note: On the HT-486 Rev 1.0, the PSTN port is only a life line port that switches to PTSN only on loss of power. FXO port: It can support all the functions of a PSTN pass through plus: - Terminate a VoIP call into the PSTN port - Allow a PSTN call to call either the FXS phone or other VoIP devices over the IP network - Route call automatically and transparently to PSTN line according to user configuration When I get the AG-168V I'll let you all know what it is. -- #Joseph ___ 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
