[asterisk-users] [Asterisk-video] (no subject)

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[asterisk-users] How to ignore REFER entirely with chan_sip or PJSIP ?

Hello, In my testing, I saw that Asterisk always included a REFER value in each INVITE's Allow header, no matter how allowtransfer/allow_tranfer was set. Is there a way to remove this REFER value entirely either globally or specifically for a given peer/endpoint ? Which telephony feature would l