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VOIP Provider Extension not working

Discussion in '3CX Phone System - General' started by wolfgangkr, Mar 18, 2011.

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  1. wolfgangkr

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    Dear forum,

    I have posted a similar topic on this, yet in between understand that my problem maybe somewhat different that I originally described.

    What I want to do is, I have an external voip provider, where I have a sip account, this sip account again holds a telephone number that I would like to rout inbound to a certain destination. So far I have created a voip provider and entered the credentials so, that I can see that the sip account has been sucessfully registered with 3cx. Once I look into the network analyzer I see the following statements:

    78739 22:41:59 18.03.2011 2962.9615642 SIP SIP:Response: SIP/2.0 200 OK {SIP:66, UDP:65, IPv4:64}
    79943 22:42:59 18.03.2011 3022.9806867 SIP SIP:Request: OPTIONS sip:000387809879@95.1.1.1xx:5060;rinstance=6d21eb2e221e7c57 SIP/2.0 {SIP:66, UDP:65, IPv4:64}

    When I place a call, I would expect to register some kind of frames on the network analyzer, yet this is not the case. What shall I do here ? I have checked the firewall and all setting are ok.

    Thank you,

    Wolfgang
     
  2. willow

    willow Member

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    You would to actually try a call. Sip will send stun messages if you are using it ever 20 minutes or so. you will not see any packets for the trunk until a call is actually going out or coming in on it. once you make a call it the sip trunk will either try and register or complete depending on if you have a ip based or register based sip account.

    you would also need an inbound role and/or outbound role to route the call to a destination.
     
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  3. wolfgangkr

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    Dear Jeremy,

    thank you for your reply. I have really tried all I can but maybe my logic is somewhat limited ;-). Is there any documentation asides from the material provided by 3cx as to how to setup a voip provider with sip accounts and route the communication as I described it ?

    Thank you,

    Wolfgang
     
  4. willow

    willow Member

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    each provider usually has a sample config that they post in their support sections. but i think that the ones that i have read all require some form of sip knowledge.
     
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