Hello everyone! This is my first time posting to the forums and I thank you all in advance for any help you can provide in regards to this.
My main question is in regards to survivability of an Asterisk system. Some background on the project is probably in order.
My employer currently has two offices located about 1 mile apart. They are connected with a 100mbps fiber link. At each office, there are approximately 40 users and 4 PSTN lines (so total 80 users + 8 PSTN lines). Currently, they have a Toshiba Strata CTX670 at one office and an ancient Comdial PBX at the other. The systems do not speak any VoIP protocols (although the Toshiba will support Toshiba VoIP phones with a costly upgrade and licensing fee). Additionally, the systems are not integrated at all so to dial between offices you need to dial the PSTN number of the other office and come in just like any other caller.
I am actually the IT manager so this is my first foray into the wonderful world of telecom. I’ve been working with our resident telecom administrator to come up with a good plan on getting an integrated system in place. After looking at several alternatives and options, we happened upon Asterisk and will probably end up at least doing a proof of concept system to test if this design will work.
Basically, what we are thinking is buying 80 Grandstream GXP-2000 phones (40 for each office), 2 FXO-to-VoIP gateways, and 2 Dell Poweredge boxes. We’ll have 1 gateway and 1 Poweredge at each office. What we’d like to accomplish is that if we lost any single component, the system will still work. Since there is high-speed connectivity between the offices, it should be possible.
My idea was having the GXP-2000 phones all register to both Asterisk servers as well as having the FXO/VoIP gateways register to both servers. As long as we can get Asterisk out of the media-path by keeping the audio codecs the same, the actual voice data should be going directly from the gateway to the end-user’s handset. Obviously, if we lose a gateway, then all calls going through that gateway would be cut off and if we lose a server, all calls currently being signaled by that server would also be cut off. But the end-user should be able to hang up and pick the other “Line” (actually the other Asterisk server) on the GXP-2000 and use that to dial out.
Has anyone been able to get something similiar to the above working in a production environment or know if it is possible with Asterisk? Any suggestions on the VoIP gateways we should look at for this kind of functionality (the D-Link ones look alright but it seems they’re vaporware at the moment)? I’ve scoured the wiki looking for information on load-balancing and fail-over but none of the things I was able to find address a distributed fail-over system like I’ve described above.
The only other question I have to pose to some of the guru’s here is with regards to voicemail. In any fail-over scenario, how do you handle voicemail during that time? Obviously, you can have the configuration for the voicemail in a database but then you need a couple more servers for redundancy of your database (unless you throw MySQL on the Asterisk box). But how do you handle the actual stored voicemails? I’m thinking some kind of sync process that shuffles VM files between the two so if you lose a server, the user can still access voicemail and incoming callers can still leave the person voicemail.
Thanks in advance for any help, advice, insights you can give on the above questions. If you need more info, I’ve got this thread tagged to notify me so I’ll post back right away whatever you ask for.