I've been using asterisk for a while, only for dialout from a SIP client over a PRI -> PSTN, this works great. Now I have a need to also dialin to asterisk over the PRI/TDM, I've been testing by creating an extension, and essentially playing back a recording on that extension. If I access the extension from a SIP client, it works great, very high quality, no chops are stumbles, If I come into the same extension from the PSTN over the PRI into asterisk, the quality degrades significantly, I can hear some noise on answer, and flutters while the message is played. I'm wondering if there are some settings I should try to increase the quality (other then the gain and echo settings in zapata.conf).
Regards MIKE _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
