FreePBX 2.9.0rc1.3 Dahdi Asterisk 1.6 running on 5 servers.
Rhino RCB24 cards for phones (four servers), Rhino R4T1 card for external interface (1 PRI and 2 T1s, plus four SIP phones). The setup is exactly the same as our old PIAF setup based on Asterisk 1.2 and an very old version of FreePBX. In fact, with the exception of the new firmware version on the R4T1 card and new hard disks, the hardware is identical. This firmware is not compatible with the old ZAP drivers, so going back isn’t a possibility on that server.
The systems are all PIAF Bronze (Asterisk 1.6 and DAHDI) installation. Ever since the installation of the new system (installed a new hard drive and installed new software) we’ve been having three problems:
All of our recordings sound like someone is talking through a fan. I was using “wav” format for our call recordings and have switched to “WAV” to see if that helps. I convert the files (using LAME) to mp3 files for long-term storage, and they sound like crap after the conversion. I thought it was a problem with LAME until I listened to one of the WAV versions of the recordings - they are identical. Note that this is happening on our SIP and DAHDI phones, so the problem is consistent across all five servers. We record on the server with the extensions, so all five servers are used for recording.
We have to execute an “asterisk reload” on every server every hour or two or we start to have clicking, popping, and audio drop-out on our outgoing calls (we have no incoming service, just outbound). This drop-out is different - it sounds like an AT&T cell phone call (talking then -silence- then more talking). If I listen in on a barge extension, the “comfort noise” drops out. I can still hear the caller talking, but the people at the remote end cannot. The recordings sound “normal”, in that they still sound like someone talking through a fan, but the audio is all there. I played around with various rx and txgain settings, trying to see if that helped. It doesn’t, except that I can make the call unintelligible through too much or too little gain. This occurs on all calls (not just the DAHDI or SIP Phones).
In small clusters, outgoing calls are terminated before the DAHDI line finishes connecting, which is then picked up by our “operator” extension as if it’s an incoming call. Since we don’t use the inbound for anything, the extension’s mailbox is now full at 100 messages of 42 second dialtone recordings. It seems to happen four or five calls in a row, randomly throughout the day. The callers get dead air (the line has hung up).
Skyking suggested call timing (in another forum), but I don’t really see how that’s possible, each of the servers is connected to one another via IAX trunks.
So, problem 1 might be solved by switching to a different recording format, but I’d really like to get some opinions on that before I start jerking all of the servers around through a list of 7 or 8 file formats.
Problem 2 isn’t “solved” but the workaround is OK until I can figure out the problem. I’m thinking there’s a leak somewhere, but I can’t figure out where to look.
Problem 3 might be related to problem 2, but I can’t say for sure. The problems seem to coincide, but the people on the phones are little more than monkeys that talk. I can’t get any kind of support for fixing the problem from them except for screeching and throwing poop.
Any suggestions would be greatly appreciated.
I’ve included the chan_dahdi.conf file for the R4T1 server below.