I’m running FreePBX Distro with a PolyCom SoundPoint IP 450 and we are seeing an issue with conferencing. When the second caller is bridged into a local conference, on the Polycom phone, we can no longer hear the first caller. Has anyone else seen this before with Asterisk 11 and Polycom phones?
How many total callers are you trying to conference?
If you are using the phones conference function (which is misleading IMHO) I think you can only have 3 members on the call. The reason for this is that if you are using the conference function of the phone the phone is combining the audio streams and the phone does not have the resources to mix more than 3 voice streams.
For more then 3 callers us a meeting room.
I’m just trying to conference 3 members (me, and two other callers). Its for those “Hey, let me add so-and-so to the call” moments.
Please elaborate on the exact procedure you are using to initiate the conference. I do this all the time and don’t have any issues.
Alan, I isolated the issue. It is a failure of the Polycom to handle conferencing when using SRTP. I need to spend a little more time debugging the SIP conversation.
You’ll need to set encryption=yes in the Asterisk extension for your Polycom.
Once you have configured your Polycom phone to offer SRTP then perform the following:
- Place a call to a one number.
- Hit conference and call a second number.
- Merge the calls and you will only hear the most recently called party.