[asterisk-users] Call quality issue across VPN-> POTS vs SIP

[Bob Pierce]

On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote:
> It's conceivable, but how would I verify this and how would I change
> it if that was the problem?

There's a few things you can do here.
1) Check the sip.conf on both sides to see what is defined there for the
trunk. Look for some disallow and allow statements. If they are there,
that will tell Asterisk what codecs to use on that trunk.

2) You could also check the codec that is in use during a call by
looking at the sip channel. From the asterisk CLI, start with "show
channel SIP/" and tab it out to complete the command showing the trunk
between your two systems. I believe the codecs are listed here as
"NativeFormats" and "ReadFormat". You could check this under both of
your scenarios to see if there is a different codec in use.

3) If you'd like to try and force the use of a compressed codec such as
GSM between your two sites, you would just need to make sure that both
sides had the following lines in the definition for the trunk in
sip.conf and then do a 'reload chan_sip.so" from the Asterisk CLI:
disallow=all
allow=gsm