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default 'sipstorecause' to no
We've decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information: http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html (issue AST-580) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@333009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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2
CHANGES
2
CHANGES
@@ -44,7 +44,7 @@ SIP Changes
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* Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
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* Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
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response. This permits the master channel to know how each channel dialled
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response. This permits the master channel to know how each channel dialled
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in a multi-channel setup resolved in an individual way. This carries a
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in a multi-channel setup resolved in an individual way. This carries a
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performance penalty and can be disabled in sip.conf using the
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performance penalty and must be enabled in sip.conf using the
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'storesipcause' option.
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'storesipcause' option.
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* Added 'externtcpport' and 'externtlsport' options to allow custom port
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* Added 'externtcpport' and 'externtlsport' options to allow custom port
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configuration for the externip and externhost options when tcp or tls is used.
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configuration for the externip and externhost options when tcp or tls is used.
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@@ -20,6 +20,10 @@
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From 1.6.2 to 1.8:
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From 1.6.2 to 1.8:
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* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
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This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
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This carries a performance penalty.
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* Asterisk now requires libpri 1.4.11+ for PRI support.
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* Asterisk now requires libpri 1.4.11+ for PRI support.
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* A couple of CLI commands in res_ais were changed back to their original form:
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* A couple of CLI commands in res_ais were changed back to their original form:
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@@ -221,7 +221,7 @@
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#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
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#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
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#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
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#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
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#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
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#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
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#define DEFAULT_STORE_SIP_CAUSE TRUE /*!< Store HASH(SIP_CAUSE,<channel name>) for channels by default */
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#define DEFAULT_STORE_SIP_CAUSE FALSE /*!< Don't store HASH(SIP_CAUSE,<channel name>) for channels by default */
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#endif
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#endif
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/*@}*/
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/*@}*/
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@@ -1007,14 +1007,13 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;----------------------------- SIP_CAUSE reporting ---------------------------------
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;----------------------------- SIP_CAUSE reporting ---------------------------------
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; storesipcause = yes ; This option causes chan_sip to set the
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; storesipcause = no ; This option causes chan_sip to set the
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; HASH(SIP_CAUSE,<channel name>) channel variable
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; HASH(SIP_CAUSE,<channel name>) channel variable
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; to the value of the last sip response.
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; to the value of the last sip response.
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; WARNING: enabling this option carries a
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; WARNING: enabling this option carries a
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; significant performance burden. It should only
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; significant performance burden. It should only
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; be used in low call volume situations. For
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; be used in low call volume situations. This
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; historical reasons, this option defaults to
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; option defaults to "no".
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; "yes".
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;-----------------------------------------------------------------------------------
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;-----------------------------------------------------------------------------------
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