skype for asterisk

Discussion in 'General' started by byaru1, Oct 5, 2009.

  1. byaru1

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    Folk,
    Has anyone gotten skype for asterisk working on an elastix 1.5.2 system? Is there any detailed guide somewhere for this?

    One other question - i'd like to hear fron anyone that has qorking SIP trunks, my SIP trink to terrasip doesnt make any outgoing call. constant error message "all circuits are busy at this time"

    thx
    eb
     
  2. byaru1

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    Folks,

    Anyone gotten skype for asterisk working on Elastix 1.5.2 ?

    thx
    eb
     
  3. Chilling_Silence

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    No idea about skype for asterisk, but I've tried a few other skype gateways etc and all worked just fine

    As for your "all circuits are busy" error, login to the Asterisk Console with:
    asterisk -r

    Now, run:
    sip show registry
    sip show peers

    What results do you get?
    SIP trunk settings are generally unique to each provider.

    http://www.terrasip.com/xoshop/faqdesk_ ... ?faqPath=8
    Google is your friend, just as much as FAQs ;)
     
  4. byaru1

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    Huh, what 'gateways' were those? The skype for asterisk configuration is really meant for the CLI savvy - I just cant figure out a way of doing it from FreePBX/elastix UI!

    I actually managed to get those working - thanks.

    I really would love to hear from anyone that managed to setup skype for asterisk on Elastix.

    eb
     
  5. Bob

    Bob

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    Skype for Asterisk (SFA) Works fine....

    We are using it mainly for incoming (which was our interest in the product). Comes in as a Trunk just like a normal call, and using inbound routes via the GUI. At the price, it was well worth it for us.

    Outgoing works, can be configured via GUI, but drop the idea that many have of dialling from a Phone. Setup as custom extensions. Also a known problem with Skype names with spaces and dots in the name. The general concept is that regular Skype names that you call would be setup as custom extensions (almost like a speed dial).

    Once the base config file is setup (about 5 lines), the rest is done via the GUI which anyone with a "monkey see monkey do" approach can look after (in otherwords a lightly trained person could do it).

    Regards
    Bob
     
  6. cassioac

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    Hi.. I'm one more migrating from trixbox from elastix. Did you use "skype for asterisk" or "skype for sip" solution? Just wondering, to make sure I use the best product.
     

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