Good Day Everyone,
Let me better explain the subject line of this post. We are using one of our Asterisk Servers strictly as a conference bridge for about 145 users. The calls range from 4 to 24 participants per call with usually no more than 3 calls going at any given time. The bridge equipment is dual quad core Xeon processors with 3 Gig of RAM running Asterisk 1.4.23 on Redhat latest release. The trunks are Qwest SIP Toll Free for the most part with BellSouth / AT&T local PRI service (Through an Avaya PBX Via Avaya SIP Server) as a backup.
I have one, maybe two, users who continually complain about being “Disconnected” during their conference call. Since they are the moderator they lose the conference call. I have fixed that by simply having them get another person to be the moderator. This also helped me determine that it was not the bridge that was “Disconnecting” them. Additionally, we discovered that they weren’t being disconnected. They would just pop in to one-way audio. They could not be heard but they could hear everyone else. So with this only happening to two users I didn’t think much of it and continued to troubleshoot. I had the main complaining user dial in via Toll Free and then via the BellSouth local lines from both his cell phone and land line with the same result on each. The REALLY strange part is that while I was talking to this user on my SIP phone from a different Asterisk server, he popped in to one-way audio about 2 minutes in to our conversation using his cell phone. So now I am thinking that it is his carrier that is not doing something right… However, the more I think about this I see that the common link between his conference call and his call to me is that all traffic comes through the Avaya SIP server for call accounting purposes (eCAS is the company standard so calls have to go through the Avaya until I get Asterisk posting to eCAS.) However, why would this only happen to one or two users out of 145 users? What I want to do is to tell him to get a conference card from another company and call it a day but I know I can’t do that and keep a happy user.
So does anyone have ideas of what I could look for? The SIP debug logs only show him disconnect the call. At the moment he goes in to one way audio I see no debug messages.
Here is a typical path of a call.
Call comes in from Qwest SIP to the Asterisk Server. Asterisk Server puts the call over the Avaya SIP Trunk to a VDN and vector that plays a greeting and asks the user to enter their conference number. Call goes back to the Asterisk server to the conference number and the user enteres the pass code. Conference begins. So there are 3 SIP sessions for each caller in a conference.
So I have had NO problems with 143 users. Just one or two.
Any thoughts on this? I appreciate anything I can look at…