My employer recently purchased a private company that uses a Cisco phone system. I am trying to support this system, but I have no experience with VOIP. In previous post concerning one way audio the primary issue seemed to be with routing. This may be my problem as well.
I am using IP communicator to connect to the call manager from our home campus and everything works fine until I try to make a call that uses the PSTN lines. However, users at the remote campus can dial out with no problems.
I set up a second laptop with IP communicator on the same switch as me and I can dial it with no problems. This leads me to believe something on the connection between our campuses is blocking the audio, but I can successfully dial users at the remote campus. I just can't dial from this campus to an outside number.
I just did a packet capture on my workstation while making an outside call. I noticed that all the RTP packets leaving my machine are G.729, but the voice-gateway is replying G.711. Could this have anything to do with the audio problem?
Okay. This seems to be a "stupid" issue. Under The Advanced Audio Settings for IP Communicator I have to set the mode to headset. Once I do this I have two way audio. However, it only works for one call before it resets itself to handset. Also, using the headset button dial pad doesn't seem to have an effect like changing the mode does.
Is this something callmanger pushes to the softphone?
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