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Add "autoframing" option to sip.conf.sample and h323.conf.sample.
The autoframing option was added to chan_sip.c in r43243 (mogorman, 2006-09-19 01:32:57), but never made its way into the sample configs. Review: https://reviewboard.asterisk.org/r/2768/ ........ Merged revisions 396994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396995 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396996 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -29,6 +29,8 @@ port = 1720
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;allow=gsm ; Always allow GSM, it's cool :)
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;allow=ulaw ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
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; preferences. Defaults to no.
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;
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; User-Input Mode (DTMF)
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;
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@@ -310,6 +310,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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;allow=ulaw ; Allow codecs in order of preference
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;allow=ilbc ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
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; for framing options
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;autoframing=yes ; Set packetization based on the remote endpoint's (ptime)
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; preferences. Defaults to no.
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;
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; This option specifies a preference for which music on hold class this channel
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; should listen to when put on hold if the music class has not been set on the
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@@ -1205,6 +1207,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; language
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; allow
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; disallow
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; autoframing
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; insecure
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; trustrpid
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; progressinband
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