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Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -877,6 +877,9 @@ static struct ast_channel *wait_for_winner(struct findme_user_listptr *findme_us
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* the caller.
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*/
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break;
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case AST_CONTROL_PVT_CAUSE_CODE:
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ast_indicate_data(caller, f->subclass.integer, f->data.ptr, f->datalen);
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break;
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case -1:
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ast_verb(3, "%s stopped sounds\n", ast_channel_name(winner));
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break;
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