Utilize SIP Trunk's Call Conferencing Feature

Hello again. I don’t know what to search or how to find resource about this.
In my previous post, we are successful in making my ISP’s VOIP service as a SIP trunk in asterisk/freepbx. My ISP VOIP has these features enabled - call waiting, call forwarding, and 3-way call conferencing.
Previously I can conference 2 mobile numbers by pressing the flash/R button on the handset.
However after integration to asterisk, I can only conference 1 mobile and 1 internal extension. When I try to call the second mobile number while the first mobile call is on hold, I get the message “all circuits are busy now”. How can I possibly accomplish this?

Is the message from Asterisk or your ISP?

The message originates from the Asterisk.

Providing Great Debug - Support Services - Documentation (freepbx.org)

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