Been browsing the forums for setting up my asterisk box. I have extensions setup and have been able to connect to Voip providers as a the trunk. I want to implement this at work company wide, but my telephony knowledge is noobish to say the least. We are putting in a satellite office in another building in our complex and as a test for a full rollout, I wanted to connect the 8 people who are linked with 802.11 G to my network with IP Phones to an Asterisk server then back to the PBX (Panasonic D1232). I have seen the posts and it appears that this can be done. None of the posts or the wiki have any detailed howto info however. I am looking for config and hardware info. For instance, I need to plug the Asterisk box into an extension so that it can dial/be called. Do I plug this just into any preassigned phone extension or does it have to directly into a jack on the PBX? Also what hardware would be needed on the asterisk box, if not a analog modem (which is in there now), I assume I need a digium card of some sort.
Any info would be greatly appreciated. If I can make this work successfully then I will roll it out corp wide and replace the PBX with Asterisk completely.
Are your other company phones like normal home phones (e.g. analog 2-wire signal), or are they proprietary digital phones?
If they are just “POTS” phones (plain old telephone service), then you need a hardware VoIP Gateway with an “FXS” interface… this type of device expects your Panasonic to provide battery and ringing tone on the loop. You would set up the VoIP GW to translate any inbound calls to a specific extension hanging off of your asterisk system. Telecom folks sometimes call this “PLAR” or “private line auto-ringdown”.
We have a a mix of phones. Some are just 2 wire and some are Panasonic digital phones. This VoIP gateway you speak of…I am assuming that this is a seperate piece of hardware that plugs into an extension of the Panasonic PBX and then into the Asterisk box?
You need something like the Sipura 3000 or the Digium TDM400 card. Basically, you need some type of interface which translates the analog extension from the Panasonic (and it has to be an analog extension, by the way … not a digital or proprietary one) into something the Asterisk box can use.
The benefit of the Sipura 3000 is that you can plug in the extension as an FXO and then plug the Sipura into your network at the main location. It will be configured to seek out the Asterisk box’s IP address through the Internet (or VPN or whatever type of network you have in place at the satellite office) and then the Asterisk box will be configured to use the FXO from the Sipura as it’s dialtone. If you use a Digium card, you would have to leave the Asterisk box at your main office. While this is more secure and makes it easier to make changes to it, it also means that your client VoIP phones at the satellite office will have a little more latency connecting to it.
Now, these are not the only two types of interfaces available. Check around on the forums to see if there are any better interfaces available. Whatever you decide to use will need to be an FXO device, not an FXS device. Just keep that in mind.
I would definitely contact the company who installed the Panasonic originally to find out if it has any available analog ports on it or if the ports are dual configurable, in which case it doesn’t matter if you plug a “house phone” or a Panasonic phone into it, the port will detect the type of phone and switch to the type of port it needs to be. Some PBXs have this feature… not sure about Panasonic’s stuff though.
thanks for the info! I will pick up a sipura 3000 and mess around with that, since they are not that expensive. The “satellite” office will actually be on the same network segment as they are about 100yds away and will be connected with a 802.11G bridge.
If you do it right, you won’t even need an Asterisk box.
All you would have to do is plug the 2 wire analog line into a sipura adapter, and have your VOIP phones register with those.
Like this:
Panasonic analog line <-> Sipura adapter <-> VOIP phone set
In this way, you can extend the analog line to the new location across your network. It would retain most of your PBX features, but you would not get certain things like message waiting.
Alternatively, (since they’re so close) run some wire. It’s not that expensive, and provides better service than a wireless bridge.
All 8 phones can register on one sipura adapter? I though about running line (cat5 and rj11) but this is in another unconnected building and I would have to dig a trench and bury the cable (not a fun activity in Alabama in June).
Also if I do setup the sipura adapter as FXO to asterisk, how many concurrent out/in calls can I get on that line? And by extension, you mean any analog extension (ie a standard extension like a phone jack where I plug in my desktop analog phone) or an extension directly connected to one of the ports of the PBX (there are like 7 of these all full from the punch down rj-11 on the wall) Please excuse my inexperience here…
I have found a linksys SPA3000. Can this be used or is it only set to work with vonage?
Each sipura adapter is capable of one concurrent call, and no, all of your VOIP phones would not register with the same single adapter.
You would install a Sipura adapter for each voip phone you wished to install. The pair would work together to extend a single analog line across your network. In this way, you would be able to give each VOIP phone it’s own extension. They would function in a way similar to any other analog phone you have on your PBX.
It would work only with analog lines. (Like the type you would use for a fax or modem.) It would probably not work with ports on the PBX that are designated for Panasonic branded phones.
When you buy a Sipura 3000 that’s not Vonage branded, you can pretty much do whatever you want with it. It can be the device that adapts analog lines to your VOIP PBX, (like a trunk) it can adapt an analog phone to to a VOIP PBX (like a station) or it can be the device that a VOIP phone can use to complete it’s calls (like a PBX with only one station and one trunk). I’ve seen it used all three ways.
I guess I’m just trying to impress upon you, is that you really don’t have to build an entire VOIP PBX (duplicating what you already have) if all you wish to do is extend dial tone from your current phone system.
You can do it very simply with a Sipura adapter paired with any VOIP phone. If you have 8 analog ports you can extend, you just buy 8 sipura adapters, and 8 grandstream phones (or something similar) and you’re in business. Everyone on the same PBX, everyone using the same voicemail, etc…
Thanks for all the info. I am not sure if I explained the problem clearly in my first post:
I have a digital/analog hybrid Panasonic PBX that handles all of my calls. The new office has nothing yet. I am not running Voip anywhere, but what I wanted to do was be able to bring phone service to the new building without buying a seperate PBX for them. If they have their own PBX, they will need their own copper connection to the PSTN as well as we will not be able to transfer calls from the main call center to any of their extensions.
What I envisioned was having Asterisk setup with a FXO trunk on an internal (to the main office) extension on the PBX. If we needed to call them or transfer a call, we would dial an extension…say 101, and get a call center message from Asterisk prompting us to dial their extension on the other side. This would keep us from having to dial an outside number, cut costs, as well as not make us look stupid when someone calls from outside and we have to tell them to dial another number. Right now they have no phones over there so I can buy either analog 2 wire or IP phones. Whichever is best for the setup. The users over at the other office need to be able to pick up and dial extensions back on this side as well as be able to have more than one person on the connection at a time.
Then again maybe I am thinking/going about this the wrong way.
I suppose it depends on what you plan to have in that site.
If you just want to extend dial tone, you can do that with simple IP phones and Sipura adapters.
If you want to buy a whole new PBX, and services for that site, then an Asterisk system isn’t a bad idea. It can easily handle 8 users. There are drawbacks, such as not being able to easily dial extension to extension calls between buildings, and not being able to share/forward voicemail messages. But it very certainly can be done.
However, you’re going to have to buy Sipura adapters and IP phones anyway. (to fit up a new pbx with stations, outside services, and an inter-building trunk[s]). Since you have to buy all those anyway, it only makes sense to me to use them to simply extend dial tone from your current PBX. That way everyone is still on the same system.
how would I use the sipura adapter to extend my panasonic pbx over there? My vision of the sip adapter is a box with a rj-11 analog phone port where you plug in the phone and an ethernet adapter that plugs into my network. The adapter then registers itself as an extension via sip to some voip service. I dont see how that will connect to my pbx since it isn’t VoiP, just old school hybrid digital/analog pbx. This leaves me with phones/sip boxes and nothing to connect them to.
I though having a voip pbx connected to my panasonic pbx would allow me to do this.
As I said before, you can have the VOIP phone register with the sipura device. One sipura adapter for each VOIP phone.
The analog line plugged into the sipura adapter would then be the dial tone for the VOIP phone. Sipura adapters have an input for analog lines. That input is the connction to/from your current PBX. The other side of the adapter hooks into your network and your VOIP phone registers there.
You would just be using your network to extend the analog dial tone. Each VOIP phone and sipura adapter pair would be a seperate extension on your current PBX.
You wouldn’t need to build a whole asterisk PBX to register your VOIP phones.
ahhh. ok. I didn’t realize the sipura box had an input for the analog line. I had assumed that was for plugging an analog phone into to register it to a Voip box. I will give this a try. Any recomendations on where to get a sipura box?
Things can also be bought on ebay, but you might be buying a Vonage adapter if you do. From voip-supply.com/ you’re certain to get unlocked, and unbranded equipment. You can then use it anywhere for anything.
Be sure to check your purchase carefully. You can also buy branded equipment there. If anything you buy comes with “1000 free minutes”, or any talk minutes at all for that matter, you could be buying an adapter for a service.
Look for a sipura SPA 3000. It has an FXO port (for an analog line), as well as an FXS port (for a station). You would ignore the FXS port in this configuration.
Ok got the gear. SPA3000 and a DLink dph-140S. They came each with one page user manuals. Can you point me a direction to find out how to set this up as I have been screwing with it all day and cannot make it work…
thanks
OK I am sure about the PBX end they all are a little different
You should be able to drop the asterisk on the network
use a sip or iax ATA plug into the pbx (with older Norstar / Nortel you have RJ11 jacks) program the line as exten.
the ata gives you tip / ring use it as a analog line.
remote office can use ip phones or get a multi port gateway to use standard phone gear.
that way most of office does things the same ole way
All you are trying to do is setup the phone to see the ATA as it’s gateway. Since I don’t know anything about the dph-140S, I can’t tell you how to do that. Every phone is a bit different.