What are SIP trunks. There is no such concept in SIP or Asterisk.
Asterisk supports REFER and REFER/Replaces when acting as a UAS, whether the peer is a phone or another PABX.
It also supports REFER, but not REFER/Replaces, as a UAC. Again it doesn’t care whether the peer is a phone or another PABX.
This is for chan_sip, but I assume that the same is true for PJSIP.
Asterisk doesn’t automatically de-trombone calls. If it receives REFER when acting as a UAS, it will not forward REFER upstream, but will rather just brdige the call internally. You have to use the Transfer application, rather than DIal, to send a REFER.
SIP Trunk is nothing but the TRUNK that is configured as a part of “Connectivity->Trunks” using the FreePBX GUI to connect two PABX.
We are currently using PJSIP. When Asterisk “user/Endpoint/IP-Phone” receives a call from Cisco CME user-1 via the SIP trunk and then transfers the call back to Cisco CME USer-2, Asterisk “user/Endpoint/IP-Phone” sends a REFER message to Asterisk server.
Asterisk consumes the REFER and sends a Re-INVITE to Cisco CME to complete the transfer process.
We are trying to find out if Asterisk can send the received REFER to Cisco CME instead of Re-INVITE to complete the transfer process?