Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time.
Please paypal your contribution to [EMAIL PROTECTED] today. Every little dollar is worth quite a lot!
I fully trust that Joachim (Zoa) and his team will complete this in a good way and look forward to improved sound quality in the SIP channel.
Read more here: http://www.astertest.com/forum/viewtopic.php?t=13
Thank you for your contribution!
/Olle
If you're going to VON in San José, meet me, Joachim and other Asterisk developers in the Asterisk Pavillion! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
