Installation report: 3CX on free.fr (France)

Discussion in '3CX Phone System - General' started by Watashi_FR, Dec 5, 2006.

  1. Watashi_FR

    Joined:
    Dec 5, 2006
    Messages:
    63
    Likes Received:
    0
    Hi there, I thought I'd share my experiences installing your product.

    Having tested Asterisk/TrixBox, I found this product to be too complicated to manage properly for lack of a decent management interface. I stumbled upon 3CX and, having seen the documentation and screenshots, immediately had a "wow" moment. I installed on a Windows 2000 Server (dual Xeon 900Mhz, 512Mb RAM) without problems. Registering our VoIP-line was somewhat tricky as we have two DSL-lines with two different providers (Orange and Free) and the server's gateway is set to Orange, but a static route in the gateway (plus the appropriate NAT entries) solved that. Now, having done a basic configuration, I have found the following challenges:

    - currently, no outbound calling is possible over VoIP. The server reports the following:
    15:10:53.703 Terminated c58 "3CX Phone"<sip:101@france.mentaljam.net> <sip:06XXXXXXXX@france.mentaljam.net> Call ended
    15:10:51.531 Calling c58 "3CX Phone"<sip:101@france.mentaljam.net> <sip:06XXXXXXXX@france.mentaljam.net> Send INVITE to [#09XXXXXXXX @Free]
    15:10:51.375 Routed c58 "3CX Phone"<sip:101@france.mentaljam.net> <sip:06XXXXXXXX@france.mentaljam.net> From: Ext:101; To: [#09XXXXXXXX @Free]
    15:10:51.375 CallConf::findDestination: Found destination [#09XXXXXXXX @Free] for caller Ext:101
    15:10:51.343 Registrar::checkAor: Registrar resolved sip:101@france.mentaljam.net as <sip:101@192.168.0.99>
    15:10:51.312 Incoming c58 "3CX Phone"<sip:101@france.mentaljam.net> <sip:06XXXXXXXX@france.mentaljam.net> Incoming call from 101
    (06XXXXXXXX is a mobile phone, 09XXXXXXXX is our VoIP number). Incoming calls are working just fine. Using XLite with a direct connection to our VoIP provider works both inbound and outbound.
    - Having configured voicemail, I called an extension from an external line (PSTN) and was eventually redirected to the extension's voicemail. After pressing the pound key and received "record message, then press pound" and did accordingly, was then given the "press 1 to save" message, but after pressing 1 no confirmation was given, the message was not recorded.
    - Having configured a Digital Receptionist to forward to an extension upon pressing 0, and redirected incoming calls to it, I received the recorded message, but pressing 0 did not work.

    Other than the above, it seems to be a marvellous system that shows a lot of promise, this is the kind of product we would like to sell to our customers!
     
  2. Watashi_FR

    Joined:
    Dec 5, 2006
    Messages:
    63
    Likes Received:
    0
    Oh and yes, that means you can contact me if you are interested in setting up something for the French market ;)
     
  3. archie

    archie Well-Known Member
    3CX Staff

    Joined:
    Aug 18, 2006
    Messages:
    1,309
    Likes Received:
    0
    Hi,

    Regarding your problem to make outbound calls through VoIP providers. Yes, I'm terribly sorry :oops:, it's our bug in latest build. We have already determined the cause of this bug and fixed it. After we alpha-test these latest fixes we'll make fixed build available for download.
     
  4. Watashi_FR

    Joined:
    Dec 5, 2006
    Messages:
    63
    Likes Received:
    0
    Installed the new build today and lo and behold, outgoing calls now work like a charm, both from local and remote phones. However, a new problem has arisen, my voip line shows as registered and on-hook in line status, but incoming calls are routed to my pstn line (which is what my interface is supposed to do when no other device has registered to handle SIP). Am I missing something?

    Edit: oops, forgot to change gateway on my server, my bad, everything works.
     

Share This Page