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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121025/f7bb698c/attachment.htm>
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 spaxxxx 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121025/bf2cd4fe/attachment.htm>
Carlos Alvarez
2012-Oct-25 20:57 UTC
[asterisk-users] high capacity analog <-> sip gateway
On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen < jkillen at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121025/5cae5c8b/attachment.htm>
Christopher Harrington
2012-Oct-25 21:14 UTC
[asterisk-users] high capacity analog <-> sip gateway
On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez <carlos at 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: 763.559.5800 Mobile Phone: 612.326.4248 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121025/a9f0df7c/attachment.htm>
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: mitul at enterux.in DID: +91-22-71967121 Cell: +91-9820332422 On Fri, Oct 26, 2012 at 2:44 AM, Christopher Harrington <chris at acsdi.com>wrote:> On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez <carlos at 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: 763.559.5800 > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121026/b52e10ce/attachment.htm>
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-usersHow 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/
Hi Carlos, To solve the echo problem from your 96 analog ports, you can use the PBXMate. Valer. ________________________________ From: Carlos Alvarez <carlos at televolve.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at 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 <jkillen at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121026/d3bbc47c/attachment-0001.htm>
I use a lot of FXS SIP gateways for analog phones. Usually it's because there's no cat5 on site and there's no budget to re-cable. Other times it's because the customer only needs $10 trimline phones, or has residents provide their own phones. I use Mediatrix FXS gateways and highly recommend them. If you choose to use Grandstream, be prepared for headaches. This is not uninformed brand-bashing. I have deployed hundreds of Grandstream gateways in 4, 8, and 24 port models, and they rarely last more than 2 years. They generate so many support calls for us, I no longer use them. Problems include: losing registration, locking up, failing to play dial tone, one way audio, strange crashes that pollute syslog. Rebooting or hard power cycle usually fixes problems for a while, but I haven't found a good way to script a regular preventative reboot (yuck). External power bricks seem to fail a lot, and autopsies on failed gateways show a lot of bad capacitors. Since switching to MTX a year or two ago, I have not had a single support call about the units. I have several dozen units in production now. I've used AudioCodes gateways as well and they're great, but usually out of our budget (and more difficult to configure). MTX pricing is somewhere between Grandstream and AudioCodes. I've found that the extra money we spend on better hardware is less than what it would cost to hire another person to take support calls and ship replacement hardware.