Skype Connect

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    crankshaft
    Participant

    Does anyone have any experience setting up skype connect ?

    I have subscribed to the service, created a trunk, and the skype page shows that the trunk registered successfully.

    I have also created an outbound route, which uses a “2[12]XX” pattern to catch a number such as 2102, 2201 etc etc.

    If I create a new extension, say 2101, how do I map this extension number to a skype account for dialing ??

     

     

     

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    crankshaft
    Participant

    It’s very frustrating having all my posts approved by a moderator, it is causing huge delays and it must be pretty obvious that I am not a spammer. 🙁

    Previously I used skype for asterisk, which was a paid module, that asterisk discontinued, I tried to install it, but it fails with an error:

    chan_skype.c:25:22: fatal error: asterisk.h: No such file or directory

    So given up with that now.

    I now have a trunk named ‘skype’ and it is registered successfully with skype.

    I have made some progress, I created a callgroup, and then edited the /etc/asterisk/ombutel/extensions_50_1_dialplan.conf with the skype username (user deleted from code below) as the html page will only allow me to enter digits:

    ;same => n,Dial(Local/102@cos-all/n&Local/12345678@cos-all/n,30,U(clean-variables))
    same => n,Dial(SIP/skypeskypeusername,90)

    If I then call the group (2000) It attempts the call and contacts skype, however the request is rejected as per below:

     -- Executing [2000@cos-all-post:9] Dial("SIP/102-00000000", "SIP/skype/skypeusername,90") in new stack
      == Using SIP RTP TOS bits 184
      == Using SIP RTP CoS mark 5
        -- Called SIP/skype/skypeusername
        -- Got SIP response 484 "Address Incomplete" back from 51.143.123.249:5060



    In TRIXBOX I can create a Custom Device / Connection and there is a ‘dial’ field that I can simply enter skype/username and it worked superbly !

    Any suggestions on what the problem may be ?

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    Check your register string, add the end of the register string you must add your DID number, e.g:

    USER:PASSWORD@HOST:PORT/YOUR-DID

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    crankshaft
    Participant

    Thanks, but I already have that, as I mentioned in the first post, the trunk is successfully registered with skype.

     

     skypesipuid:password@sip.skype.com/skypesipuid
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    crankshaft
    Participant

    This is the outgoing trunk config:

     

    [skype]

    type=friend
    username=skypeSipUserId
    secret=skypeSipPassword
    canreinvite=no
    insecure=port,invite
    dtmfmode=rfc2833
    host=sip.skype.com
    nat=no
    qualify=yes
    fromuser=skypeSipUserId
    fromdomain=sip.skype.com
    disallow=all
    allow=g729
    allow=ulaw
    allow=alaw

    Registration String: skypeSipUserId:skypeSipPassword@sip.skype.com/skypeSipUserId

     

    So how would I now dial another skype user (not  a PSTN number) using this trunk ?

     

     

     

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    You should post the asterisk log output generated when an incoming call arrives to VitalPBX.

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    crankshaft
    Participant

    Thanks, but any idea how to setup an incoming trunk for skype connect ?

     

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    crankshaft
    Participant

    As my outgoing trunk is set to “type=friend” I presume that both an incoming and outgoing trunk is created by vitalPBX ?

    Here’s the output from the asterisk log:

    chan_sip.c:24889 handle_response_peerpoke: Peer 'skype' is now Reachable. (165ms / 5000ms)

    So the trunk is registered, this is also confirmed on the skype manager profile for the user that is assigned to the sip profile, and you can see in the screenshot below that the trunk is registered.

    But when I call the skype user that is associated with the sip account, I see no activity in the asterisk logs at all, skype attempts to make the call but the call fails after about 30 seconds.

    So as far as I can tell, I have outbound/inbound trunk configured, but I am unable to either receive or make calls.

    Surely someone else must have done this on vitalPBX or am I the first ?

     

     

     

     

     

     

     

     

     

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    PitzKey
    Participant
    US

    I think you need to set type=user

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    You may use sngrep to verify if the call is really arriving to your PBX

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    sauro2
    Participant

    hi all, has anyone managed to make vitalpbx work with skype-connect? thanks

     

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