tommy_fc

Member

Last active 14 years ago

  1. 14 years ago
    Wed Feb 17 12:28:59 2010
    tommy_fc posted in Transfer causes DTMF signals.

    Could you please refer to the specific bug?

    Using trunk is not an option in production, but upgrading to 1.6.2.3 might be, if the issue is fixed.

  2. Wed Feb 17 10:25:25 2010
    tommy_fc started the conversation Transfer causes DTMF signals.

    When clicking on transfer in the user interface, there is a sound of DTMF signals in the users conversation. I assume this must be matched in the dialplan? What signales does it send? And how can I match this in my dialplan?

    I am using asterisk 1.6.2.2, a pretty stripped down version, so no trixbox or other GUI.

  3. Fri Feb 5 12:21:24 2010
    tommy_fc posted in Presence Update.

    I see the same issue with only numeric sip users such as SIP/25 and so forth.

    Is there a newer updated beta? It's been a few weeks since I downloaded it.

  4. Thu Feb 4 18:21:36 2010
    tommy_fc posted in Presence Update.

    I can verify this behavior on 64bit as well. The event manager seems fine, but the GUI does not update - unless a refresh and a re-login of the page is done.

  5. Thu Feb 4 16:58:48 2010
    tommy_fc posted in Having the phone update status..

    More on this, I see that it does transfer a message that gets logged in the user interface:

    Event: UserEvent
    Privilege: user,all
    UserEvent: FOP2ASTDB
    Action: UserEvent
    Family: fop2state
    Channel: SIP/26
    Value: Do not Disturb

    And the database thus shows:

    asterisk*CLI> database show
    /fop2state/SIP/26 : Do not Disturb

    I can easily enough program me around this for do not disturb, but will I be able to map it with hints?

    And on a related note - will there be custom messages such as (out until xx) or similar?

  6. Thu Feb 4 16:32:27 2010
    tommy_fc posted in Having the phone update status..

    Thanks for the quick reply.

    Changing my status in the FOP2 planel to "Do not disturb" does not seem to be detected by asterisk. Is there some configuration needed for that?

    Edit: I am using asterisk 1.6.2, and have these settings set in manager.conf:
    permit=127.0.0.1/255.0.0.0
    read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
    write = system,call,agent,user,config,command,reporting,originate

  7. Thu Feb 4 16:16:40 2010
    tommy_fc started the conversation Having the phone update status..

    Our phones have the possibility of accessing a URI, when they change state, either by DND, ringing or other. It would be very nice to use the phone to change our status in the FOP interface. Is that possible?

    Is there some URL I can hit to accomplish that? In which case, what URL?

  8. Thu Dec 3 17:03:03 2009
    tommy_fc posted in Not log in on centos5.

    Interesting reading. Thank you very much for the quick, precise and informative post.

  9. Thu Dec 3 10:45:09 2009
    tommy_fc posted in Not log in on centos5.

    I am having the exact same issue, except on Ubuntu 8.04

    It works in our testing environment, but not in production. There are two main differences: The test environment runs a free version while the prod is registered. The production environment is also fire-walled.

    Do I understand you correctly when you say that the browser has to have access to the fop2 service itself. That the event traffic does not go through apache?