From: Jeff Peeler Date: Tue, 6 Jul 2010 22:30:06 +0000 (+0000) Subject: Merged revisions 274316 via svnmerge from X-Git-Tag: 1.6.2.11-rc1~51 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=0257ac0bf36bf19253109f7e839f1c0f7ea5b79a;p=thirdparty%2Fasterisk.git Merged revisions 274316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ 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) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@274347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 0b099a496e..e47d96f827 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -215,12 +215,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;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