diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d7c63502e0..a26609aed4 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -12667,7 +12667,6 @@ static int add_rpid(struct sip_request *req, struct sip_pvt *p) const char *fromdomain; const char *privacy = NULL; const char *screen = NULL; - const char *anonymous_string = "\"Anonymous\" "; struct ast_party_id connected_id; if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) { @@ -12693,12 +12692,11 @@ 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)) { - if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) { - ast_str_set(&tmp, -1, "%s", anonymous_string); - } else { - ast_str_set(&tmp, -1, "\"%s\" ", lid_name, lid_num, fromdomain); - } + ast_str_set(&tmp, -1, "\"%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"); + } } else { ast_str_set(&tmp, -1, "\"%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 d46d468fea..54ae05d9c2 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -1431,7 +1431,8 @@ 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 README.callingpres for more information + ; See function CALLERPRES documentation for possible + ; values. ;[xlite1] ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!