From ATAs via 3CX through a 2811 to a PBX and out - doable?

Discussion in '3CX Phone System - General' started by Sagewah, Aug 31, 2007.

  1. Sagewah

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    Hello again!

    I posted a while back about trying to get an ugly setup to work. With help from here I got close to making it work, then I went away to get married and all the networking gear got upgraded.

    Things still don't work right :)

    The current setup: head office is still running a traditional PBX setup with phones in a 91xx range. We now have a number of home users hanging off private links, with analogue phones behind linksys PAP2Ts. Their numbers are in the 431x range; their IPs 192.168.4.x. We also have two branch offices, but for the moment I don;t care about them. They've got little PBXs and that just complicates things.

    At this point there is no central sip server. The PBX is set up to dump any calls starting with 43.. over an e&m to a cisco 2811. The cisco then decodes the dtmf, looks up the dial peer and routes the call directly to the fixed IP of the relevant phone.

    When the home users want to call in, they can either call a 91.. extension which works, or if they dial a PSTN number with a rpeceding zero that works too. however, they can't call each other, and they can't pick up parked calls - 91.. users transfer a call to a 52xx number, which the receiving party then calls to pickup the conversation.

    I'd love to simplify things by running 3cx. It would basically enable the 43.. users to call each other and have their own on hold and messaging service and so on. the problem is that I can't find a way to make 3cx route calls to the 2811 properly.

    currently, the ata's have this for a dial plan:
    Code:
    (9xxx<:@192.168.1.254>|430x<:@maroochydore-sip.local>|431x<:@192.168.1.254>|5xxx<:@192.168.1.254>|3xxx<:@townsville-sip.supa.local>|0xxxxxxxxxx<:@192.168.1.254>|0xxxxxxxx<:@192.168.1.254>)
    
    Fairly straight forward, but not entirely effective. Calls to other 43.. numbers don't work, presumably because the pap boxes aren;t forwarding DTMF properly. The real show stopper is that calls to 52.. don't work. I'm under a little pressure on that point...

    Is there a way to have the 3cx server do roughly the same thing? There's no VOIP provider to register with, the 2811's not a gateway device in the sense 3cx seems to want, but it does have the functionality - if only I could convince 3cx to just send calls to it and accept calls from it.

    If I set up a dial peer on the cisco like this:
    Code:
    dial-peer voice 4334 voip
     destination-pattern 4334
     session protocol sipv2
     session target ipv4:192.168.1.70
     codec g711alaw
    
    i.e., pointing directly at the ATA, it works. when I point it at the 3cx IP, it doesn't, even if the phone is registered. And outbound calls from the ATA boxes only work when they're told to use the cisco as the outbound proxy and have this dial plan, effectively bypassing 3cx.

    The really weird thing is that calls to extensions behind the PBX work, but not to other IP devices. Should I be throwing a bit of redirect ip2ip into the mix on the cisco?

    I'm open to ideas :)
    [/code]
     
  2. Hoover87

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    How did things workout wth this project?

    How is the marriage working out? Just got married in Aug myself. Congrats!
     
  3. Sagewah

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    G'day!

    In the end, I had to abandon the use of 3CX. I just couldn't make it and the cisco play nicely.

    What I have in place now is a very basic setup: the ATAs hang off the network on static IPs and there is a dial peer defined for each unit on the cisco. the dial plan is set up on the ATAs to basically just pump out any calls directly to the cisco; 4 digit numbers get passed directly to the PBX via the E&M lines and regular numbers are dialled with a leading 0 which gets stripped by the cisco which then passes them to the PBX. Works a treat but they don't do anything fancy - no voicemail, no music on hold and so on. Initially I had a lot of trouble getting them to talk to each other even though that could call out OK; turns out I had not remembered to allow connections sip to sip - one config line later and away she went.

    There will hopefulyl be some down time here over the christmas break and I'll have another stab at it then. I've got to come in a rebuild / upgrade the domain controller/exchange box so I'll be in here with not much to do a lot of the time; may as well put it to good use!

    congrats on getting married! :) We had the weeding next to rather than on the beach because it rained. It was beautiful warm and sunny the day before; it was beautiful, warm and sunny the day after, but on the Day it was drizzling rain and damn near freezing :) But the recpetion was a bang-up party so it didn't seem to matter once everyone thawed out ;)
     
  4. Hoover87

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    Give the following link a whirl:

    http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_chapter09186a0080751974.html#wp1262484

    Also, add 'dtmf-relay sip-notify' to your dial-peer the next time you try.

    Good times with the wedding! We got married near a lake here in Seattle. A cloudy morning made us think it was going to rain, but fortunantely the sun made an appearance in the afternoon and we got lucky!

    Keep us updated on your project. Have fun!
     

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