Running Asterisk 1.4.24 and FreePBX 2.6.01 connected to legacy pbx via T1 crossover.
I have integrated a legacy PBX with Asterisk and most everything works very well. What I would like to do now is use Asterisk voicemail with the 30 legacy extensions. What I cannot figure out is how to send calls directly to voicemail. The legacy extensions do not exist on the Asterisk box, but voicemail boxes do exist for them.
Voicemail is currently handled with the legacy system by transferring calls to a VM extension and sending the VM Mailbox ID along with the transfer. I am trying to figure out how to pass that info to Asterisk, but I think my brick wall is:
How do I dial directly to voicemail without being prompted for a mailbox number?
Hmm, I did know that, but I guess I was thinking more along the lines of dialing voicemail and then sending the extension for the vmbox it’s looking for. That way I could create some extensions, the legacy pbx could dial those extensions when a call goes to voicemail and then send the digits of the box it is looking for.
In essence, that is what *1234 does but I need to break it down to a two step process I think. I need to first dial * and then let the legacy pbx send the 4 digit extension. Let me play with that for a bit and come back to this.
You haven’t shared what kind of legacy PBX it is or how you connect to the Asterisk box. I will assume you have a T1 and created a trunk group on the legacy system.
If the legacy system supports “Busy/No Answer Forwarding” then set it to forward to the trunk access code to the Asterisk, then send the * plus the extension. Obviously, you’ll need to create the extensions on the Asterisk box so that voice mail exists for them.
The only glitch I see is notification of message waiting back to the legacy system. If voicemail to email is available, that will handle the notifications.
The legacy system is a Toshiba CIX/CTX670 and as I said in my original post I did use a T1 crossover to connect.
I actually have everything working quite well now, though I haven’t moved it into production just yet.
To put callers to Asterisk VM the Toshiba rings the Asterisk box, which answers and listens for tones and then puts the caller into the designated mailbox (the Toshiba automatically sends these tones). The only thing I am waiting on right now is an x100p card to set the MWI on the phones (probably a couple of them but I’m testing with one). I could not figure out how to dial #63${EXTEN} through the T1 trunk to set the MWI and #64${EXTEN} to unset the MWI, but I believe that it will work just fine with an analog card. I know it would work with a modem (tested), but I coudln’t get Asterisk to just dial a modem. Yes, I know, modems don’t work with Asterisk unless it’s the Intel 537 chipset, but I wasn’t trying to use them as voice cards – I just wanted the modem to dial a string and hangup. when trying to pass this string over the T1 trunk, it would just ring and then eventually fail or return a busy/congestion message. I guess I can’t pass legacy feature codes over the trunk. /shrug
I am not duplicating any legacy extensions on the Asterisk box. I just added the legacy VM boxes in voicemail.conf. I am very happy with how everything has played out! I should have taken on this project years ago.
I am considering eventually changing it from
PSTN --> Toshiba --> Asterisk
to
PSTN --> Asterisk --> Toshiba
so that I can have trunk side recording for all calls. Our call recording with the legacy PBX is done by plugging the phones into the microphone jack on the PC. It works well, but there are obvious flaws in that technique.
Thanks for the response – I’m sure I’ll be back with more questions!
I am trying to integrade FreePBX with our CTX670 but using the T1 crossover but since I am very new at this I have no clue where to start. Is there a step by step tutorial where I can learn how to do this?
What hardware do I need? Thank You
Hey Mark, I realize I’m going way back in time here and I hope you are still around. It looks like I am trying to do what you have done here. I know little about PBX systems so I am ramping up my education pretty quick. We currently have a Toshiba CTX670 and we have a DOS Key Voice system as our Auto-attendant and Voicemail server. I’m trying to do two phases of a project. Phase one would be to swap out the DOS box with a FreePBX server to handle our Voicemail and Auto-attendant while still working through the Toshiba system for the phone extensions. Phase two would be to add/switch to SIP phone extension as needed. While the current Voicemail box is connected via analog and digital cards, there is also a PRI Crossover Interface attaching the CTX to our network. Let me know if you think you could help me from here. Thanks.
I’ll be happy to help you where I can though I will point out that I am not at all familiar with the DOS Key Voice system.
I have been using FreePBX for a few years now with the 670 and I am actually finally deploying Digium phones this week and will be able to completely remove the 670 as soon as this deployment is completed.
Your phases look to be the exact same as mine. You will need a 2 port PRI card from Digium and I assume you have 8 voicemail ports on the 670 so you will need an 8 port analog card from Digium as well. That is the first step.
You will then need a T1 crossover cable to put between your FreePBX box and the 670.
Our current setup has our Toshiba system running out through an Adtran Total Access 924e. Do you know if that might be a connection point already for the FreePBX server? I just don’t want to over-complicate the solution if there is an available option.
Looking at that Adtran box here http://www.adtran.com/web/page/portal/Adtran/product/4242924L1/107 I’m thinking you may be able to build a SIP trunk for trunking the two PBX together. What is the overall goal? Will the end result include the Adtran? (maybe necessary?)
I am not familiar enough with that Adtran and haven’t done anything with them in years and I never was an Adtran expert anyway.
I didn’t have anything fancy like that and our setup only included the single CTX670, Amanda@Work Voicemail, and a bunch of phones. I used a PRI Crossover cable to trunk between the two. Telco --> FreePBX --> Toshiba.
The Adtran is really how our ISP combines our phone service with our data bandwidth. So it needs to stay. I’m not 100% certain but I also think that our Toshiba system is a CIX670, not CTX670. At least that is what it says in our eManager software. Not sure if that matters.
You’ll have to poke around in eManager then and see if you can create a SIP trunk. There is most likely licensing involved in that, though. Otherwise we are back to getting something like a Digium TE220 dual-Span T1/PRI card and a crossover cable. If you want to use Asterisk for voicemail, you’ll also need an 8 port card like the AEX800. These are likely older model numbers that I’m giving you – it’s what I used.
Take a look at what kind of wiring you have for voicemail, too. I have 8 “ports” but they were actually only 4 2-channel ports. I built/modified my own splitters to make 8 separate lines to connect to the AEX800.
Once we are off the Toshiba, I’m replacing the FXS ports on the AEX800 to FXO and that’s where our fax server will connect. We use a particular fax software that integrates with our line of business software so I don’t use Digium Fax or whatever it is called.