Hi Guys,
We are upgrading to new servers to I decided to take the new Thirdlane ISO for a spin. We installed Asterisk 1.6.0.6 that comes with the ISO and did a complete auto setup and so far I am very happy with everything).
I only have a few tenants running in MTE and I have noticed that on Conference Calls users who are speaking seem to sound like a skipping records at times (much like the 1980's Max Hedrom). To my astonishment this is all happening even on low latency internal connections.
I had 3 people dial into a conference internally (via a dedicated VoIP LAN we have) and within a few seconds of joining we have people sounding like the old 1980's MAx Hedrom with skipping or repeating audio.
The PBX only had 3 calls on it. Its running 8 Procs and 8GB of RAM. It has 2 bonded and dedicated Intel Server Pro Series GB Ethernet Cards. We have a dedicated 10GB LAN backbone and nothing seems to be stressed.
Whats weird is that no matter the load on the system we seem to get the same issue. This issue doesn't seem to be on any calls, but with the conference bridge lines only.
Is there anything I should know of related things like this on Asterisk 1.6.0.6 (I know eeman and erik HATE asterisk 1.6)? Should I just roll back to a 1.4.x.x version instead (we don't have any of these issues on our current 1.4.x.x systems on low end hardware).
If I roll back is there anything I will lose on features? Thanks in advance for you assistance and as always I look forward to the responses.
vr,
Fixed!
Hi Everyone,
This always happens once I post here. I was messing with some settings and I noticed that the conference bridge was recording all the calls coming in. So I disabled the call recording for the line and the problem went away.
Its seems there is an issue with the recording features that causes the jitter/voice quality to skip here and there. Does anyone else have or has had this issue? If so how was it fixed as we would like to record calls in the future to keep meetings minutes of Stock holder meetings and customer meeting where we are doing requirements gathering.
Its a great feature and I would hate to have to banish it forever.
vr,
Timothy
conference room
I have the same issue...
How do we verify the timing source... we use all SIP trunks so we have no line cards that do timing.
Regards,
Andrew
dahdi drivers are loaded and
dahdi drivers are loaded and installed? If you upgrade dahsi you must do re-compile of asterisk all the way down from 'make distclean'.
Sorry guys,
I made a mistake when I quoted my versions:
OLD Asterisk System not having this issue: Asterisk 1.6.0.6
NEW Asterisk system having this issue: Asterisk 1.6.2.11
I guess the answer seems simple to me. Roll back to the 1.6.0.6 edition and not have to worry about the issue or reinstalling Thirdlane Webmin Plugin (or worse backup and restore!).
Sorry again for the misquoting my versions. As always suggestions are appreciated.
vr,
Timothy