Hello, It is my understanding that on the softphone side, asterisk is only responsible for establishing the session between two phones. If this is the case, does it matter what type of audio codecs the two phones are using? And if it does matter, are there any codecs that cause problems with asterisk bridging two SIP connections? Thanks for your helpful input,
Daniel _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
