>> Interestingly, that isn't completely true. If it goes out a SIP trunk
>> to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300
>> (where the T1 goes), it has the same problem. This was leading me to
>> believe that the problem was on the 8300.
>Well, that doesn't disprove my statement :-) Note that I said that when the
>audio is directed at endpoints that have a proper jitter buffer, there is no
>issue. If you send the call over SIP to >this 'SV8300' device and still have
>audio issues, that would imply that this device does not have a jitter buffer
>capable of handling this level of jitter.
You can try and improve audio quality and "compensate" on the problems with the
'SV8300' device by using quality improvement software like PBXMate. It should
be able and take care of server-side echo cancellation.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
