Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 13

author zuul <zuul@gerrit.asterisk.org>

Wed, 30 Nov 2016 16:48:13 +0000 (10:48 -0600)

committer Gerrit Code Review <gerrit2@gerrit.digium.api>

Wed, 30 Nov 2016 16:48:13 +0000 (10:48 -0600)
author zuul <zuul@gerrit.asterisk.org>
Wed, 30 Nov 2016 16:48:13 +0000 (10:48 -0600)
committer Gerrit Code Review <gerrit2@gerrit.digium.api>
Wed, 30 Nov 2016 16:48:13 +0000 (10:48 -0600)
diff --cc include/asterisk/channel.h
Simple merge
diff --cc main/channel.c
Simple merge
author	zuul <zuul@gerrit.asterisk.org>
	Wed, 30 Nov 2016 16:48:13 +0000 (10:48 -0600)
committer	Gerrit Code Review <gerrit2@gerrit.digium.api>
	Wed, 30 Nov 2016 16:48:13 +0000 (10:48 -0600)
		1	2
include/asterisk/channel.h	patch \|	diff1 \|	diff2 \|	blob \| history
main/channel.c	patch \|	diff1 \|	diff2 \|	blob \| history