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Commit 1d1cc62f authored by Jonathan Rose's avatar Jonathan Rose
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Reverting r411189 so that it can be put up for public review

```yaml:frontmatter
  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)
```
........

Merged revisions 412328 from http://svn.asterisk.org/svn/asterisk/branches/1.8


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
parent 6bf7f01a
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......@@ -12503,6 +12503,7 @@ 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;
......@@ -12527,11 +12528,12 @@ 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");
 
......@@ -1414,8 +1414,7 @@ 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)!
......
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