Reverting r411189 so that it can be put up for public review

---
  r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines

  chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325)

  Prior to this patch, the P-Asserted-Identity header would include anonymous
  caller id information which seems to go against the point of the
  P-Asserted-Identity header. Now the real caller ID information will be
  included in this header. Also, no privacy header would be included.
  This patch adds 'Privacy: id' to outgoing SIP messages that include the
  P-Asserted-Identity header.

  (closes issue AST-1301)
---
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Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 412329 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 412330 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412331 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 24a9370..cf71ac4 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12631,6 +12631,7 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
        const char *privacy = NULL;
        const char *screen = NULL;
        struct ast_party_id connected_id;
+       const char *anonymous_string = "\"Anonymous\" <sip:anonymous@anonymous.invalid>";
 
        if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
                return 0;
@@ -12655,11 +12656,12 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p)
        lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);
 
        if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
-               ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
-               add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
                if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
-                       add_header(req, "Privacy", "id");
+                       ast_str_set(&tmp, -1, "%s", anonymous_string);
+               } else {
+                       ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
                }
+               add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
        } else {
                ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");
 

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 2b26589..46af790 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -1431,8 +1431,7 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
 ;allow=g729                      ; Pass-thru only unless g729 license obtained
 ;callingpres=allowed_passed_screen ; Set caller ID presentation
-                                 ; See function CALLERPRES documentation for possible
-                                 ; values.
+                                 ; See README.callingpres for more information
 
 ;[xlite1]
 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!