I propose to establish a SIP trunk between asterisk and CUCM, making sure that both calling and called party information (name and number) passes through, as well as dtmf, correct codec selection, etc
Relevant Skills and Experience
I've been working with both CUCM and asterisk for over 10 years. I have interconnected them numerous times, for both regular calls or special functions (asterisk as voicemail or conference platform).
Proposed Milestones
₹1500 INR - SIP interconnection
Je parle français, si cela est plus pratique pour vous.