diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 24a9370a716005c4a972b750b61ed6316e954607..cf71ac45325e549a5ab6edbef9b3708ab6e87151 100644 --- 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 2b26589d900735696a7980c53b35bdd3dca26d38..46af790434c6bea82712d6a695aee18eab5b9ed8 100644 --- 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)!