[asterisk-commits] jrose: trunk r331371 - in /trunk: CHANGES channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: jrose
Date: Wed Aug 10 10:45:57 2011
New Revision: 331371

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=331371
Log:
SIP display-name needed to be empty for Avaya IP500

In order to address a compatability issue with certain features on certain devices
which rely on display name content to change behavior, initreqprep in chan_sip.c
has been changed to no longer substitute cid_number into the display name when
cid_name isn't present.  Instead, it will send no display name in that case.

(closes issue ASTERISK-16198)
Reported by: Walter Doekes

Review: https://reviewboard.asterisk.org/r/1341/



Modified:
    trunk/CHANGES
    trunk/channels/chan_sip.c

Modified: trunk/CHANGES
URL: http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=diff&rev=331371&r1=331370&r2=331371
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Wed Aug 10 10:45:57 2011
@@ -7,6 +7,17 @@
 === and the other UPGRADE files for older releases.
 ===
 ==============================================================================
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
+------------------------------------------------------------------------------
+
+SIP Changes
+-----------
+ * Asterisk will no longer substitute CID number for CID name into display
+   name field if CID number exists without a CID name. This change improves
+   compatibility with certain device features such as Avaya IP500's directory
+   lookup service.
 
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=331371&r1=331370&r2=331371
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Aug 10 10:45:57 2011
@@ -11792,6 +11792,8 @@
 	const char *d = NULL;	/* domain in from header */
 	const char *urioptions = "";
 	int ourport;
+	int cid_has_name = 1;
+	int cid_has_num = 1;
 
 	if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) {
 	 	const char *s = p->username;	/* being a string field, cannot be NULL */
@@ -11838,10 +11840,15 @@
 		l = p->mwi_from;
 	}
 
-	if (ast_strlen_zero(l))
+	if (ast_strlen_zero(l)) {
+		cid_has_num = 0;
 		l = default_callerid;
-	if (ast_strlen_zero(n))
+	}
+	if (ast_strlen_zero(n)) {
+		cid_has_name = 0;
 		n = l;
+	}
+
 	/* Allow user to be overridden */
 	if (!ast_strlen_zero(p->fromuser))
 		l = p->fromuser;
@@ -11862,10 +11869,20 @@
 	}
 
 	ourport = (p->fromdomainport) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
+
+	/* If a caller id name was specified, add a display name. */
+	if (cid_has_name || !cid_has_num) {
+		snprintf(from, sizeof(from), "\"%s\" ", n);
+	} else {
+		from[0] = '\0';
+	}
+
 	if (!sip_standard_port(p->socket.type, ourport)) {
-		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, d, ourport, p->tag);
+		size_t offset = strlen(from);
+		snprintf(&from[offset], sizeof(from) - offset, "<sip:%s@%s:%d>;tag=%s", l, d, ourport, p->tag);
 	} else {
-		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, d, p->tag);
+		size_t offset = strlen(from);
+		snprintf(&from[offset], sizeof(from) - offset, "<sip:%s@%s>;tag=%s", l, d, p->tag);
 	}
 
 	if (!ast_strlen_zero(explicit_uri)) {