CUCM 7.1 => OPensips+Freeswitch SIP TRUNK no busy tone
we set up a sip trunk between a Cisco Call Manager 7.1 and a SIP telephony infrastructure (Opensips as balancer + Freeswitch as media server).
We are experiencing a strange issue on the sip trunk: when we call a busy SIP phone (currently on a call, registered to Opensips + Freeswitch) from an SCCP Phone (registered on CUCM) the call is dropped on the SCCP PHone without any indication (busy tone on audio or message on display).
The opensips sends the busy 486 message to CUCM that correctly send the ack in response.
Everything else seems to be fine normal call are correctly completed, transfers are ok.
We tried to disable the MTP without success, MRGL is correctly configured on SIP trunk. Ports are ok.
Can you please help in debugging?
Please find below a trace captured on SIP OPensips balancer, here you are the hosts involved:
10.100.0.70 is OpenSips balancer IP
172.22.72.2 is CUCM 7.1 server IP
9013 is the SIP Phone registered on Opensips + freeswitch system
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