[OpenSIPS-Users] Call_center module with FreeSwitch as media

[Артем Друзь]

Hello.
I'm trying to configure Call_center module in OpenSIPS 2.4.7 with
FreeSwith as media server.
I have configured flow and agent. Agent "testoper" (with URI
101 at example.local) successfully logged in to the flow "techsupport"
with fifo command:
root at os1:~# opensipsctl fifo cc_agent_login testoper 1
root at os1:~# opensipsctl fifo cc_list_agents
Agent:: testoper Ref=0 Loged in=YES State=free
root at os1:~# opensipsctl fifo cc_list_flows
Flow:: techsupport Avg Call Duration=0 Processed Calls=0 Logged
Agents=1 Ongoing Calls=0 Ref=0

When I try to call from SIP user 103 at example.local to number 777 (it
configured in dialplan table for modify $rU to "QUEUE_techsupport") I
hear the "message_welcome" (while 183) and get then 480. Call is
rejecting and not send to the active and free agent of flow after
that.
As I see in SIP dump - 480 came from FreeSwitch.

Links to Pastebin:
Some dumps from DB, configs OpenSIPS and FreeSwitch and simple log:
https://pastebin.com/2x39D0Pc
Debug log: https://pastebin.com/Sc7j5MDM

Can you help me with configuration of call_center module or tell me
how must be configured FreeSwitch for correct work with OpenSIPS?

P.S. Sorry for my English.