Re: [asterisk-users] high capacity analog - sip gateway
Hi Carlos, To solve the echo problem from your 96 analog ports, you can use the PBXMate. Valer. From: Carlos Alvarez car...@televolve.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 25, 2012 10:57 PM Subject: Re: [asterisk-users] high capacity analog - sip gateway On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I’m looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I’d like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I’ve been able to find with a 48 port capacity are these two: I have a deployment of 96 analog ports using a Digium T1 card ($500 on eBay) and Rhino analog channel banks (also cheap on eBay). We have extremely high reliability from this configuration. In fact, other than the normal analog annoyances like occasional echo, they are rock solid. Are you doing this instead of VoIP phones for cost reasons? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] high capacity analog - sip gateway
I'm looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 10/25/2012 04:21 PM, Justin Killen wrote: just talking in general terms here I have found this sort of hardware is not the most reliable, and the more physical devices you spread it across the more fault tolerant you are of a single fault taking down a big chunk of your users. I wouldn't go more than a 24port device and for 100 users I would get 5 or 6 of them depending on the exact numbers and have one as a hot spare that can just be swapped in quickly if one of the others dies. my analog stuff is all on spa or pap2t right now and I find that working out better for me than T1 card and channel bank was in the past, but the cabling is not as neat and tidy. Its a lot easier pill to swallow when 2 extensions die than 24 for me. I'm looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
Agree with 24 port being the max for a single device. In that vein I just deployed a handful of Grandstream 24 port FXS devices that seem to be working well at a decent price point. I don't normally recommend Grandstream for anything, and in the past we have only deployed Audiocodes for this kind of task, but the Grandstream was half the price... j On 10/25/2012 03:29 PM, jon pounder wrote: On 10/25/2012 04:21 PM, Justin Killen wrote: just talking in general terms here I have found this sort of hardware is not the most reliable, and the more physical devices you spread it across the more fault tolerant you are of a single fault taking down a big chunk of your users. I wouldn't go more than a 24port device and for 100 users I would get 5 or 6 of them depending on the exact numbers and have one as a hot spare that can just be swapped in quickly if one of the others dies. my analog stuff is all on spa or pap2t right now and I find that working out better for me than T1 card and channel bank was in the past, but the cabling is not as neat and tidy. Its a lot easier pill to swallow when 2 extensions die than 24 for me. I'm looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
What would be the advantage of using 100 single units vs. just buying VoIP phones? That doesn't seem very cost effective to me in the long run. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen jkil...@allamericanasphalt.com wrote: I’m looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I’d like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I’ve been able to find with a 48 port capacity are these two: I have a deployment of 96 analog ports using a Digium T1 card ($500 on eBay) and Rhino analog channel banks (also cheap on eBay). We have extremely high reliability from this configuration. In fact, other than the normal analog annoyances like occasional echo, they are rock solid. Are you doing this instead of VoIP phones for cost reasons? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
Cost and ease of deployment, yes. At this specifc location we are currently using Centrex lines (ATT hosted) and are looking for a way to move into something cheaper without throwing away the existing phones. I like the idea of using a channel bank - I'll look into that as an option as well. -Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Thursday, October 25, 2012 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] high capacity analog - sip gateway On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen jkil...@allamericanasphalt.commailto:jkil...@allamericanasphalt.com wrote: I'm looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: I have a deployment of 96 analog ports using a Digium T1 card ($500 on eBay) and Rhino analog channel banks (also cheap on eBay). We have extremely high reliability from this configuration. In fact, other than the normal analog annoyances like occasional echo, they are rock solid. Are you doing this instead of VoIP phones for cost reasons? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
That is just silly. You mean to say that the Adtran and the Adit units are not as reliable as these new devices. No way. Get Adtrans, buy a four port T1 card or even better get the redfone device and do HA Linux between to boxes, you have immediate failover. http://www.red-fone.com/products-new/80.html I seriously doubt any product on the market is as solid, tried, and true as the traditional channel bank. You can pickup these channel banks very cheap used, and often find them in telco closets that have been abandoned. Thanks, Steve Totaro On Thu, Oct 25, 2012 at 4:29 PM, jon pounder j...@inline.net wrote: On 10/25/2012 04:21 PM, Justin Killen wrote: just talking in general terms here I have found this sort of hardware is not the most reliable, and the more physical devices you spread it across the more fault tolerant you are of a single fault taking down a big chunk of your users. I wouldn't go more than a 24port device and for 100 users I would get 5 or 6 of them depending on the exact numbers and have one as a hot spare that can just be swapped in quickly if one of the others dies. my analog stuff is all on spa or pap2t right now and I find that working out better for me than T1 card and channel bank was in the past, but the cabling is not as neat and tidy. Its a lot easier pill to swallow when 2 extensions die than 24 for me. I’m looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I’d like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I’ve been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 25/10/12 9:49 pm, Justin Killen wrote: What would be the advantage of using 100 single units vs. just buying VoIP phones? That doesn't seem very cost effective to me in the long run. In older buildings with existing single pair cabling, there might not be a great deal of choice. We were faced with a similar scenario at a hotel in Lincolnshire a few months ago - listed building, lots of old pre-ethernet cable, no likelihood of being able to replace the cable. In answer to the OP, I concur with suggestions for 24-port channel banks - you really don't want one or two devices responsible for all 100 extensions. I would not encourage individual SPA or PAP units - it'd be an administative (and cabling) nightmare - it's bad enough with a dozen of the things. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 5:01 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: On 25/10/12 9:49 pm, Justin Killen wrote: What would be the advantage of using 100 single units vs. just buying VoIP phones? That doesn't seem very cost effective to me in the long run. In older buildings with existing single pair cabling, there might not be a great deal of choice. We were faced with a similar scenario at a hotel in Lincolnshire a few months ago - listed building, lots of old pre-ethernet cable, no likelihood of being able to replace the cable. In answer to the OP, I concur with suggestions for 24-port channel banks - you really don't want one or two devices responsible for all 100 extensions. I would not encourage individual SPA or PAP units - it'd be an administative (and cabling) nightmare - it's bad enough with a dozen of the things. Kind regards, Chris -- This email is made from 100% recycled electrons Wow, single pair? never came across that before. You can run 10bastT on CAT3. The worst I have found where Amphenol cables, one per station. Not really bad since it is a 25 pair cable. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen jkil...@allamericanasphalt.com wrote: ** Cost and ease of deployment, yes. At this specifc location we are currently using Centrex lines (ATT hosted) and are looking for a way to move into something cheaper without throwing away the existing phones. I like the idea of using a channel bank – I’ll look into that as an option as well. You should be able to also connect the Centrex lines to the channel banks, I believe. I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote: I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. Out of curiosity, would you mind sharing that with us? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 2:14 PM, Christopher Harrington ch...@acsdi.comwrote: On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote: I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. Out of curiosity, would you mind sharing that with us? The phone? Grandstream. They have people who love them and hate them, but so far we're pretty happy with them. A GXP2124 is on my desk right now, and I have access to any phone I want. I like it. There are low-end models still with a display for under $60. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
yealink T18 and T20 are decent phones available for $60 Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Fri, Oct 26, 2012 at 2:44 AM, Christopher Harrington ch...@acsdi.comwrote: On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote: I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. Out of curiosity, would you mind sharing that with us? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
I think if we were to go to VoIP phones, one thing that we would have to consider very highly in a phone would be that they have VLAN settings and a built-in Ethernet hub/switch so that we can just inject it into the user's computer LAN connection. The cost and time of rewiring some of these locations is not something we're comfortable with doing. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Thursday, October 25, 2012 2:28 PM To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] high capacity analog - sip gateway yealink T18 and T20 are decent phones available for $60 Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.inmailto:mi...@enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Fri, Oct 26, 2012 at 2:44 AM, Christopher Harrington ch...@acsdi.commailto:ch...@acsdi.com wrote: On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.commailto:car...@televolve.com wrote: I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. Out of curiosity, would you mind sharing that with us? -- -Chris Harrington ACSDi Office: 763.559.5800tel:763.559.5800 Mobile Phone: 612.326.4248tel:612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 2:34 PM, Justin Killen jkil...@allamericanasphalt.com wrote: ** ** ** I think if we were to go to VoIP phones, one thing that we would have to consider very highly in a phone would be that they have VLAN settings and a built-in Ethernet hub/switch so that we can just inject it into the user’s computer LAN connection. The cost and time of rewiring some of these locations is not something we’re comfortable with doing. The Grandstreams do that. Most phones do. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 10/25/2012 05:09 PM, Carlos Alvarez wrote: On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen jkil...@allamericanasphalt.com mailto:jkil...@allamericanasphalt.com wrote: Cost and ease of deployment, yes. At this specifc location we are currently using Centrex lines (ATT hosted) and are looking for a way to move into something cheaper without throwing away the existing phones. I like the idea of using a channel bank -- I'll look into that as an option as well. You should be able to also connect the Centrex lines to the channel banks, I believe. Best to check the specs of the actual phones, around here some of them are norstar phones that I am pretty sure are some sort of isdn (bri) thing rather than being a pure analog device. Better still take one of them and plug it in a raw analog line someplace and see what you get. I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 10/25/2012 05:01 PM, Steve Totaro wrote: That is just silly. You mean to say that the Adtran and the Adit units are not as reliable as these new devices. No way. I have had channel banks fail yes, and I stick by my assertion that failing a small $50 box is a lot less painful on the wallet and users than a channel bank with most of the extensions on it, this changes as the scale goes up though. I would only use a channel bank where the size can justify at least 3 of them, and I would never use a T1 based one again I would use the ethernet to FXS ones. I use a combination of analog and voip phones and there are various reasons for each being the type it is, one solution doesn't always fit everything, even within a single system. Get Adtrans, buy a four port T1 card or even better get the redfone device and do HA Linux between to boxes, you have immediate failover. http://www.red-fone.com/products-new/80.html I seriously doubt any product on the market is as solid, tried, and true as the traditional channel bank. You can pickup these channel banks very cheap used, and often find them in telco closets that have been abandoned. Thanks, Steve Totaro On Thu, Oct 25, 2012 at 4:29 PM, jon pounder j...@inline.net wrote: On 10/25/2012 04:21 PM, Justin Killen wrote: just talking in general terms here I have found this sort of hardware is not the most reliable, and the more physical devices you spread it across the more fault tolerant you are of a single fault taking down a big chunk of your users. I wouldn't go more than a 24port device and for 100 users I would get 5 or 6 of them depending on the exact numbers and have one as a hot spare that can just be swapped in quickly if one of the others dies. my analog stuff is all on spa or pap2t right now and I find that working out better for me than T1 card and channel bank was in the past, but the cabling is not as neat and tidy. Its a lot easier pill to swallow when 2 extensions die than 24 for me. I’m looking for an fxs - sip gateway/router/switch for about 100 existing analog phones. I’d like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I’ve been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 5:35 PM, jon pounder j...@inline.net wrote: On 10/25/2012 05:09 PM, Carlos Alvarez wrote: On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen jkil...@allamericanasphalt.com wrote: Cost and ease of deployment, yes. At this specifc location we are currently using Centrex lines (ATT hosted) and are looking for a way to move into something cheaper without throwing away the existing phones. I like the idea of using a channel bank – I’ll look into that as an option as well. You should be able to also connect the Centrex lines to the channel banks, I believe. Best to check the specs of the actual phones, around here some of them are norstar phones that I am pretty sure are some sort of isdn (bri) thing rather than being a pure analog device. Better still take one of them and plug it in a raw analog line someplace and see what you get. I always advocate throwing out old analog phones as they will be a pain, but understand if you absolutely cannot. Just keep in mind you can get a decent VoIP phone for $60 that is very likely to be nicer than what they have now and do much more. -- Carlos Alvarez TelEvolve 602-889-3003 Very true. If they have lots of lights and a display, they are most likely digital phones. What kind of PBX and phones do you have. Before digital, phones needed 25 pair to control the phone's various lights, lines, mwi. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 26/10/2012, at 10:09 AM, jon pounder j...@inline.net wrote: On 10/25/2012 05:01 PM, Steve Totaro wrote: That is just silly. You mean to say that the Adtran and the Adit units are not as reliable as these new devices. No way. I have had channel banks fail yes, and I stick by my assertion that failing a small $50 box is a lot less painful on the wallet and users than a channel bank with most of the extensions on it, this changes as the scale goes up though. I would only use a channel bank where the size can justify at least 3 of them, and I would never use a T1 based one again I would use the ethernet to FXS ones. I use a combination of analog and voip phones and there are various reasons for each being the type it is, one solution doesn't always fit everything, even within a single system. The Xorcom product range are pretty handy for this. You can plug in to your box via USB and they look like Dahdi extensions, so you can see state change etc. http://www.xorcom.com/telephony-interfaces/telephony-interfaces.html I had some issues with the latest kernel, but other than that they are great in the hotels They are externally powered and have modules within the chassis so you can get a large chassis then add lines We are using a 32 port in one hotel happily. I find the SPAs horrible to debug if there is an issue to do with actual call. They do work, but if something is not quite right with a call its really hard to see whats happening. I prefer to be able to see everything on the CLI Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On Thu, Oct 25, 2012 at 5:09 PM, jon pounder j...@inline.net wrote: On 10/25/2012 05:01 PM, Steve Totaro wrote: That is just silly. You mean to say that the Adtran and the Adit units are not as reliable as these new devices. No way. I have had channel banks fail yes, and I stick by my assertion that failing a small $50 box is a lot less painful on the wallet and users than a channel bank with most of the extensions on it, this changes as the scale goes up though. I would only use a channel bank where the size can justify at least 3 of them, and I would never use a T1 based one again I would use the ethernet to FXS ones. I use a combination of analog and voip phones and there are various reasons for each being the type it is, one solution doesn't always fit everything, even within a single system. Of course one solution doesn't fit everything but most likely would in the case at bar (providing it is truly an analog system). T1 cards are dirt cheap and channel banks are too. They are also modular, so if a card goes out, you lose 4 extensions. If the chassis goes out, then you lose all but it is all solid state. I mean this stuff is the mainstay of telephony. A SIP FXS box will obviously have a substantial lower mean time between failures. Look it up. What brand did you see fail and what failed? I have only had cards blow out and that is EXTREMELY rare. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] high capacity analog - sip gateway
On 10/25/2012 01:21 PM, Justin Killen wrote: I'm looking for an fxs- sip gateway/router/switch for about 100 existing analog phones. I'd like to get this done cheaply, but I want to make sure that whatever we buy works well with asterisk as well. As far as I can tell, digium make no such device. The only ones I've been able to find with a 48 port capacity are these two: Sangoma Vega 5000 50 FXS + 2 FXO Gateway (http://www.voipsupply.com/sangoma-vega-5000-50fxo-2fxs) Realtone WSS120 VoIP Gateway (http://www.realtonetech.com/product/voip-gateways/86-wss120-series-voip-gateways.html#description) Does anyone have any experience with either of these products/vendors, or any suggestions for a different piece of hardware? Thanks -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How about this for a setup: 4 port T1 cards (1) Digium TE405P (PCI)~$600 (used) or (1) Digium TE420 (PCI-e 1x)~$1300 (used) and then (4) Adtran Total Access 624 (TA624)~$75 (used) 24 port channel bank We use the TA624's CPE all the time. They are very hard to kill. -- Jim Lucas http://www.cmsws.com/ http://www.cmsws.com/examples/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
