I’m having a setup where my asterisk PBX connects to PSTN via a single SIP trunk. Now, when I transfer or redirect incoming calls from the SIP
trunk to another number which is routed through the SIP trunk, my asterisk stays on the way; it just dials out the new destination number the call is transferred/redirected to and connects the newly dialed channel to the existing incoming channel.
Since these two channels are in the same SIP trunk, would it be possible to tell the trunk SIP server to not involve my asterisk anymore, both for signaling and media data? Or is this inherently not possible via SIP?