Hi! Apparently this is possible; my asterisk server is doing this when my SIP phone redirects the call with a SIP REFER message. The phone is excluded from the call after it transfers the call.
I'll contact my ITSP if their trunk can also do this. Regards, Sree On 03/09/2017 11:03 PM, Sree Harsha Totakura wrote: > Hi! > > 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? > > Regards, > Sree > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
