On Mon, Sep 12, 2022 at 11:02 AM Dan Cropp <d...@amtelco.com> wrote: > If the transfer is using SIP REFER, the carrier/switch (equipment that > sends call into Asterisk) would be required to support it. > > > > SIP REFER tends to be used with switch vendors as opposed to SIP providers. > > Not sure if there are any SIP providers who support the REFER feature. > > Generally, SIP REFER requires SIP Provider or Switch equipment to perform > an internal patch/bridge for the 2 parties to hear each other. >
The only other option from a SIP perspective is re-INVITEs for direct media between both sides, which requires precise conditions in order to achieve and is enabled respectively in SIP channel driver using the direct media option. (directmedia in chan_sip, direct_media in chan_pjsip). -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
