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Caller ID

Discussion in '3CX Phone System - General' started by Anonymous, Sep 28, 2006.

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  1. Anonymous

    Anonymous Guest

    I am not getting caller ID at the internal ext from my VOIP line. I can see the CLID in the server status, but when the call is presented to the ext, I get the ID for the VOIP line not the caller.

    How do I fix this?

    Absolutely outstanding product! I have messed with Asterisk@home for 6mo to a year on and off. Never could get it to except calls from my VOIP provider. Had yours up and working in about 20 minutes.
  2. archie

    archie Well-Known Member
    3CX Support

    Aug 18, 2006
    Likes Received:
    Thank you for your feedback!
    Please, show us records which have CLID (just copy it from server status window and paste in message) and we will try to fix it.
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  3. Anonymous

    Anonymous Guest

    08:47:34.796 >>> Log started >>>
    08:47:34.812 SessionMgr::run ** Adding transports **
    08:47:34.812 SessionMgr::run ** Initialize DUM **
    08:47:34.812 SessionMgr::run ** Initialize Server Authentication **
    08:47:34.812 SessionMgr::run ** Initialize Client Authentication **
    08:47:34.984 SessionMgr::run ** Initialize Lines Manager **
    08:47:35.062 ::DBA ** Database connection Ok **
    08:47:35.250 ::ListenConnect SL: connected turske-tn4tuygv:0/MediaServer at [turske-tn4tuygv:0]/MediaServer
    08:47:35.250 ::MediaServerConnected Media Server is connected
    08:47:36.875 LineImpl::update Line: register line 8778987152
    08:47:36.875 SessionMgr::run ** Initialize Status Monitor **
    08:47:36.937 SessionMgr::run ** Start Session Manager **
    08:47:36.937 SessionMgr::thread ** Entering Main Loop **
    08:47:37.578 LineImpl::registered Line [#8778987152 @Quantum Voice] has been registered at provider
    08:49:31.609 ::ListenConnect SL: connected turske-tn4tuygv:0/PHPExtension_0 at [turske-tn4tuygv:0]/PHPExtension_0
    08:50:18.343 CallConf::saveCallEvent Incoming c1 <sip:4192825261@> <sip:8778987152@> Incoming call (before routing)
    08:50:18.734 CallConf::saveCallEvent Routed c1 <sip:4192825261@> <sip:8778987152@> Src=<sip:8778987152@sip.digitalrealm.net>; Dst=<sip:100@;cid=0>
    08:50:18.796 CallConf::saveCallEvent Calling c1 <sip:4192825261@> <sip:8778987152@> Send INVITE to <sip:100@;cid=0>
    08:50:24.171 CallConf::saveCallEvent Terminated c1 <sip:4192825261@> <sip:8778987152@> Call ended
  4. Anonymous

    Anonymous Guest

    What I get at the 100 ext is the 8778987152 number (the one I'm calling) and not the 4192825261 (where I'm calling from)
  5. 3CXsupport

    3CXsupport New Member

    Aug 21, 2006
    Likes Received:

    I am sorry but this is currently not supported, but we will defenitly be looking into it.

    Thank you for the very encouraging words!
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