[Asterisk-code-review] res pjsip: Re-use IP version of signaling (SIP) for media (R... (asterisk[13])
George Joseph
asteriskteam at digium.com
Mon Sep 12 09:20:42 CDT 2016
George Joseph has posted comments on this change.
Change subject: res_pjsip: Re-use IP version of signaling (SIP) for media (RTP).
......................................................................
Patch Set 3: Code-Review-1
> Build failed (gate pipeline). Please check the logs referenced
> below. For more information on how to proceed, please see
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> - https://jenkins.asterisk.org/jenkins/job/gate-asterisk-channel-drivers/2354/
> : UNSTABLE in 1h 11m 51s
Several of the ipv6 tests tests fail because audio isn't detected on the channel.
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Gerrit-MessageType: comment
Gerrit-Change-Id: I01a85a8c6723fcc12e86139f80e090e2078d04bb
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-HasComments: No
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