Hans Witvliet wrote:
And to: "Asterisk 11.0.0-rc2 Now Available"
skimming through
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2
I did not see any reference towards Motif/XMPP.
So your code is still only in SVN, not in the RC2?
The commits in question:
* [r374850] Fix an issue where outgoing calls would fail to establish
audio due to ICE negotiation failures.
This change removes the requirement for ufrag and pwd in the transport
stanza and also makes us the controlling agent.
* [r374877] Fix a bug where audio on Google Voice would not work due to
ignoring candidates.
Instead of ignoring parts of the message that are not known just
ignore the ones we know may be present and that would cause a problem.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
