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Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 14
authorJoshua Colp <jcolp@digium.com>
Wed, 30 Nov 2016 19:23:44 +0000 (13:23 -0600)
committerGerrit Code Review <gerrit2@gerrit.digium.api>
Wed, 30 Nov 2016 19:23:44 +0000 (13:23 -0600)

Trivial merge
Mirror of https://github.com/asterisk/asterisk.git