PROBLEM -: I want to enable Universal Voice feature on my on-premise Adobe connect server using our internal Cisco call manager server.
There are some basic network configurations needed for this integration as -:
1. SIP port 5060 should be opened between Adobe connect and cisco call manager for SIP communications
2. Port range 5000 – 6000 should be opened between Adobe connect and cisco call manager for transversing voice packets via RTP.
3. DTMF mode RFC 2833 must be supported by cisco call manager as well as its carrier.
Integration steps -:
1. I need to create username/password based SIP account on cisco call manager
2. Login into cisco call manager and Navigate to Device -> Phone -> Add new ->
Refer to the screenshot below
Select Third party Basic device
Select Sip profile as standard sip profile
Select Digest user as LDAP ID of a user x
Check mark Media Termination Point Protocol.
3. Navigate to User Management > End User .
Search for LDAP ID of user x
The user-id of the LDAP user x would be the username of SIP account
The pin would be the password of the SIP account.
4. Configure your adobe connect SIP account settings with username of SIP account , password of the SIP account as per step 3.