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

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---
  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
This commit is contained in:
Jonathan Rose
2014-04-15 16:13:35 +00:00
parent c87f8a599b
commit cc4a0a7fc9
2 changed files with 6 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
channels/chan_sip.c
View File

@@ -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");

3
configs/sip.conf.sample
View File

@@ -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)!