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BCM450 to 3CX

Discussion in '3CX Phone System - General' started by jholcombe, Oct 21, 2015.

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  1. jholcombe

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    I'm trying to connect a BCM450 and 3CX system together. Does anybody have any experience with this? It would be especially helpful if there was some way to get 3-digit extension dialing to work between systems so this transition could be transparent to our end-users.

    Thank you,

    --John
     
  2. jasit

    jasit New Member

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    I am pretty sure the last system I had was a BCM450, it's been a couple of years. are you using PRI for your phone lines or are you already on voip?

    Depending on how many users you have, if it's around 50 or less, you are probably going to have a easier time just doing a switch over then trying to train the users on how to call the new phones versus the old phones.

    If you are able to do sip from your BCM450, you will need to link it using Bridge mode on your 3CX system (Bridge mode is the only way that you will be able to do the direct extension dialing otherwise it has to go through the digital reception on either end.
     
  3. jholcombe

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    Thank you Jasit!

    Currently the BCM450 is connected PRI for the phone lines. I had also read that the 3CX and BCM systems could possibly be interconnected via a T1 Trunk. We're migrating about 100 users. The users would be migrated as they move to their new offices. The construction is going in phases and will take possibly a year or more to complete. Some of the offices aren't even connected to the same network (they are using older Nortel analog phones). I'd rather not cable these offices with Ethernet since they're going to be demolished anyway within a year. The new offices will be connected with POE switches, so when each user moves over they can transition to the new system.

    I was thinking on 3cx I could set up a rule to dial 8+old extension to talk to the users on the BCM system. On the BCM side I was thinking I could edit the dial plan to create a similar 8+extension to talk to someone on 3cx. Then I would edit the users extension on BCM to FWD ALL of their calls to 8+extension as they move. Likewise, on 3CX, I would FWD ALL of their calls to the BCM (until the user moves to their new office), which would route the call through the T1 to the new 3CX system. Their extension and phone # would stay the same so they don't have to print new cards, etc...

    On outbound calls from 3CX... I was thinking easiest might be to use a SIP provider and change caller ID to their old number until I get everybody moved... Unless there's a way to also forward those calls through the T1. I haven't got that far yet.

    Thanks for your help.

    Take care,

    --John
     
  4. jasit

    jasit New Member

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    K, you will want to Talk to Patton, they have a SN4970 which has 4 T1 PRI ports on it and converts it to VOIP for the 3cx box to connect to.. You can work with their support team, They can help you with Routing Traffic between the PRI to the 3cx system and also the BCM450.

    I had worked with them when we first tried the transition, We didn't get it working correctly and that was because I didn't know how the bridge mode worked in 3cx. Every call that was made ended up going to the digital receptionists versus the call going to the actual desktop.
     
  5. jholcombe

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    I now have a T1 Loopback cable connecting a Patton SN4970 to the BCM450 through a Digital Trunk Interface. I am able to call between both devices, but the audio quality is awful. I've made sure the framing and all settings match on both sides, with the exception of the clock (one must be the master, one must be the slave), and the Endpoint Type (one must be Net, and one must be User). There's a hissing noise and it sounds like the gain is set way too high on both sides. Any suggestions on getting the sound quality better? Other than the sound quality I think I have something that will work!!!
     
  6. jasit

    jasit New Member

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    I would check and see if you have the PBX handle the audio set. It might help.

    John
     
  7. jholcombe

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    Just an update, there were several issues with the template included with 3CX that prevent the Patton T1 gateway device from functioning properly in the USA. Most specifically, US is not one of the countries listed. I called Patton and they helped make the necessary manual changes to the device so it would work.
     
  8. jholcombe

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    Regard this issue, I did speak to someone with Patton tech support who was very helpful. The issue is (was - not sure about the current templates - I see there was a template update) that there was no US option in the template. Here are the lines I believe that are necessary to fix the problem:

    1. Regarding the static noise, the pcm law-select setting was wrong and should be:
    system

    ic voice 0
    pcm law-select uLaw


    2. The call tones were all European, and I wanted them to sound more US-like. Here are the better tones for the US:
    profile call-progress-tone defaultDialtone
    play 1 1000 425 -6

    profile call-progress-tone defaultAlertingtone
    play 1 1000 425 -13
    pause 2 4000

    profile call-progress-tone defaultBusytone
    play 1 300 425 -7
    pause 2 200

    profile call-progress-tone defaultReleasetone
    play 1 300 425 -7
    pause 2 400

    profile call-progress-tone defaultCongestiontone
    play 1 300 425 -7
    pause 2 400

    profile call-progress-tone US_Dialtone
    play 1 1000 350 -13 440 -13

    profile call-progress-tone US_Alertingtone
    play 1 1000 440 -19 480 -19
    pause 2 3000

    profile call-progress-tone US_Busytone
    play 1 500 480 -24 620 -24
    pause 2 500

    profile call-progress-tone US_Releasetone
    play 1 250 480 -24 620 -24
    pause 2 250

    profile tone-set default
    profile tone-set US
    map call-progress-tone dial-tone US_Dialtone
    map call-progress-tone ringback-tone US_Alertingtone
    map call-progress-tone busy-tone US_Busytone
    map call-progress-tone release-tone US_Releasetone
    map call-progress-tone congestion-tone US_Busytone


    interface isdn IF_ISDN_0
    route call dest-table RT_ISDN_TO_SIP
    call-reroute emit
    call-hold disable
    diversion emit
    use profile tone-set US
    caller-name
    user-side-ringback-tone


    Blessings!,

    --John
     
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