Hi all,
I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz processor. I am running the latest build of White Box Enterprise Linux.
Our call routing is like this:
SJPHONE on Windows -> QoS-enabled Switch -> Asterisk -> T1 Line -> Broadvoice SIP account -> PSTN
Calls seem to work great from user to user. However, calls from a SJPhone user to the PSTN are not so great. The SJPhone user hears the person on the PSTN perfectly – I mean, completely flawless. However, the user on the PSTN end hears choppy / jittery, extraneous clicks, etc.
Here is the SJPhone config:
Audio Compression: G.711
Driver buffer size: 20 msec
Driver input queue length: 6
Driver output queue length: 4
RTP jitter queue length: 6
Silence Suppression: No
DTMF Sending: RFC 2833
Signal Duration (ms): 270
RTP Payload type: 101
Signal volume: 10
Pause duration (ms): 100
And the sip extension config (in Asterisk Management Portal):
Allow: blank
Canreinvite: no
Disallow: gsm
Dtmfmode: rfc2833
Host: dynamic
Nat: yes (some users are behind NAT)
Qualify: no
Any ideas on what to do to get rid of the choppiness?
Thanks!
Tim
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