Setup to use Elastix as a SIP Gateway for PBX.

Discussion in 'General' started by pinkertonfloyd, Apr 2, 2010.

  1. pinkertonfloyd

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    I personally run Elastix in my office, runs our whole system... works great. Using voip.ms for my service and cut our phone bills down $330/mo on per minute charges alone.

    I have a customer that has a very large digital (not voip) PBX (with 700+ Extensions). They don't have the $$$ to replace the whole PBX. The system currently has 2 T1 cards at each site, one that goes back to a Master PBX, and one that goes out to the local Telco).

    What I'm thinking is we take an elastix box with a T1 Card (I even have an extra Rhino card sitting here in fact), plug one of the master PBX's T1 into that, make that route on the old PBX the lowest cost route, and then create a IAX2 Trunk out to a voip provider. The idea is they'd get cost savings from VOIP, yet still have the POTS lines as failover. I don't see why this couldn't be done... I even found a whitepaper from Digium talking about it... but I've yet to find anyone who's done it, or some routing examples to push me in the right direction.

    Diagram

    Old PBX (site) <-T1-> Main PBX <-T1-> Elastix Box -> IAX Trunk to Voip Provider
     
  2. dicko

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  3. mbit

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    Hi

    Basically yes you can definitely set this up. What you will need to do is set the elastix machine as the T1 network terminator. The PRI mode you will be using is PRI_NET. The cable you will need between the phone systems is a T1 cross over. This uses 2 pairs in your RJ45 connector. Pins 1,2 then 4,5. At the other end you swap the pins. You can look up a T1 cross over in google.

    Once you have set all this up you the elastix machine will just act like a normal T1 provider. The original PBX can send phone calls to it like normal. From there you just route your calls out through the IAX provider.
     
  4. fmvillares

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    take in account that you will only have 23 simultanous calls between the 2 systems...for 700 extensions mmmm looks to me that is a little short...
     
  5. dicko

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    greeting fmvillares:


    My suggestion to pinkertonfloyd , is not to go "hmmm" but to go to
    Code:
    http://en.wikipedia.org/wiki/Erlang_(unit)
    where practical computational formulae may be found to compute this particular case (or any) needs.

    But if it currently carries the traffic then it will of course continue to, as that bit is not changing.

    The more pertinent bit is to compute the network bandwidth needs, there are several tools posted here and elsewhere as to how to dimension that variable but basically provide a minimum 25% overhead for g711 over IP compared with the same over TDM, and of course the Erlang stuff if you don't know how many concurrent ds0's (B channels) are maximally used

    Trunking can improve that overhead, but you need to find a provider that will so do that.

    You can of course also replace your PBX/PBX tie trunking with VOIP also but be aware you will likely loose your five nine's reliability for all such replaced trunks.

    JM2CWAE

    dicko
     
  6. fmvillares

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    hey dicko whats up...i didnt want to go there to erlang calculations because reading the mail i think the original owner of the post is not an "expert per se" so i dont want to spice things up in the first mail jeje

    you understand...
     
  7. pinkertonfloyd

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    Thanks everyone for the help...

    This customer is using ITT 3100's aka: Cortelco/Eon Millennium Switches. While they have 700 extensions, about 90% of calls are intra-extensions. They're currently using 1 T1 per site, then a second T1 that goes back to the main switch. My idea is to setup a test box for one of the T1's to prove the concept, then move up to replacing more of the T1's one by one. The Millennium is a PITA to program, but the hardware is rock solid.

    One of their sites is running a Hipath (Siemens), and the licensing fees just for SIP outbound on that system alone can buy a REALLY nice Elastix Server, with T1's out the wazoo.

    Internet traffic is Fiber, they have over a 200Mb pipe... not worried about bandwidth or latency. They also have Gigabit Ethernet fiber between sites.
     
  8. fmvillares

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    but with your scenario you only could upgrade with t1 connections...nos sip because the costs of license and not analog...so did you have enough t1 to interconnect the systems???
     
  9. pinkertonfloyd

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    The customer has enough T1's interconnecting their sites, the idea is really to bring the ability to use a SIP/IAX provider to their system to lower per-minute charges for outgoing calls. The current Telco charges 10-15 cents per minute... Bills go into the thousands per month. So the idea is to swap some of the T1's with Elastix boxes, and set them as the lowest-cost route. They have spare T1 cards available, so I'd most likely be adding routes.

    We could use the Elastix boxes to replace the site-to-site T1's, but the issue in this case is that Management doesn't feel Asterisk systems are "reliable" (We know that's false). So the idea is to slowly bring things in, and start saving $$$ and proving that the system. My guess is that after some time, they'll replace the whole systems with Elastix as it will prove itself as reliable (or more) as the older system, plus add Unified Communication features (VM to Email, etc) without the high license fees that systems like the Hipath requires.
     
  10. fmvillares

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    ok then its just a matter of disabling some t1 to teco and connect to asterisk boxes with iax interconnects and voila...my asterisk systems are about 99,95% reliabililty in dual server redundant configs...

    but of your bosses think asterisk is bad choice tell them that yahoo migrates all of its infraestructure to asterisk...google uses freeswitch...etc etc etc
    and the biggest individual player right now in the telephony market is asterisk making cisco avaya alcatel etc. eat dirt....

    best regards
     
  11. dicko

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    Then an Asterisk based solution is a good choice, although Asterisk is highly reliable, unfortunately largely the "internet" is only as reliable as the underlying carrier, and they are generally less than five-nines, YMMV, and you generally get what you pay for :)

    To integrate the VMail of Asterisk with your legacy PBX is quite possible but takes a little extra work as you need to have the PBX send unanswered calls back out the trunk with some idea of what "directory number" was not answered and so handle it in Asterisk/FreePBX, you will also have to write a custom script to send the message waiting info back into the PBX, (turn the MWI lamps off and on) and further program the "Voice mail" button to go to the external Asterisk system, so this usually takes a lot of familiarity with the provisioning of the legacy PBX. and a modicum of scripting ability in linux, there are some examples in the original post I sent you.

    To maintain the ability to failover to your TDM trunks you will need more T1 interfaces and either let the PBX do the fail-over or Asterisk, if you let Asterisk do it then I suggest you look into a clustered system and put your T1's on a redfone or xorcom or . . . device that can better handle a hardware/power/other FU failure at the service level your client expects.

    A chain is as strong as it's weakest link and your PBX's have probably been "up" for years, but are built to a higher level of redundancy than your average VOIP box is.

    dicko
     
  12. w1zard

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    I'm stuck in the "...route your calls out through the IAX provider..". help..... I don't know out to route call's form one trunk to another...
     
  13. dicko

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    It's much better written up in the blogs under the support tab at the top of this screen.
     

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