Running Asterisk 16 w/PJSIP
I’m trying to figure out how to handle Attended Transfers when running a cluster of Asterisk Servers. Currently we’re sort of clustering users by their department using Kamailio in front of Asterisk, but that makes it impossible to guarantee that attended transfers can be made between departments.
I’ve found this article on the WIKI, and I’m looking for some clarification: https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers
If I add the external_replaces context, configuring it so we’ll allow performing remote attended transfers to other asterisk boxes in our cluster, that should be good (I’m guessing).
However, I’m pondering this statement on the wiki article:
When Server B receives this INVITE, it will essentially swap this new call in for the call referenced by the Replaces header. By doing this, the final picture looks something like the following:
Does that mean that if Asterisk A receives a REFER request pointing to a call it doesn’t know, and executes the external_replaces extension, which in turn then would make a Dial to Asterisk B - will Asterisk B will magically notice this Replaces-header, and perform the swap, or do I need to handle this myself in whatever dialplan context the INVITE from Asterisk A comes into?