FOOD FOR THOUGHT 3CX and Analog Hardware Integration

[dicko]

I will also note that for those that have relied on Digium (like) hardware, read DAHDI, NONE are supported by 3CX although FreePBX, and FreeSwitch, do and have for 15 years, If you need that then you are basically hosed as yet with 3CX, please think about that before you put your hands in a closed source commercial solution, I would suggest that 3CX needs to immediately attest to that if they want to play in this market, I assume they have "c" programmers on-board, they have no excuse if they join the Linux/voip life style otherwise, dahdi can be compiled on almost anything but windoze, otherwise you have only a SIP solution, no PRI's nor channel banks nor legacy PBX integration, no FXO's nor FXS's, think door phones and Polycom's, as yet it is not close to a competitive replacement . . .

 

[Nick Galea]

I will also note that for those that have relied on Digium (like) hardware, read DAHDI, NONE are supported by 3CX although FreePBX, and FreeSwitch, do and have for 15 years, If you need that then you are basically hosed as yet with 3CX, please think about that before you put your hands in a closed source commercial solution, I would suggest that 3CX needs to immediately attest to that if they want to play in this market, I assume they have "c" programmers on-board, they have no excuse if they join the Linux/voip life style otherwise, dahdi can be compiled on almost anything but windoze, otherwise you have only a SIP solution, no PRI's nor channel banks nor legacy PBX integration, no FXO's nor FXS's, think door phones and Polycom's, as yet it is not close to a competitive replacement . . .

3CX supports PSTN, door phones via VoIP gateways and SIP. Unfortunately 3CX does not support Digium hardware directly. Thats because Digium hardware has been developed for Asterisk (makes sense, they wrote Asterisk). Our architecture is to work with VoIP Gateways, so its not easy to support Digium hardware (we would if it was easy, we don't make hardware anyway). But the way things are going, and evidenced by user polls, most companies are migrating to SIP trunks. That said you could easily keep those cards and machines with asterisk acting as a Proxy / VoIP Gateway to 3CX

We do support Polycom phones, with extended support for VVX models coming soon (depends also on Polycom)

 

[dicko]

Sorry to be blunt, but you miss the point, before SIP was Zaptel hardware, that is what Asterisk was originally built on, it surely is an "open standard" in VOIP, Global SIP trunking came a little later, as did 3CX

I agree it is not "easy" to encompass the legacy state but both the digium source and the asterisk chan_dahdi source are both a download away, reverse engineer away ;-) , it would I agree take money, talent and commitment, but ignoring it as a reality seems a little evasive in an open source (open standard , whatever that means) environment, do you not agree?

I don't see a reasonable way for a Freepbx pbx to "proxy" for a 3CX instance, please elucidate as to how.

 

[wardmundy]

Haven't personally tried it, but...

Another option might be something like the OBi508vs with 8 FXS ports for $320. You can add 3 FXO ports as well using $40 ObiLine adapters.

 

[jerrm]

Speaking as one skeptical of the PIAF/3CX situation, I don't see any issue here. If you need dahdi, move on, 3CX is not the answer. It's not like they are misrepresenting anything.

Dahdi is a design choice. Very few if any situations actually require a Dahdi solution (but it may be the better/easier option in many cases).

That said, folks who would like to use 3CX but can't/won't should make the reasons known. Squeaky wheels work.

Asterisk history and whether Dahdi support is easy or hard is irrelevant if 3CX believes resources are best used elsewhere. They've been around long enough in the Windows world to have pretty good idea of where the issues are. Entering the Linux world may change their outlook, but they have to get there first.

But the Linux version is likely a cloud/virtualization play, making Dahdi even less relevant.

 

[SkykingOH]

I would have to check but I think the original Digium Tiger Jet cards existed before Asterisk and Mark wrote Asterisk and DAHDI project around the design.

With regards to Sangoma Wanpipe it predated Asterisk.

Sangoma's Vegastream gateway line works great with 3CX.

 

[VictorFlash]

I like the 3CX starter idea. In reference to not supporting DAHDI cards, I have a client that is going to change phone/internet providers. Currently they have a Switchvox system with a card for the fax and a T1/PRI card for voice and Polycom phones. I would like to consider switching to 3CX or Wazo with SIP lines. (I am installing both in VMs for my on system to test.) Looking for opinions on leaving PRI to SIP. Are SIP lines solid for businesses. I use a Vitelity SIP for my business and it has worked great for me. Just interested on opinions from those with more experience than me in the phone world. Thanks for your feedback.

 

[dicko]

PRI's from a legitiamete PSTN provider are 5x9 guarenteed (thats about less than 3 minutes a year downtime) , SIP will never be close

 

[atsak]

The PRI guarantee doesn't really change whether it goes down or not though - just that you get a partial service credit when it does. The one I manage has a 4 hour MTTR and went down for a DAY a couple years ago. It has been quite reliable otherwise. I've used a <ducks> Sangoma Vega PRI gateway for several years with it no issue; and have used a Sangoma FXO Vega as well. For FXO and FXS these days I've used the Grandstream's. While most of their stuff is frankly not that great, it's been reliable in a few installations though it did need to be tuned to get the noise out on the FXO interfaces. What was great about this is that I could also setup a backup PBX without having to switch the physical lines. It doesn't control for failures on the gateway, but in my experience thus far those happen less frequently than hardware failures on PBX servers.

My SIP carrier has had a similar outage profile (Voip Innovations) - hadn't had an outage for a couple years then they ran into a big DoS issue earlier this year with spotty service for a few days.

I therefore disagree that DAHDI is a must have. Gateways work equally as well in my experience, Sangoma and Grandstream have both been successful for me so fair, and I'm sure the OBi one is pretty good too that Ward recommended.

 

[VictorFlash]

Thanks dicko and atsak. To expand on my question some. From what I have seen, that PRI interface is no longer coming in on a copper T1. That PRI is coming in to the office on the same cable/fiber as my data. So is there an advantage to translating the data connection to a PRI connection that goes into my PRI card? Where as my SIP connection is tied directly into my data connection without any translation. And assuming I am using the SIP services of my data carrier they would be responsible for my SIP connections.

Atsak, same interruption happened at a clients a few months ago. Construction crew, took out out fiber connection which had Internet and phone, and took out our T1 that we for our fax lines. Which reminds me we had to keep the T1 because AT&T was not going to provide us with fax lines on our fiber connection. The service interruption was over a day for the fiber and two days for the PRI/T1.

 

[atsak]

Thanks dicko and atsak. To expand on my question some. From what I have seen, that PRI interface is no longer coming in on a copper T1. That PRI is coming in to the office on the same cable/fiber as my data. So is there an advantage to translating the data connection to a PRI connection that goes into my PRI card? Where as my SIP connection is tied directly into my data connection without any translation. And assuming I am using the SIP services of my data carrier they would be responsible for my SIP connections.

If your PRI is coming in on fiber and being converted, there is actually a disadvantage. Simply stay with straight SIP instead in this instance. In theory this setup should be the best of both worlds - reliability of a dedicated connection and simplicity of a SIP trunk on the server . . .

 

[dicko]

Couple of problems with PRI's over Network.

There is no reliable clock source, you need to provide your own.
They will likely push you into compression (g729) that will cost you unnecessary licenses, even though you bought the bandwidth for uncompressed.
If you have alarm systems , fax machines , credit card machines, modems etc.etc. they will say "keep your FXO's" , we don't support them.
If AT&T give you a cheap router , then they will NOT be able to give you g711 all the time, Cisco's can do that but you need to fight them for faes and other g711 data services.
If you have a blended system with legacy PBX' you will be exposed to a significant cost for g729 licenses (Mitel, Nortel, AT&whatever)
They are generally pretty ignorant of how SIP is supposed to work, no video, no g722 . . .
Most will not allow "foreign" CID's to be announced, if they do, then probably not for "TollFree"

Ironically AT&T require you install a PSTN line for the 9600 baud modem they supply so they can get to the router if "there is a problem"

This is my experience, YMMV and it is JM2CWAE