[asterisk-users] force sip URI call through PBX

[Julien Sansonnens]

Hello,

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


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Julien Sansonnens