[asterisk-commits] rmudgett: branch 1.8 r312866 - /branches/1.8/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: rmudgett
Date: Tue Apr  5 10:38:14 2011
New Revision: 312866

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=312866
Log:
Responding to OPTIONS packet with 404 because Asterisk not looking for "s" extension.

The get_destination() function was not using the "s" extension when the
request URI did not specify an extension.  This is a regression caused
when the URI parsing code was extracted into parse_uri().

Made get_destination() substitute the "s" extension when the parsed URI
results in an empty string.

(closes issue #18348)
Reported by: shmaize
Patches:
      issue18348_v1.8.patch uploaded by rmudgett (license 664)
Tested by: shmaize

Modified:
    branches/1.8/channels/chan_sip.c

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=312866&r1=312865&r2=312866
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Tue Apr  5 10:38:14 2011
@@ -14398,16 +14398,20 @@
 	return 0;
 }
 
-/*! \brief Find out who the call is for.
-	We use the request uri as a destination.
-	This code assumes authentication has been done, so that the
-	device (peer/user) context is already set.
-	\return 0 on success (found a matching extension), non-zero on failure
-
-  \note If the incoming uri is a SIPS: uri, we are required to carry this across
-	the dialplan, so that the outbound call also is a sips: call or encrypted
-	IAX2 call. If that's not available, the call should FAIL.
-*/
+/*!
+ * \brief Find out who the call is for.
+ *
+ * \details
+ * We use the request uri as a destination.
+ * This code assumes authentication has been done, so that the
+ * device (peer/user) context is already set.
+ *
+ * \return 0 on success (found a matching extension), non-zero on failure
+ *
+ * \note If the incoming uri is a SIPS: uri, we are required to carry this across
+ * the dialplan, so that the outbound call also is a sips: call or encrypted
+ * IAX2 call. If that's not available, the call should FAIL.
+ */
 static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id)
 {
 	char tmp[256] = "", *uri, *domain, *dummy = NULL;
@@ -14433,6 +14437,14 @@
 
 	SIP_PEDANTIC_DECODE(domain);
 	SIP_PEDANTIC_DECODE(uri);
+	if (ast_strlen_zero(uri)) {
+		/*
+		 * Either there really was no extension found or the request
+		 * URI had encoded nulls that made the string "empty".  Use "s"
+		 * as the extension.
+		 */
+		uri = "s";
+	}
 
 	ast_string_field_set(p, domain, domain);