Dismiss Notice
We would like to remind you that we’re updating our login process for all 3CX forums whereby you will be able to login with the same credentials you use for the Partner or Customer Portal. Click here to read more.

Frustrated with GXW4108

Discussion in '3CX Phone System - General' started by Firetec, Oct 23, 2007.

Thread Status:
Not open for further replies.
  1. Firetec

    Joined:
    Oct 15, 2007
    Messages:
    4
    Likes Received:
    0
    Hey guys, please help,

    I have a GXW4108 connected I followed everything perfectly in the 3CX provided setup here: http: //www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE. NEW LINK: http://www.3cx.com/support/

    One thing with the instructions, it says "Set the “Off hook Auto Dial(VoIP)” field to “ch1-4:yyyyy;” where yyyyy is the SIP User ID for the first line defined in the “Channels” page." In my FXO lines page there is nothing about Off Hook Auto Dial VoIP! I am using the latest firmware 1.0.1.2, but am having nothing but problems.

    Inbound routing seems to work, kind of, it takes 6 rings until the SIP phones start to ring!

    As for outbound, it doesn't work at all, do you have any suggestions? what other info do you need?

    -Rod
     
  2. alicic

    alicic New Member

    Joined:
    Oct 3, 2007
    Messages:
    119
    Likes Received:
    0
    Change "Call Progress Tones" in Gateway to Country secific values, that it might work. You can see call progress tones for every countr at this web page

    http://www.3amsystems.com

    regards,
     
  3. Firetec

    Joined:
    Oct 15, 2007
    Messages:
    4
    Likes Received:
    0
    FYI, I am trying to put this behind a Panasonic TDA100 system with analog terminal adapters for the FXO lines (thereby simulating the PSTN). If I connect it directly to a real POTS line, it works fine, if I call the number, after 1 or two rings the SIP phones start to ring, but if it is behind the Panasonic ATAs, it the SIP phones will only start to ring after 6 rings into the FXOs.

    Also how would I set it up so when SIP phone number 1 goes off hook it allways uses FXO port 1, and lets say SIP phone 7 goes off hook, it and only it uses FXO port #7.

    You see basically what I am doing here is an analog to SIP converter.
     
  4. namster

    Joined:
    Sep 15, 2007
    Messages:
    4
    Likes Received:
    0
    I did this and it works:

    Contact Grandstream Support for the latest firmware or visit http://www.grandstream.com.
    Set the following values in the FXO Lines web configuration page:
    Enable Current Disconnect to Yes (if the PSTN provider utilizes Current Disconnect).
    Current Disconnect Threshold: 300
    Min Delay Before Dial PSTN: 750



     
  5. Firetec

    Joined:
    Oct 15, 2007
    Messages:
    4
    Likes Received:
    0
    I did all that you suggested
    "Contact Grandstream Support for the latest firmware or visit http://www.grandstream.com.
    Set the following values in the FXO Lines web configuration page:
    Enable Current Disconnect to Yes (if the PSTN provider utilizes Current Disconnect).
    Current Disconnect Threshold: 300
    Min Delay Before Dial PSTN: 750 "

    That was not the problem, the issue is that being behind a phone switch such as a Nortel or a Panasonic, you have to deal with the different ring cadences, or ring tones. When I was testing the system I was always calling internally thereby passing the internal ring cadence to the[/color] analog terminal adapter (SIP phones didn't ring till 6 rings). When I called from the outside world, back into the ATA, it will ring after 5 seconds, because the ring cadence is close to the regular PSTN, 2 on 4 off.


    I am still having issues with dialing out. like I said before, when I put a real PSTN line into the gateway, it works great everytime, but when I put it behind the ATA, it doesn't work. I listen to the dial tone of the PSTN and the dial tone of the ATA and the are identical, I am using 1 stage calling. Any suggestions?

    -Rod
     
Thread Status:
Not open for further replies.