diff --git a/channels/chan_sip.c b/channels/chan_sip.c index dbd3903a34cae7faea929590dc4337fd5ba12101..231f376a8d163b574bdf6472f6decebdd7ec11cd 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -12645,7 +12645,6 @@ 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\" <sip:anonymous@anonymous.invalid>"; struct ast_party_id connected_id; if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) { @@ -12671,12 +12670,11 @@ 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)) { + 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) { - 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, "Privacy", "id"); } - 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 46af790434c6bea82712d6a695aee18eab5b9ed8..2b26589d900735696a7980c53b35bdd3dca26d38 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)!