[asterisk-dev] Speex codec in asterisk 1.4.2

Jure Petrovic fonz at siol.net
Wed Apr 4 23:55:14 MST 2007


I just upgraded my system from 1.2.10 to 1.4.2
Using speex, sound is totally garbled and destroyed.
My calling party says he understands me ok.

In 1.2.10 speex codec worked ok. As a SIP client I am using ekiga with
narrowband speex (8000bps) enabled.

Were there any changes in timing mechanism?
Any ideas?

Jure Petrovic

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