CIKTel account

Discussion in 'General' started by itjumper, Jun 5, 2009.

  1. itjumper

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    I have an account from CIK (www.CIKTel.com). This account is supposed to be an extension in CIK system, i.e. using SIP phone. I want to config it as a trunk in asterisk. This is my trunk setting

    canreinvite=no
    context=from-trunk
    fromdomain=CMTor1.ciktel.com
    fromuser=xxxxxxx
    host=CMTor1.ciktel.com
    insecure=very
    nat=yes
    qualify=yes
    secret=yyyy
    type=friend
    username=zzzzzzzzz

    Note that CIK has different username and auth id. With the above setting, I can make call from asterisk. But I kept getting SERVICE UNAVAILABE in the log and I could not receive any call.

    I got the same symptom when using other provider (extension, not real trunk)

    Anyone has any idea.
     
  2. jgutierrez

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    Try the following:
    1. Set on /etc/asterisk/sip_general_custom.conf
    context=from-pstn
    2. Execute from the shell:
    asterisk -rx "reload"
    3. Set the appropiate inbound routes, under PBX->Inbound Routes
    I would confiure a route, that would accept any number, for instance:
    any DID/any CID
     
  3. itjumper

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    I should have mentioned that I have also defined an inbound route to an IVR. I think from-pstn is the same as from-trunk.

    thanks
     
  4. dicko

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    Maybe you need a registration entry also?
     
  5. itjumper

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    I did have a registration entry. I should have mentioned that too. My bad.

    Anyone try treating an extension on another system as a sip trunk?
     
  6. dicko

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    Then does:

    sip show registry

    from asterisk CLI confirm it is working?
     
  7. itjumper

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    yes, sip show peers confirms that it is registered.

    I can make outgoing calls, just incoming cannot reach asterisk
     
  8. dicko

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    If you are successfully registered with the provider (VSP, extension on another SIP device, gateway, whatever they are all essentially the same thing) then any ensuing sip "invite" to that resolvable SIP url, will be so passed to your system (your device is the ONLY device attempting to register to that SIP endpoint, correct?), then you maybe should do

    sip debug ip <ip of the registered provider>

    and call that "endpoint" to see how it is being handled/rejected, possibly incompatible codec or unavailable resource but the packets should definitely arrive and give you a clue (your SIP register packets were obviously sent and successfully acknowledged so it can't be a routing problem) as to what isn't working.
     

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