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author | Guido Falsi <madpilot@FreeBSD.org> | 2018-03-25 16:11:53 +0800 |
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committer | Guido Falsi <madpilot@FreeBSD.org> | 2018-03-25 16:11:53 +0800 |
commit | 2bfe6d42e0b8261873b25dbaa4d36f3cb60821b1 (patch) | |
tree | e1f2903428adfcf326e4b4fc6e3e58d0299c7b95 | |
parent | befdad111898b01066b1fea577830add7063489a (diff) | |
download | freebsd-ports-2bfe6d42e0b8261873b25dbaa4d36f3cb60821b1.tar.gz freebsd-ports-2bfe6d42e0b8261873b25dbaa4d36f3cb60821b1.tar.zst freebsd-ports-2bfe6d42e0b8261873b25dbaa4d36f3cb60821b1.zip |
Reword option description to include all asterisk versions.
Notes
Notes:
svn path=/head/; revision=465504
-rw-r--r-- | net/pjsip/Makefile | 2 |
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 |