Forwarding to other PBX systems

Good’ay everyone!

Sorry if this is the wrong part of the forum to post in but I have an odd one for you.

For what ever reason my IT director went with a PBX system which we quickly outgrew, it allows for a maximum device list of 80 devices, and a max number of users of 200.

Previously we had more locations and have since moved one into another, this then exceeds the limit of the PBX, but now we have 2 phone servers with the same limitations. Ive been instructed by him to install asterisk as a kind of middle server, which will take the incoming calls and pass them onto the other servers dependent on the number being called.

everything from 001 - 499 should be passed to server1 and everything from 500 - 999 should be passed to server 2.

I’m mostly new to setting up SIP, I have done end technician stuff previously but nothing in the backend as it were. simply adding new devices, adding new users and the odd redirect and number blocking here and there.

so the extent of my knowledge is limited when it comes to PBX and SIP systems.

Current setup:
1 in use Telephone server. External SIP config to the current number.
80 SIP devices connected in a number of ways via DECT / analog cables.
160 + users configured with extensions ranging between 001 - 999
1 no longer in use Telephone server. External SIP config to the old number used in the old building.

What, I think, I understand I will have to do if this is at all possible with asterisk and our current systems:

  • Configure the SIP provider account onto asterisk.
  • Create an account on Asterisk for connection to Server1
  • Create an account on Asterisk for connection to Server2
  • Allocate enough channels (?) to each of these accounts for concurrent calls. (currently we have 10 concurrent externals calls available)
  • Create a Dial plan to redirect all incoming calls to extensions 001 - 499 to server 1.
  • Create a Dial plan to redirect all incoming calls to extensions 500 - 999 to server 2.
  • Create a Dial plan to allow for call transfer between the two preexisting servers, or at minimum allow for calls between the two servers for internal communication.

Thing is I have no idea how to go about this, or whether this is possible.

both preexisting servers are from Agfeo, asterisk has been installed onto a server which had no real use since the move from the previous location and is the only thing on the server.

Note: I have no say in the equipment we use, and replacing it is out of the question. If this isn’t possible I can go back to my boss and say this isn’t possible with Asterisk however this just means I have to look for another solution entirely for the same thing. he is dead set on using the preexisting servers for endpoint connection as we have spent so much on the licenses required for softphones, ldap connection, fax addon cards, and the servers themselves.

Any help would be appreciated.

It is possible that Asterisk is over-engineering for this, and a SIP proxy might be as good.

Most of this can be done with one or two line dialplans.

The ability to transfer will be more dependent on the commercial PABXes. It might not be possible to optimise transfers, so the signalling and media may have to go the long way round. Also, it may be necessary to provide routing digits to get the commercial PABX to send calls to the other half of the number range to Asterisk.

By default, Asterisk doesn’t limit the number of simultaneous calls.

Whilst the dialplan should be simple, there is more work here than you are likely to get for free here, and that is likely to involve studying the documentation for the commercial switches.

Many thanks for the quick reply!

I expected the internal calls between the two servers might have to take the long way round but was hoping for otherwise.

Ill take a look at a couple of sip proxy alternatives as well. I was just recommended by the IT director to use asterisk.

Just to confirm:
If I were to use Asterisk, was i correct in assuming that I would need to essentially remove the external sip config from the current server and have this activated on the Asterisk / proxy server instead, and then the dial plans to move the calls to the other servers?

Again, I really have no idea about SIP or VOIP other than configuring end point devices and creating users and extensions in our current system.

I also understand I’m asking alot for what seems free support. so again thank you.

You would have to move the ITSP account.

Most of the rest of this is about routing, rather than VoIP.

Take a look at Kamailio, or dSIP Router which is a GUI with very limited control over Kamailio. A Kamailio server with very limited resources can easily handle what you need.

With that being said, while Kamailio is probably a better solution for this, it is way harder to configure than Asterisk. So if you don’t have a budget to hire a Kamailio consultant, I’d suggest doing it with Asterisk.

Your boss is on to something! :slight_smile:

That’s two lines of dial plan to put into your extensions.conf file (to do call routing) – it might end up being three or four lines, total, is all you need in that file. Most of the work will be in your pjsip.conf file configuration for your three SIP connections (server 1, server 2, and ITSP). And that is about it for the Asterisk part, at least for testing purposes.

Asterisk can do those things, too; except, it is often doable on Asterisk without additional license fees (the trade-off is that your time is spent instead.)

That depends on the ITSP as they might offer some sort of fail-over registration option. For example, you could configure Asterisk as the primary to receive all 000-999, but still allow servers 1 and 2 to act as backups for their respective halves of the DID pool. (And if they don’t, there’s probably other fish in the local ITSP sea that do.)