Re-fix how local tag is generated when sending a 481 to an INVITE.

Match our local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the sip_pvt,
it has been changed to a string field.

(closes issue ASTERISK-19892)
reported by Walter Doekes

Review: https://reviewboard.asterisk.org/r/1977
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Merged revisions 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369353 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2012-06-25 19:26:31 +00:00
parent 82a7409c15
commit e0883154cf
2 changed files with 18 additions and 17 deletions

View File

@@ -1557,7 +1557,7 @@ static char *generate_random_string(char *buf, size_t size);
static void build_callid_pvt(struct sip_pvt *pvt);
static void change_callid_pvt(struct sip_pvt *pvt, const char *callid);
static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
static void make_our_tag(char *tagbuf, size_t len);
static void make_our_tag(struct sip_pvt *pvt);
static int add_header(struct sip_request *req, const char *var, const char *value);
static int add_header_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
static int add_content(struct sip_request *req, const char *line);
@@ -7855,9 +7855,9 @@ static void build_callid_registry(struct sip_registry *reg, const struct ast_soc
}
/*! \brief Make our SIP dialog tag */
static void make_our_tag(char *tagbuf, size_t len)
static void make_our_tag(struct sip_pvt *pvt)
{
snprintf(tagbuf, len, "as%08lx", ast_random());
ast_string_field_build(pvt, tag, "as%08lx", ast_random());
}
/*! \brief Allocate Session-Timers struct w/in dialog */
@@ -7999,7 +7999,7 @@ struct sip_pvt *sip_alloc(ast_string_field callid, struct ast_sockaddr *addr,
p->do_history = recordhistory;
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
make_our_tag(p);
p->ocseq = INITIAL_CSEQ;
p->allowed_methods = UINT_MAX;
@@ -8294,7 +8294,7 @@ static void forked_invite_init(struct sip_request *req, const char *new_theirtag
memcpy(&p->flags, &original->flags, sizeof(p->flags));
copy_request(&p->initreq, &original->initreq);
ast_string_field_set(p, theirtag, new_theirtag);
ast_copy_string(p->tag, original->tag, sizeof(p->tag));
ast_string_field_set(p, tag, original->tag);
ast_string_field_set(p, uri, original->uri);
ast_string_field_set(p, our_contact, original->our_contact);
ast_string_field_set(p, fullcontact, original->fullcontact);
@@ -11103,7 +11103,7 @@ static int transmit_response_using_temp(ast_string_field callid, struct ast_sock
}
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
make_our_tag(p);
p->ocseq = INITIAL_CSEQ;
if (useglobal_nat && addr) {
@@ -13970,7 +13970,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char *
return 0;
} else {
p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
make_our_tag(p->tag, sizeof(p->tag)); /* create a new local tag for every register attempt */
make_our_tag(p); /* create a new local tag for every register attempt */
ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
}
} else {
@@ -23929,7 +23929,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_string_field_set(p, exten, "s");
/* Initialize our tag */
make_our_tag(p->tag, sizeof(p->tag));
make_our_tag(p);
/* First invitation - create the channel. Allocation
* failures are handled below. */
@@ -26119,7 +26119,7 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
/* Initialize tag for new subscriptions */
if (ast_strlen_zero(p->tag))
make_our_tag(p->tag, sizeof(p->tag));
make_our_tag(p);
if (!strncmp(eventheader, "presence", MAX(event_len, 8)) || !strncmp(eventheader, "dialog", MAX(event_len, 6))) { /* Presence, RFC 3842 */
unsigned int pidf_xml;
@@ -26627,14 +26627,15 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
if (!p->initreq.headers && req->has_to_tag) {
/* If this is a first request and it got a to-tag, it is not for us */
if (!req->ignore && req->method == SIP_INVITE) {
/* We will be subversive here. By blanking out the to-tag of the request,
* it will cause us to attach our own generated to-tag instead. This way,
* when we receive an ACK, the ACK will contain the to-tag we generated,
* resulting in a proper to-tag match.
/* Just because we think this is a dialog-starting INVITE with a to-tag
* doesn't mean it actually is. It could be a reinvite for an established, but
* unknown dialog. In such a case, we need to change our tag to the
* incoming INVITE's to-tag so that they will recognize the 481 we send and
* so that we will properly match their incoming ACK.
*/
char *to_header = (char *) sip_get_header(req, "To");
char *tag = strstr(to_header, ";tag=");
*tag = '\0';
char totag[128];
gettag(req, "To", totag, sizeof(totag));
ast_string_field_set(p, tag, totag);
p->pendinginvite = p->icseq;
transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
/* Will cease to exist after ACK */

View File

@@ -1025,6 +1025,7 @@ struct sip_pvt {
AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
AST_STRING_FIELD(redircause); /*!< Referring cause */
AST_STRING_FIELD(theirtag); /*!< Their tag */
AST_STRING_FIELD(tag); /*!< Our tag for this session */
AST_STRING_FIELD(username); /*!< [user] name */
AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
AST_STRING_FIELD(authname); /*!< Who we use for authentication */
@@ -1076,7 +1077,6 @@ struct sip_pvt {
*/
unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
unsigned short natdetected:1; /*!< Whether we detected a NAT when processing the Via */
char tag[11]; /*!< Our tag for this session */
int timer_t1; /*!< SIP timer T1, ms rtt */
int timer_b; /*!< SIP timer B, ms */
unsigned int sipoptions; /*!< Supported SIP options on the other end */