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res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited

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Per RFC3325, the 'From' header is now anonymized on outgoing calls when
caller id presentation is prohibited.

TID = trust_id_outbound
PRO = Set(CALLERID(pres)=prohib)
USR = endpoint/from_user
DOM = endpoint/from_domain
PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes)

Conditions          |Result
--------------------|----------------------------------------------------
TID PRO USR DOM     |PAI    FROM
--------------------|----------------------------------------------------
Y   Y   abc def.ghi |PRI    "Anonymous" <sip:abc@def.ghi>
Y   Y   abc         |PRI    "Anonymous" <sip:abc@anonymous.invalid>
Y   Y       def.ghi |PRI    "Anonymous" <sip:anonymous@def.ghi>
Y   Y               |PRI    "Anonymous" <sip:anonymous@anonymous.invalid>

Y   N   abc def.ghi |YES    <sip:abc@def.ghi>
Y   N   abc         |YES    <sip:abc@<ip_address>>
Y   N       def.ghi |YES    "Caller Name" <sip:<caller_exten>@def.ghi>
Y   N               |YES    "Caller Name" <sip:<caller_exten>@<ip_address>>

N   Y   abc def.ghi |NO     "Anonymous" <sip:abc@def.ghi>
N   Y   abc         |NO     "Anonymous" <sip:abc@anonymous.invalid>
N   Y       def.ghi |NO     "Anonymous" <sip:anonymous@def.ghi>
N   Y               |NO     "Anonymous" <sip:anonymous@anonymous.invalid>

N   N   abc def.ghi |YES    <sip:abc@def.ghi>
N   N   abc         |YES    <sip:abc@<ip_address>>
N   N       def.ghi |YES    "Caller Name" <sip:<caller_exten>@def.ghi>
N   N               |YES    "Caller Name" <sip:<caller_exten>@<ip_address>>

ASTERISK-25791 #close
Reported-by: Anthony Messina

Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9
This commit is contained in:
George Joseph
2016-02-24 16:25:09 -07:00
parent afef0dc038
commit 2b9849625c
6 changed files with 151 additions and 78 deletions
Download Patch File Download Diff File Expand all files Collapse all files

98
res/res_pjsip_session.c
View File

@@ -30,6 +30,7 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/callerid.h"
#include "asterisk/datastore.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
@@ -800,6 +801,75 @@ static pjmedia_sdp_session *generate_session_refresh_sdp(struct ast_sip_session
return create_local_sdp(inv_session, session, previous_sdp);
}
static void set_from_header(struct ast_sip_session *session)
{
struct ast_party_id effective_id;
struct ast_party_id connected_id;
pj_pool_t *dlg_pool;
pjsip_fromto_hdr *dlg_info;
pjsip_name_addr *dlg_info_name_addr;
pjsip_sip_uri *dlg_info_uri;
int restricted;
if (!session->channel || session->saved_from_hdr) {
return;
}
/* We need to save off connected_id for RPID/PAI generation */
ast_party_id_init(&connected_id);
ast_channel_lock(session->channel);
effective_id = ast_channel_connected_effective_id(session->channel);
ast_party_id_copy(&connected_id, &effective_id);
ast_channel_unlock(session->channel);
restricted =
((ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED);
/* Now set up dlg->local.info so pjsip can correctly generate From */
dlg_pool = session->inv_session->dlg->pool;
dlg_info = session->inv_session->dlg->local.info;
dlg_info_name_addr = (pjsip_name_addr *) dlg_info->uri;
dlg_info_uri = pjsip_uri_get_uri(dlg_info_name_addr);
if (session->endpoint->id.trust_outbound || !restricted) {
ast_sip_modify_id_header(dlg_pool, dlg_info, &connected_id);
}
ast_party_id_free(&connected_id);
if (!ast_strlen_zero(session->endpoint->fromuser)) {
dlg_info_name_addr->display.ptr = NULL;
dlg_info_name_addr->display.slen = 0;
pj_strdup2(dlg_pool, &dlg_info_uri->user, session->endpoint->fromuser);
}
if (!ast_strlen_zero(session->endpoint->fromdomain)) {
pj_strdup2(dlg_pool, &dlg_info_uri->host, session->endpoint->fromdomain);
}
ast_sip_add_usereqphone(session->endpoint, dlg_pool, dlg_info->uri);
/* We need to save off the non-anonymized From for RPID/PAI generation (for domain) */
session->saved_from_hdr = pjsip_hdr_clone(dlg_pool, dlg_info);
/* In chan_sip, fromuser and fromdomain trump restricted so we only
* anonymize if they're not set.
*/
if (restricted) {
/* fromuser doesn't provide a display name so we always set it */
pj_strdup2(dlg_pool, &dlg_info_name_addr->display, "Anonymous");
if (ast_strlen_zero(session->endpoint->fromuser)) {
pj_strdup2(dlg_pool, &dlg_info_uri->user, "anonymous");
}
if (ast_strlen_zero(session->endpoint->fromdomain)) {
pj_strdup2(dlg_pool, &dlg_info_uri->host, "anonymous.invalid");
}
}
}
int ast_sip_session_refresh(struct ast_sip_session *session,
ast_sip_session_request_creation_cb on_request_creation,
ast_sip_session_sdp_creation_cb on_sdp_creation,
@@ -867,6 +937,12 @@ int ast_sip_session_refresh(struct ast_sip_session *session,
}
}
/*
* We MUST call set_from_header() before pjsip_inv_(reinvite|update). If we don't, the
* From in the reINVITE/UPDATE will be wrong but the rest of the messages will be OK.
*/
set_from_header(session);
if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
if (pjsip_inv_reinvite(inv_session, NULL, new_sdp, &tdata)) {
ast_log(LOG_WARNING, "Failed to create reinvite properly.\n");
@@ -1082,6 +1158,7 @@ static pjsip_module session_reinvite_module = {
.on_rx_request = session_reinvite_on_rx_request,
};
void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata,
ast_sip_session_response_cb on_response)
{
@@ -1095,19 +1172,6 @@ void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, session_module.id,
MOD_DATA_ON_RESPONSE, on_response);
if (!ast_strlen_zero(session->endpoint->fromuser) ||
!ast_strlen_zero(session->endpoint->fromdomain)) {
pjsip_fromto_hdr *from = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_FROM, tdata->msg->hdr.next);
pjsip_sip_uri *uri = pjsip_uri_get_uri(from->uri);
if (!ast_strlen_zero(session->endpoint->fromuser)) {
pj_strdup2(tdata->pool, &uri->user, session->endpoint->fromuser);
}
if (!ast_strlen_zero(session->endpoint->fromdomain)) {
pj_strdup2(tdata->pool, &uri->host, session->endpoint->fromdomain);
}
}
handle_outgoing_request(session, tdata);
internal_pjsip_inv_send_msg(session->inv_session, session->endpoint->transport, tdata);
@@ -1133,9 +1197,17 @@ int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data
#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
pjmedia_sdp_neg_set_answer_multiple_codecs(session->inv_session->neg, PJ_TRUE);
#endif
/*
* We MUST call set_from_header() before pjsip_inv_invite. If we don't, the
* From in the initial INVITE will be wrong but the rest of the messages will be OK.
*/
set_from_header(session);
if (pjsip_inv_invite(session->inv_session, tdata) != PJ_SUCCESS) {
return -1;
}
return 0;
}