Force Sip URI Call Through PBX

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When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc …

Is there a way to force URI calls through the PBX? I have found no configuration at the client or at the server level. Do you know any softphone that will allows me to do this ?

Thank you and have a nice day, Julien

One thought on - Force Sip URI Call Through PBX

  • If two phones are calling each other directly then there is no server setting that will reach across the Internet and grab the call. You need to insure that the call is proxied through your PBX. That’s just a setup in your softphone. You will need to ask about that on a mailing list for that software.

    One thing that I do so that I can call SIP phones from my regular phone through an ATA is set up extensions for them in Asterisk like this.

    exten => 6135553638,1,Dial(SIP/

    Looks like a regular call to my users but bypasses the PSTN. This might work for you as well.

    Whatever solution you use, you may want to look at directmedia settings. If you can talk directly to another SIP client Asterisk may step out of the picture anyway not allowing you to record the call. Turning that off forces all the calls to be proxied through you even if they could talk directly.