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Issue #6674: Set the URI correctly on BYEs when we have an incoming call

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Reported by aubergine, fix by oej


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2006-04-07 08:19:52 +00:00
parent 255e77ae6e
commit b7d32b9e0b
1 changed files with 65 additions and 38 deletions
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103
channels/chan_sip.c
View File

@@ -5918,15 +5918,11 @@ static void reg_source_db(struct sip_peer *peer)
register_peer_exten(peer, TRUE);
}
/*! \brief Parse contact header for 200 OK on INVITE */
/*! \brief Save contact header for 200 OK on INVITE */
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
{
char contact[250];
char *c, *n, *pt;
int port;
struct hostent *hp;
struct ast_hostent ahp;
struct sockaddr_in oldsin;
char *c;
/* Look for brackets */
ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
@@ -5937,28 +5933,54 @@ static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
/* Save URI for later ACKs, BYE or RE-invites */
ast_string_field_set(pvt, okcontacturi, c);
/* We should return false for URI:s we can't handle,
like sips:, tel:, mailto:,ldap: etc */
return TRUE;
}
/*! \brief Change the other partys IP address based on given contact */
static int set_address_from_contact(struct sip_pvt *pvt)
{
struct hostent *hp;
struct ast_hostent ahp;
int port;
char *c, *host, *pt;
char *contact;
if (ast_test_flag(&pvt->flags[0], SIP_NAT_ROUTE)) {
/* NAT: Don't trust the contact field. Just use what they came to us
with. */
pvt->sa = pvt->recv;
return 0;
}
/* Work on a copy */
contact = ast_strdupa(pvt->fullcontact);
/* Make sure it's a SIP URL */
if (strncasecmp(c, "sip:", 4)) {
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", c);
if (strncasecmp(contact, "sip:", 4)) {
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
} else
c += 4;
contact += 4;
/* Ditch arguments */
n = strchr(c, ';');
if (n)
*n = '\0';
host = strchr(contact, ';');
if (host)
*host = '\0';
/* Grab host */
n = strchr(c, '@');
if (!n) {
n = c;
host = strchr(contact, '@');
if (!host) { /* No username part */
host = contact;
c = NULL;
} else {
*n = '\0';
n++;
*host = '\0';
host++;
}
pt = strchr(n, ':');
pt = strchr(host, ':');
if (pt) {
*pt = '\0';
pt++;
@@ -5966,24 +5988,17 @@ static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
} else
port = DEFAULT_SIP_PORT;
oldsin = pvt->sa;
if (!ast_test_flag(&pvt->flags[0], SIP_NAT_ROUTE)) {
/* XXX This could block for a long time XXX */
/* We should only do this if it's a name, not an IP */
hp = ast_gethostbyname(n, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Invalid host '%s'\n", n);
return -1;
}
pvt->sa.sin_family = AF_INET;
memcpy(&pvt->sa.sin_addr, hp->h_addr, sizeof(pvt->sa.sin_addr));
pvt->sa.sin_port = htons(port);
} else {
/* Don't trust the contact field. Just use what they came to us
with. */
pvt->sa = pvt->recv;
/* XXX This could block for a long time XXX */
/* We should only do this if it's a name, not an IP */
hp = ast_gethostbyname(host, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Invalid host name in Contact: (can't resolve in DNS) : '%s'\n", host);
return -1;
}
pvt->sa.sin_family = AF_INET;
memcpy(&pvt->sa.sin_addr, hp->h_addr, sizeof(pvt->sa.sin_addr));
pvt->sa.sin_port = htons(port);
return 0;
}
@@ -9746,6 +9761,17 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
/* This is important when we have a SIP proxy between us and the phone */
if (outgoing) {
parse_ok_contact(p, req);
if(set_address_from_contact(p)) {
/* Bad contact - we don't know how to reach this device */
/* We need to ACK, but then send a bye */
/* OEJ: Possible issue that may need a check:
If we have a proxy route between us and the device,
should we care about resolving the contact
or should we just send it?
*/
if (!ignore)
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* Save Record-Route for any later requests we make on this dialogue */
build_route(p, req, 1);
@@ -9761,9 +9787,9 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
ast_queue_frame(p->owner, &ast_null_frame);
}
} else {
/* It's possible we're getting an ACK after we've tried to disconnect
/* It's possible we're getting an 200 OK after we've tried to disconnect
by sending CANCEL */
/* THIS NEEDS TO BE CHECKED: OEJ */
/* First send ACK, then send bye */
if (!ignore)
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
@@ -10793,6 +10819,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
copy_request(&p->initreq, req);
if (debug)
ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
parse_ok_contact(p, req);
} else { /* Re-invite on existing call */
/* Handle SDP here if we already have an owner */
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {