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@example.local) successfully logged in to the flow "techsupport" with fifo command: root@os1:~# opensipsctl fifo cc_agent_login testoper 1 root@os1:~# opensipsctl fifo cc_list_agents Agent:: testoper Ref=0 Loged in=YES State=free root@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@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. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users