[asterisk-commits] jpeeler: branch 1.6.2 r274347 - in /branches/1.6.2: ./ configs/sip.conf.sample
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jul 6 17:30:10 CDT 2010
Author: jpeeler
Date: Tue Jul 6 17:30:06 2010
New Revision: 274347
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=274347
Log:
Merged revisions 274316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r274316 | jpeeler | 2010-07-06 17:23:35 -0500 (Tue, 06 Jul 2010) | 14 lines
Merged revisions 274283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r274283 | jpeeler | 2010-07-06 17:15:21 -0500 (Tue, 06 Jul 2010) | 7 lines
Correct sip.conf.sample comments for prematuremedia option.
(closes issue #17513)
Reported by: festr
Patches:
patch uploaded by festr (license 443)
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Modified:
branches/1.6.2/ (props changed)
branches/1.6.2/configs/sip.conf.sample
Propchange: branches/1.6.2/
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Binary property 'trunk-merged' - no diff available.
Modified: branches/1.6.2/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/configs/sip.conf.sample?view=diff&rev=274347&r1=274346&r2=274347
==============================================================================
--- branches/1.6.2/configs/sip.conf.sample (original)
+++ branches/1.6.2/configs/sip.conf.sample Tue Jul 6 17:30:06 2010
@@ -215,12 +215,14 @@
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
-;prematuremedia=no ; Some ISDN links send empty media frames before
- ; the call is in ringing or progress state. The SIP
- ; channel will then send 183 indicating early media
- ; which will be empty - thus users get no ring signal.
- ; Setting this to "no" will stop any media before we have
- ; call progress. Default is "yes".
+;prematuremedia=no ; Some ISDN links send empty media frames before
+ ; the call is in ringing or progress state. The SIP
+ ; channel will then send 183 indicating early media
+ ; which will be empty - thus users get no ring signal.
+ ; Setting this to "yes" will stop any media before we have
+ ; call progress (meaning the SIP channel will not send 183 Session
+ ; Progress for early media). Default is "yes". Also make sure that
+ ; the SIP peer is configured with progressinband=never.
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
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