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ghannelais

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Posts posted by ghannelais

  1. Trixbox + Linksys SPA-942 phones + Uplink driver for Skype

     

    1- Skype to Skype call (direct, Uplink desactivated):

    If called party does not answer, ring lasts 60 seconds and then Skype Voice Mail is activated (which is normal as I set the Advanced Voicemail settiing to 60 seconds).

     

    2- Skype to Skype call (through Uplink)

    If called party does not answer, ring lasts 30 seconds and then I get the Skype busy signal.

     

    3- Normal call from Extension A to Extension B on the Asterisk system (no Skype, neither Uplink involved):

    If called party does not answer, ring lasts as long as programmed in the Asterisk server (120 seconds in my case)

     

    Can somebody confirm if the Uplink driver forces the disconnection after 30 seconds when no answer ?

    These 30 seconds are not long enough for permitting the Asterisk server to hunt correctly the extensions to ring.

    Any idea how to extend this time limitation?

  2. In my case, it's a Trixbox install.

     

    The Asterisk is a Trixbox 2.2 installation, no zaptel devices (ie, for timing).

     

    Dave

     

    Hi Dave

    Trixbox here too (v. 1.2.3) with Zaptel board and no problem with choppy sound.

    Asterisk on a Linux server

    Uplink on a Windows server

    SIP phones are Linksys SPA942 on all work stations

     

    This is my Trixbox trunk settings in case it could help:

    allow=ulaw

    canreinvite=yes

    context=from-trunk

    disallow=all

    host=dynamic

    nat=yes ;very important. uplink won't work without this

    secret=xxxx ;define your own password as in uplink

    type=friend

    username=xxxx ;make sure this is same to trunk name

     

    Good luck,

    Georges

  3. It seems the DTMF does not work.

     

    Same problem here.

    Uplink 1.30 (registered) works fine with our Asterisk server but we are unable to use the Digital Recepcionist, nor the Voice Mail as DTMF does not pass-thru.

     

    For a SIP-SKYPE -> SKYPE-SIP call, it seems DTMF tone is sent back to the caller who pressed the key and not forwarded to the called party.

     

    Please NCH try to solve this point asap or at least to explain the situation (many users have already mentionned the problem but so far no feedback).

    The Uplink software is pretty good and it is really disapointing that the basic DTMF is not functioning!

     

    Georges

    Mexico City

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