[asterisk-commits] dvossel: branch 1.6.0 r206947 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: dvossel
Date: Fri Jul 17 11:18:49 2009
New Revision: 206947

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=206947
Log:
Merged revisions 206939 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r206939 | dvossel | 2009-07-17 11:13:22 -0500 (Fri, 17 Jul 2009) | 20 lines
  
  Merged revisions 206938 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r206938 | dvossel | 2009-07-17 11:05:06 -0500 (Fri, 17 Jul 2009) | 14 lines
    
    SIP incorrect From: header information when callpres is prohib
    
    Some ITSP make use of the "Anonymous" display name to detect a
    requirement to withhold caller id across the PSTN. This does
    not work if the display name is "Unknown".
    
    (closes issue #14465)
    Reported by: Nick_Lewis
    Patches:
          chan_sip.c-callerpres.patch uploaded by Nick (license 657)
          chan_sip.c-callerpres_trunk.patch uploaded by dvossel (license 671)
    Tested by: Nick_Lewis, dvossel
  ........
................

Modified:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=206947&r1=206946&r2=206947
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Fri Jul 17 11:18:49 2009
@@ -221,7 +221,8 @@
 #define MAX(a,b) ((a) > (b) ? (a) : (b))
 #endif
 
-#define CALLERID_UNKNOWN        "Unknown"
+#define CALLERID_UNKNOWN             "Anonymous"
+#define FROMDOMAIN_INVALID           "anonymous.invalid"
 
 #define DEFAULT_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
 #define DEFAULT_QUALIFYFREQ          60 * 1000        /*!< Qualification: How often to check for the host to be up */
@@ -9239,6 +9240,7 @@
 	char tmp_l[SIPBUFSIZE/2];	/* build a local copy of 'l' if needed */
 	const char *l = NULL;	/* XXX what is this, exactly ? */
 	const char *n = NULL;	/* XXX what is this, exactly ? */
+	const char *d = NULL;	/* domain in from header */
 	const char *urioptions = "";
 	int ourport;
 
@@ -9264,6 +9266,7 @@
 
 	snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);
 
+	d = S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr));
 	if (p->owner) {
 		l = p->owner->cid.cid_num;
 		n = p->owner->cid.cid_name;
@@ -9273,6 +9276,7 @@
 	    ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)) {
 		l = CALLERID_UNKNOWN;
 		n = l;
+		d = FROMDOMAIN_INVALID;
 	}
 	if (ast_strlen_zero(l))
 		l = default_callerid;
@@ -9299,9 +9303,9 @@
 
 	ourport = ntohs(p->ourip.sin_port);
 	if (!sip_standard_port(p->socket.type, ourport) && ast_strlen_zero(p->fromdomain))
-		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, ast_inet_ntoa(p->ourip.sin_addr), ourport, p->tag);
+		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, d, ourport, p->tag);
 	else
-		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)), p->tag);
+		snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, d, p->tag);
 
 	/* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
 	if (!ast_strlen_zero(p->fullcontact)) {