[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
 > https://wiki.asterisk.org/wiki/display/AST/Continuous+Integration
 > 
 > - 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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