I am trying to do test in the laboratory with callmanager 6.1 and I have the following problem: I configured the callmanager using the MGCP Protocol and I am using the directory number of PABX analogic. When I start the conversation the IP Phone (DN 3500 - IP Phone CallManager) to DN 2486 (DN PABX analogic) there is only audio in IP Phone. I verified the configuration CallManager and Gateway and didn`t find any problem. I configured the command voice rtp send-recv, but, it doesn`t solve the problem. Any Idea?
It would be interesting to see if you are recieving any packets on the end points, what is the connectivity between both the phones, i am guessing the same network..
you need to take a sniffer from the IP phone to see if you are recieving the packets or not, even easier; hit the "?" button on the IP phone twice to see if you are recieving or sending the packtes...once the call connects the call manager is out of the picture for rtp flow part..
Can you ping the IP phone from the vlan in which the other phone is; and viceversa?
Every one-way voice problem I have seen has been due to one-way routing - i.e. Phone A can reach Phone B, but Phone B cannot reach Phone A. This will be independent of the call setup which will be between each extension/gateway and Call Manager.
I would check the IP routing path between the two phones as the first thing!
Are you getting this error “Installer User Interface Mode Not Supported. The installer cannot run in this UI mode. To specify the interface mode, use the -i command-line option, followed by the UI mode identifier. The value UI mode identifiers...
The below trick might come handy when you have to add a new node to a cluster but you don't have or is unsure of the security password for the publisher. This procedure has been around for ages.
1) Login into the CLI of the Publisher.