Eugene

Member

Last active 13 years ago

  1. 13 years ago
    Thu Jun 9 06:15:34 2011
    Eugene started the conversation I can't see OOH323 trunk on panel.

    Hi!
    The OOH323 custom trunk is not appeared on FOP2
    autoconfig-buttons-freepbx.sh output is:

    [OOH323/TRUNKNAME/$OUTNUM$]
    type=trunk
    label=OOH323-trunk
    queuecontext=from-queue
    privacy=none
    channel=1-4

    but I can't see the appopriate trunk button

  2. Fri Dec 10 06:21:08 2010

    Did. However, because then what version of the module has not changed, it had to uninstall and reinstall it, with the loss of settings fop2 buttons, groups and users

  3. Thu Dec 9 12:17:39 2010

    Please update to the latest beta...

    done
    new autoconfig-buttons-freepbx.sh don't found field "external"

  4. Thu Dec 9 11:09:26 2010
    Eugene started the conversation FOP 2.2 Doubling buttons on reboot dialplan.

    FOP2 Buttons is cloned with each reboot dialplan. After 3 reboots I get a triple set of all elements of the panel

  5. Mon Dec 6 06:25:44 2010

    OK! It's not a bug, it's a feature.

    Thank you, Nicolas!

  6. Fri Dec 3 05:21:45 2010
    Eugene started the conversation Hungup the channel from button menu.

    Hi, Nicolas
    Now I can hangup whole trunk only via HANGUP button on topbar.
    It would be nice feature to have a reset single channel from the context menu, similar to the CLI function "channel request hangup <CHANNELNAME>"

  7. Fri Dec 3 04:09:43 2010

    .... Be sure to have the "all" permission in both read/write in /etc/asterisk/manager.conf...

    read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate
    write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate

    ...Another issue that I see is the dial extension with the g0/ prefix. Not that is horrible, but that behavior is new, and I think is not good, as you dial a number, the g0 is used by the dial command only...

    This issue does not affect the display of channel status button. Вut I don't see channel status while DAHDI-->SIP is ringing.
    DAHDI/i1/100 ringing to SIP/6030 screenshot
    [url=http://www.flickr.com/photos/56594160@N02/5227634625/:ex7qglin]http://farm6.static.flickr.com/5124/5227634625_0854707b1a_z.jpg[/url:ex7qglin]

    And now call is answered
    [url=http://www.flickr.com/photos/56594160@N02/5227632049/:ex7qglin]http://farm6.static.flickr.com/5288/5227632049_e8f514956e_z.jpg[/url:ex7qglin]

  8. Thu Dec 2 10:05:06 2010

    ....If it works of some calls, it should work for others as long as you have DAHDI/ix as channel names. In any case send me the debug output as before for the other type of calls, look at the device names there.

    ОК, I will send it soon. There is no indications while DAHDI/iN/YYY rings to SIP extension. The indication is appears when call is timed out and transfered to voicemail system.
    *EDIT: The indication is appears when I pickup the phone too, but not while phone ringing.

    And to make it clear to other users, this happens with CUSTOM DAHDI TRUNKS for ISDN, not for regular DAHDI usage... right?

    I think it's not a CUSTOM DAHDI TRUNKS for ISDN, it's STANDARD DAHDI TRUNKS for ISDN (DIGIUM Wildcard TE122). But new version of DAHDI and LibPRI used in Asterisk 1.8 was changed for best q931 support. Therefore, such a system of naming ISDN channels is standard for all installations of Asterisk 1.8

  9. Thu Dec 2 04:33:49 2010

    Yes, it works, but for SIP->Dahdi calls only. There is no indication for DAHDI->SIP call until it has been transfered to voicemail. I see call indication within voicemail prompt.
    Also, I think "DAHDI/i$ZAPNUM" hardcoded in autoconfig-freepbx-buttons.sh is not suitable for a mixed installation of analog and digital DAHDI cards.

  10. 14 years ago
    Wed Dec 1 03:50:23 2010

    done

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