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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
