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authorsobomax <sobomax@FreeBSD.org>2005-05-03 21:58:26 +0800
committersobomax <sobomax@FreeBSD.org>2005-05-03 21:58:26 +0800
commit92830d644a2299543682d0c009f0e6a898f33700 (patch)
tree02bf63be943e88d759ef5fb9ab9c55598a8f97cf /net/asterisk16/Makefile
parent9b6fb998f96e80cebf98cb05d2d88992beb4f7ef (diff)
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o chan_sip.c:
- Improve codec negotiation logic. - make sure to parse SDP on re-INVITE. o chan_zap.c: - If unanswered Zap channnel is hanged up wait until the calling party in turn hangs up (by detecting ring timeout). Otherwise next ring will trigger a "new" incoming call. This appears to be design limitation of the analogue telephone system - there is no way to reject the call without answering it first. Sponsored by: Porta Software Ltd, PBXpress Inc
Diffstat (limited to 'net/asterisk16/Makefile')
-rw-r--r--net/asterisk16/Makefile2
1 files changed, 1 insertions, 1 deletions
diff --git a/net/asterisk16/Makefile b/net/asterisk16/Makefile
index 00cb6ce94061..2ef12fe3a651 100644
--- a/net/asterisk16/Makefile
+++ b/net/asterisk16/Makefile
@@ -7,7 +7,7 @@
PORTNAME= asterisk
PORTVERSION= 1.0.7
-PORTREVISION= 3
+PORTREVISION= 4
CATEGORIES= net
MASTER_SITES= ftp://ftp.asterisk.org/pub/telephony/asterisk/ \
ftp://ftp.asterisk.org/pub/telephony/asterisk/old-releases/