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authorGuido Falsi <madpilot@FreeBSD.org>2018-03-25 16:11:53 +0800
committerGuido Falsi <madpilot@FreeBSD.org>2018-03-25 16:11:53 +0800
commit2bfe6d42e0b8261873b25dbaa4d36f3cb60821b1 (patch)
treee1f2903428adfcf326e4b4fc6e3e58d0299c7b95
parentbefdad111898b01066b1fea577830add7063489a (diff)
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Reword option description to include all asterisk versions.
Notes
Notes: svn path=/head/; revision=465504
-rw-r--r--net/pjsip/Makefile2
1 files changed, 1 insertions, 1 deletions
diff --git a/net/pjsip/Makefile b/net/pjsip/Makefile
index ccf5d81476d2..9ecc1b351781 100644
--- a/net/pjsip/Makefile
+++ b/net/pjsip/Makefile
@@ -29,7 +29,7 @@ OPTIONS_DEFINE= AMR DEBUG EXTSRTP FFMPEG G711 G722 G7221 GSM ILBC IPV6 \
SHARED SOUND SPEEX SPEEXAEC V4L VIDEO WEBRTC
OPTIONS_DEFAULT=G711 G722 G7221 GSM ILBC L16 SHARED SPEEX SPEEXAEC
-EXTSRTP_DESC= Use libsrtp port (needed to get SRTP support in asterisk13 pjsip backend)
+EXTSRTP_DESC= Use libsrtp port (needed to get SRTP support in asterisk pjsip backend)
G711_DESC= G.711 codec support
G722_DESC= G.722 codec support
G7221_DESC= G.722.1 codec support