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support optional sending of Remote-Party-ID headers (issue #2471, heavily modified to actually work properly)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Kevin P. Fleming 20 years ago
parent 266529a503
commit 8af7725f98
2 changed files with 113 additions and 5 deletions
Show Stats Download Patch File Download Diff File

116
channels/chan_sip.c
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@ -546,6 +546,8 @@ struct sip_auth {
/* Call states */
#define SIP_CALL_ONHOLD (1 << 28)
#define SIP_CALL_LIMIT (1 << 29)
/* Remote Party-ID Support */
#define SIP_SENDRPID (1 << 30)
/* a new page of flags for peer */
#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0)
@ -622,6 +624,8 @@ static struct sip_pvt {
char fullcontact[128]; /* The Contact: that the UA registers with us */
char accountcode[AST_MAX_ACCOUNT_CODE]; /* Account code */
char our_contact[256]; /* Our contact header */
char *rpid; /* Our RPID header */
char *rpid_from; /* Our RPID From header */
char realm[MAXHOSTNAMELEN]; /* Authorization realm */
char nonce[256]; /* Authorization nonce */
char opaque[256]; /* Opaque nonsense */
@ -2074,6 +2078,13 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
p->registry->call = NULL;
ASTOBJ_UNREF(p->registry,sip_registry_destroy);
}
if (p->rpid)
free(p->rpid);
if (p->rpid_from)
free(p->rpid_from);
/* Unlink us from the owner if we have one */
if (p->owner) {
if (lockowner)
@ -3013,7 +3024,7 @@ static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int useg
build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain);
else
ast_copy_string(p->callid, callid, sizeof(p->callid));
ast_copy_flags(p, (&global_flags), SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_DTMF | SIP_REINVITE | SIP_PROG_INBAND | SIP_OSPAUTH);
ast_copy_flags(p, (&global_flags), SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | SIP_PROG_INBAND | SIP_OSPAUTH);
/* Assign default music on hold class */
strcpy(p->musicclass, global_musicclass);
p->capability = global_capability;
@ -4482,6 +4493,90 @@ static void build_contact(struct sip_pvt *p)
snprintf(p->our_contact, sizeof(p->our_contact), "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip));
}
/*--- build_rpid: Build the Remote Party-ID & From using callingpres options ---*/
static void build_rpid(struct sip_pvt *p)
{
int send_pres_tags = 1;
const char *privacy;
const char *screen;
char buf[256];
const char *clid = default_callerid;
const char *clin = NULL;
char iabuf[INET_ADDRSTRLEN];
const char *fromdomain;
if (p->rpid || p->rpid_from)
return;
if (p->owner && p->owner->cid.cid_num) {
clid = strdup(p->owner->cid.cid_num);
}
if (p->owner && p->owner->cid.cid_name) {
clin = strdup(p->owner->cid.cid_name);
}
if (!clin || ast_strlen_zero(clin))
clin = clid;
switch (p->callingpres) {
case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED:
privacy = "off";
screen = "no";
break;
case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN:
privacy = "off";
screen = "pass";
break;
case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN:
privacy = "off";
screen = "fail";
break;
case AST_PRES_ALLOWED_NETWORK_NUMBER:
privacy = "off";
screen = "yes";
break;
case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED:
privacy = "full";
screen = "no";
break;
case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN:
privacy = "full";
screen = "pass";
break;
case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN:
privacy = "full";
screen = "fail";
break;
case AST_PRES_PROHIB_NETWORK_NUMBER:
privacy = "full";
screen = "fail";
break;
case AST_PRES_NUMBER_NOT_AVAILABLE:
send_pres_tags = 0;
break;
default:
ast_log(LOG_WARNING, "Unsupported callingpres (%d)\n", p->callingpres);
if ((p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED)
privacy = "full";
else
privacy = "off";
screen = "no";
break;
}
fromdomain = ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain;
snprintf(buf, sizeof(buf), "\"%s\" <sip:%s@%s>", clin, clid, fromdomain);
if (send_pres_tags)
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), ";privacy=%s;screen=%s", privacy, screen);
p->rpid = strdup(buf);
snprintf(buf, sizeof(buf), "\"%s\" <sip:%s@%s>;tag=as%08x", clin,
ast_strlen_zero(p->fromuser) ? clid : p->fromuser,
fromdomain, p->tag);
p->rpid_from = strdup(buf);
}
/*--- initreqprep: Initiate new SIP request to peer/user ---*/
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod)
{
@ -4601,7 +4696,12 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
add_header(req, "Via", p->via);
/* SLD: FIXME?: do Route: here too? I think not cos this is the first request.
* OTOH, then we won't have anything in p->route anyway */
add_header(req, "From", from);
/* Build Remote Party-ID and From */
if (ast_test_flag(p, SIP_SENDRPID)) {
build_rpid(p);
add_header(req, "From", p->rpid_from);
} else
add_header(req, "From", from);
ast_copy_string(p->exten, l, sizeof(p->exten));
build_contact(p);
add_header(req, "To", to);
@ -4609,6 +4709,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
add_header(req, "Call-ID", p->callid);
add_header(req, "CSeq", tmp);
add_header(req, "User-Agent", default_useragent);
if (ast_test_flag(p, SIP_SENDRPID))
add_header(req, "Remote-Party-ID", p->rpid);
}
/*--- transmit_invite: Build REFER/INVITE/OPTIONS message and transmit it ---*/
@ -6827,7 +6929,7 @@ static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipme
if (debug)
ast_verbose("Found peer '%s'\n", peer->name);
/* Take the peer */
ast_copy_flags(p, peer, SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH);
ast_copy_flags(p, peer, SIP_TRUSTRPID | SIP_SENDRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH);
/* Copy SIP extensions profile to peer */
if (p->sipoptions)
@ -11197,6 +11299,10 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
ast_set_flag(mask, SIP_TRUSTRPID);
ast_set2_flag(flags, ast_true(v->value), SIP_TRUSTRPID);
res = 1;
} else if (!strcasecmp(v->name, "sendrpid")) {
ast_set_flag(mask, SIP_SENDRPID);
ast_set2_flag(flags, ast_true(v->value), SIP_SENDRPID);
res = 1;
} else if (!strcasecmp(v->name, "useclientcode")) {
ast_set_flag(mask, SIP_USECLIENTCODE);
ast_set2_flag(flags, ast_true(v->value), SIP_USECLIENTCODE);
@ -11576,7 +11682,7 @@ static struct sip_peer *temp_peer(const char *name)
ast_copy_flags(peer, &global_flags,
SIP_PROMISCREDIR | SIP_USEREQPHONE | SIP_TRUSTRPID | SIP_USECLIENTCODE |
SIP_DTMF | SIP_NAT | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE |
SIP_PROG_INBAND | SIP_OSPAUTH);
SIP_PROG_INBAND | SIP_OSPAUTH | SIP_SENDRPID);
strcpy(peer->context, default_context);
strcpy(peer->subscribecontext, default_subscribecontext);
strcpy(peer->language, default_language);
@ -11675,7 +11781,7 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int
ast_copy_flags(peer, &global_flags,
SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_USECLIENTCODE |
SIP_DTMF | SIP_REINVITE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE |
SIP_PROG_INBAND | SIP_OSPAUTH);
SIP_PROG_INBAND | SIP_OSPAUTH | SIP_SENDRPID);
peer->capability = global_capability;
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;

2
configs/sip.conf.sample
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@ -69,6 +69,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
@ -267,6 +268,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)

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