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automerge commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@18297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -11658,12 +11658,13 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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format &= ((AST_FORMAT_MAX_AUDIO << 1) - 1);
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if (!format) {
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ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(global_capability));
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*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */
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return NULL;
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}
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p = sip_alloc(NULL, NULL, 0, SIP_INVITE);
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if (!p) {
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ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory)\n", (char *)data);
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*cause = AST_CAUSE_CONGESTION;
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*cause = AST_CAUSE_SWITCH_CONGESTION;
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return NULL;
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}
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@@ -11671,7 +11672,7 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
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if (!p->options) {
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sip_destroy(p);
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ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
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*cause = AST_CAUSE_CONGESTION;
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*cause = AST_CAUSE_SWITCH_CONGESTION;
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return NULL;
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}
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