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Merged revisions 65683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines Merged revisions 65682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines ensure that variables are set on a newly created channel before we start a PBX on it ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -4289,15 +4289,17 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
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pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
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if (i->rtp)
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ast_jb_configure(tmp, &global_jbconf);
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/* Set channel variables for this call from configuration */
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for (v = i->chanvars ; v ; v = v->next)
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pbx_builtin_setvar_helper(tmp, v->name, v->value);
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if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
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ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
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tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
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ast_hangup(tmp);
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tmp = NULL;
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}
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/* Set channel variables for this call from configuration */
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for (v = i->chanvars ; v ; v = v->next)
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pbx_builtin_setvar_helper(tmp,v->name,v->value);
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if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY))
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append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
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