[svn-commits] oej: trunk r151019 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Sun Oct 19 02:20:41 CDT 2008
Author: oej
Date: Sun Oct 19 02:20:40 2008
New Revision: 151019
URL: http://svn.digium.com/view/asterisk?view=rev&rev=151019
Log:
Adding changes from train and flight back home from SIPit23 in Lannion, France.
- Additional comments on TCP/TLS implementation
- Some additions for new drafts/rfcs (no new functionality really, mostly documentation)
- Other random small fixes
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=151019&r1=151018&r2=151019
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Oct 19 02:20:40 2008
@@ -31,41 +31,8 @@
* ********** IMPORTANT *
* \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
* settings, dialplan commands and dialplans apps/functions
+ * See \ref sip_tcp_tls
*
- * ******** TCP implementation changes needed
- * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
- * \todo Save TCP/TLS sessions in registry
- * If someone registers a SIPS uri, this forces us to set up a TLS connection back.
- * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
- * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
- * The tcpbindaddr config option should only be used to open ADDITIONAL ports
- * So we should propably go back to
- * bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
- * if tlsenable=yes, open TLS port (provided we also have cert)
- * tcpbindaddr = extra address for additional TCP connections
- * tlsbindaddr = extra address for additional TCP/TLS connections
- * udpbindaddr = extra address for additional UDP connections
- * These three options should take multiple IP/port pairs
- * Note: Since opening additional listen sockets is a *new* feature we do not have today
- * the XXXbindaddr options needs to be disabled until we have support for it
- *
- * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
- * specially to communication with other peers (proxies).
- * \todo We need to test TCP sessions with SIP proxies and in regards
- * to the SIP outbound specs.
- * \todo transport=tls was deprecated in RFC3261 and should not be used at all. See section 22.2.2.
- *
- * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
- * message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
- * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
- * multiple domains in our TLS implementation, meaning one socket and one cert per domain
- * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
- * also considering outbound proxy options.
- * First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
- * Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
- * DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
- * Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
- *
*
* ******** General TODO:s
* \todo Better support of forking
@@ -120,6 +87,84 @@
* the sip_hangup() function
*/
+/*! \page sip_tcp_tls SIP TCP and TLS support
+ * The TCP and TLS support is unfortunately implemented in a way that is not
+ * SIP compliant and tested in a SIP infrastructure. We hope to fix this for
+ * at least release 1.6.2. This code was new in 1.6.0 and won't be fixed for
+ * that release, due to the current release policy. Only bugs compared with
+ * the working functionality in 1.4 will be fixed. Bugs in new features will
+ * be fixed in the next release. As 1.6.1 is already in release
+ * candidate mode, there will be a buggy SIP channel in that release too.
+ *
+ * If you have opinions about this release policy, send mail to the asterisk-dev
+ * mailing list.
+ *
+ * \par tcpfixes TCP implementation changes needed
+ * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
+ * \todo Save TCP/TLS sessions in registry
+ * If someone registers a SIPS uri, this forces us to set up a TLS connection back.
+ * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
+ * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
+ * The tcpbindaddr config option should only be used to open ADDITIONAL ports
+ * So we should propably go back to
+ * bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
+ * if tlsenable=yes, open TLS port (provided we also have cert)
+ * tcpbindaddr = extra address for additional TCP connections
+ * tlsbindaddr = extra address for additional TCP/TLS connections
+ * udpbindaddr = extra address for additional UDP connections
+ * These three options should take multiple IP/port pairs
+ * Note: Since opening additional listen sockets is a *new* feature we do not have today
+ * the XXXbindaddr options needs to be disabled until we have support for it
+ *
+ * \todo re-evaluate the transport= setting in sip.conf. This is right now not well
+ * thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
+ * even if udp is the configured first transport.
+ *
+ * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
+ * specially to communication with other peers (proxies).
+ * \todo We need to test TCP sessions with SIP proxies and in regards
+ * to the SIP outbound specs.
+ * \todo transport=tls was deprecated in RFC3261 and should not be used at all. See section 22.2.2.
+ *
+ * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
+ * message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
+ * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
+ * multiple domains in our TLS implementation, meaning one socket and one cert per domain
+ * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
+ * also considering outbound proxy options.
+ * First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
+ * Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
+ * DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
+ * Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
+ * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
+ * devices directly from the dialplan. UDP is only a fallback if no other method works,
+ * in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
+ * MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
+ *
+ * When dialling unconfigured peers (with no port number) or devices in external domains
+ * NAPTR records MUST be consulted to find configured transport. If they are not found,
+ * SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
+ * If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
+ * \note this only applies if there's no outbound proxy configured for the session. If an outbound
+ * proxy is configured, these procedures might apply for locating the proxy and determining
+ * the transport to use for communication with the proxy.
+ * \par Other bugs to fix ----
+ * __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
+ * - sets TLS port as default for all TCP connections, unless other port is given in contact.
+ * parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
+ * - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
+ * a bad guess.
+ * - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
+ * get_destination(struct sip_pvt *p, struct sip_request *oreq)
+ * - Doesn't store the information that we got an incoming SIPS request in the channel, so that
+ * we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
+ * fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
+ * channel variable in the dialplan.
+ * get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
+ * - As above, if we have a SIPS: uri in the refer-to header
+ * - Does not check transport in refer_to uri.
+ */
+
/*** MODULEINFO
<depend>chan_local</depend>
***/
@@ -507,7 +552,7 @@
char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
struct sockaddr_in ip; /*!< Currently used IP address and port */
time_t last_dnsupdate; /*!< When this was resolved */
- enum sip_transport transport;
+ enum sip_transport transport;
int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
/* Room for a SRV record chain based on the name */
};
@@ -620,7 +665,8 @@
#define SIP_OPT_FROMCHANGE (1 << 17)
#define SIP_OPT_RECLISTINV (1 << 18)
#define SIP_OPT_RECLISTSUB (1 << 19)
-#define SIP_OPT_UNKNOWN (1 << 20)
+#define SIP_OPT_OUTBOUND (1 << 20)
+#define SIP_OPT_UNKNOWN (1 << 21)
/*! \brief List of well-known SIP options. If we get this in a require,
@@ -646,6 +692,8 @@
{ SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
/* Disable the REFER subscription, RFC 4488 */
{ SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
+ /* SIP outbound - the final NAT battle - draft-sip-outbound */
+ { SIP_OPT_OUTBOUND, NOT_SUPPORTED, "outbound" },
/* RFC3327: Path support */
{ SIP_OPT_PATH, NOT_SUPPORTED, "path" },
/* RFC3840: Callee preferences */
@@ -690,7 +738,7 @@
/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT 5060
-/*! \brief Standard SIP TLS port for sips: from RFC 3261. DO NOT CHANGE THIS */
+/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_TLS_PORT 5061
/*! \note in many SIP headers, absence of a port number implies port 5060,
@@ -1801,11 +1849,11 @@
/* --- Sockets and networking --------------*/
-/*! \brief Main socket for SIP communication.
+/*! \brief Main socket for UDP SIP communication.
*
* sipsock is shared between the SIP manager thread (which handles reload
- * requests), the io handler (sipsock_read()) and the user routines that
- * issue writes (using __sip_xmit()).
+ * requests), the udp io handler (sipsock_read()) and the user routines that
+ * issue udp writes (using __sip_xmit()).
* The socket is -1 only when opening fails (this is a permanent condition),
* or when we are handling a reload() that changes its address (this is
* a transient situation during which we might have a harmless race, see
@@ -2021,6 +2069,7 @@
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
+
/*!
* \brief generic function for determining if a correct transport is being
* used to contact a peer
@@ -3666,6 +3715,8 @@
* \verbatim
* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
* \endverbatim
+ *
+ * \todo This function needs to look for ;transport= too
*/
static int parse_uri(char *uri, char *scheme,
char **ret_name, char **pass, char **domain, char **port, char **options)
@@ -7845,7 +7896,15 @@
add_header(req, "Route", r);
}
-/*! \brief Set destination from SIP URI */
+/*! \brief Set destination from SIP URI
+ *
+ * Parse uri to h (host) and port - uri is already just the part inside the <>
+ * general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
+ * If there's a port given, turn NAPTR/SRV off. NAPTR might indicate SIPS preference even
+ * for SIP: uri's
+ *
+ * If there's a sips: uri scheme, TLS will be required.
+ */
static void set_destination(struct sip_pvt *p, char *uri)
{
char *h, *maddr, hostname[256];
@@ -7853,9 +7912,8 @@
struct hostent *hp;
struct ast_hostent ahp;
int debug=sip_debug_test_pvt(p);
-
- /* Parse uri to h (host) and port - uri is already just the part inside the <> */
- /* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...] */
+ int tls_on = FALSE;
+ int use_dns = global_srvlookup;
if (debug)
ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);
@@ -7866,10 +7924,12 @@
++h;
else {
h = uri;
- if (!strncasecmp(h, "sip:", 4))
+ if (!strncasecmp(h, "sip:", 4)) {
h += 4;
- else if (!strncasecmp(h, "sips:", 5))
+ } else if (!strncasecmp(h, "sips:", 5)) {
h += 5;
+ tls_on = TRUE;
+ }
}
hn = strcspn(h, ":;>") + 1;
if (hn > sizeof(hostname))
@@ -7883,9 +7943,9 @@
/* Parse port */
++h;
port = strtol(h, &h, 10);
- }
- else
- port = STANDARD_SIP_PORT;
+ use_dns = FALSE;
+ } else
+ port = tls_on ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
/* Got the hostname:port - but maybe there's a "maddr=" to override address? */
maddr = strstr(h, "maddr=");
@@ -7896,6 +7956,8 @@
hn = sizeof(hostname);
ast_copy_string(hostname, maddr, hn);
}
+
+ /*! \todo XXX If we have use_dns on, then look for NAPTR/SRV, otherwise, just look for A records */
hp = ast_gethostbyname(hostname, &ahp);
if (hp == NULL) {
@@ -9707,8 +9769,8 @@
ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to));
c = get_in_brackets(to);
- if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
- ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
+ if (strncasecmp(to, "sip:", 4) && strncasecmp(to, "sips:", 5)) {
+ ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", to);
return -1;
}
mto = remove_uri_parameters(c);
@@ -10341,6 +10403,7 @@
char referto[256];
char *ttag, *ftag;
char *theirtag = ast_strdupa(p->theirtag);
+ int use_tls=FALSE;
if (sipdebug)
ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);
@@ -10359,21 +10422,23 @@
ast_copy_string(from, of, sizeof(from));
of = get_in_brackets(from);
ast_string_field_set(p, from, of);
- if (!strncasecmp(of, "sip:", 4))
+ if (!strncasecmp(of, "sip:", 4)) {
of += 4;
- else if (!strncasecmp(of, "sips:", 5))
+ }else if (!strncasecmp(of, "sips:", 5)) {
of += 5;
- else
- ast_log(LOG_NOTICE, "From address missing 'sip(s):', using it anyway\n");
+ use_tls = TRUE;
+ } else {
+ ast_log(LOG_NOTICE, "From address missing 'sip(s):', assuming sip:\n");
+ }
/* Get just the username part */
if ((c = strchr(dest, '@')))
c = NULL;
else if ((c = strchr(of, '@')))
*c++ = '\0';
if (c)
- snprintf(referto, sizeof(referto), "<sip:%s@%s>", dest, c);
+ snprintf(referto, sizeof(referto), "<sip%s:%s@%s>", use_tls ? "s" : "", dest, c);
else
- snprintf(referto, sizeof(referto), "<sip:%s>", dest);
+ snprintf(referto, sizeof(referto), "<sip%s:%s>", use_tls ? "s" : "", dest);
/* save in case we get 407 challenge */
sip_refer_allocate(p);
@@ -10632,6 +10697,7 @@
char contact_buf[256];
char contact2_buf[256];
char *contact, *contact2;
+ int use_tls = FALSE;
/* Work on a copy */
ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
@@ -10650,11 +10716,14 @@
We still need to be able to send to the remote agent through the proxy.
*/
if (tcp) {
- if (parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
+ if (!parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
+ use_tls = TRUE;
+ } else {
if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
+ /*! \todo XXX why are we setting TLS port if there's no port given? parse_uri needs to return the transport. */
} else {
if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
@@ -10663,6 +10732,10 @@
/* XXX This could block for a long time XXX */
/* We should only do this if it's a name, not an IP */
+ /* \todo - if there's no PORT number in contact - we are required to check NAPTR/SRV records
+ to find transport, port address and hostname. If there's a port number, we have to
+ assume that the domain part is a host name and only look for an A/AAAA record in DNS.
+ */
hp = ast_gethostbyname(host, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Invalid host name in Contact: (can't resolve in DNS) : '%s'\n", host);
@@ -10764,6 +10837,12 @@
ast_string_field_build(pvt, our_contact, "<%s>", curi);
/* Make sure it's a SIP URL */
+ /*! \todo This code assumes that the Contact is using the same transport as the
+ REGISTER request. That might not be true at all. You can receive
+ sips: requests over any transport. Needs to be fixed.
+ Does not parse the ;transport uri parameter at this point, which might be handy
+ in some situations.
+ */
if (pvt->socket.type == SIP_TRANSPORT_TLS) {
if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL)) {
if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL))
@@ -10780,6 +10859,7 @@
/* Check that they're allowed to register at this IP */
/* XXX This could block for a long time XXX */
+ /*! \todo Check NAPTR/SRV if we have not got a port in the URI */
hp = ast_gethostbyname(host, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Invalid host '%s'\n", host);
@@ -10788,7 +10868,7 @@
return PARSE_REGISTER_FAILED;
}
memcpy(&testsin.sin_addr, hp->h_addr, sizeof(testsin.sin_addr));
- if ( ast_apply_ha(global_contact_ha, &testsin) != AST_SENSE_ALLOW ||
+ if (ast_apply_ha(global_contact_ha, &testsin) != AST_SENSE_ALLOW ||
ast_apply_ha(peer->contactha, &testsin) != AST_SENSE_ALLOW) {
ast_log(LOG_WARNING, "Host '%s' disallowed by rule\n", host);
*peer->fullcontact = '\0';
@@ -10796,6 +10876,8 @@
return PARSE_REGISTER_FAILED;
}
+ /*! \todo This could come before the checking of DNS earlier on, to avoid
+ DNS lookups where we don't need it... */
if (!ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE)) {
peer->addr.sin_family = AF_INET;
memcpy(&peer->addr.sin_addr, hp->h_addr, sizeof(peer->addr.sin_addr));
@@ -10823,7 +10905,7 @@
snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expire, peer->username, peer->fullcontact);
/* Saving TCP connections is useless, we won't be able to reconnect
XXX WHY???? XXX
- \todo check this
+ \todo Fix this immediately.
*/
if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
ast_db_put("SIP/Registry", peer->name, data);
@@ -11235,12 +11317,15 @@
* Terminate the uri at the first ';' or space.
* Technically we should ignore escaped space per RFC3261 (19.1.1 etc)
* but don't do it for the time being. Remember the uri format is:
+ * (User-parameters was added after RFC 3261)
*\verbatim
*
- * sip:user:password at host:port;uri-parameters?headers
- * sips:user:password at host:port;uri-parameters?headers
+ * sip:user:password;user-parameters at host:port;uri-parameters?headers
+ * sips:user:password;user-parameters at host:port;uri-parameters?headers
*
*\endverbatim
+ * \todo As this function does not support user-parameters, it's considered broken
+ * and needs fixing.
*/
static char *terminate_uri(char *uri)
{
@@ -11497,6 +11582,9 @@
if (ast_strlen_zero(tmp))
return 0;
+ /*! \todo This function does not take user-parameters into consideration.
+ First look for @, then start looking for ; to find uri-parameters.
+ */
params = strchr(tmp, ';');
exten = get_in_brackets(tmp);
@@ -11551,6 +11639,10 @@
\return 0 on success (found a matching extension),
1 for pickup extension or overlap dialling support (if we support it),
-1 on error.
+
+ \note If the incoming uri is a SIPS: uri, we are required to carry this across
+ the dialplan, so that the outbound call also is a sips: call or encrypted
+ IAX2 call. If that's not available, the call should FAIL.
*/
static int get_destination(struct sip_pvt *p, struct sip_request *oreq)
{
@@ -11764,7 +11856,12 @@
}
/*! \brief Call transfer support (the REFER method)
- * Extracts Refer headers into pvt dialog structure */
+ * Extracts Refer headers into pvt dialog structure
+ *
+ * \note If we get a SIPS uri in the refer-to header, we're required to set up a secure signalling path
+ * to that extension. As a minimum, this needs to be added to a channel variable, if not a channel
+ * flag.
+ */
static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
{
@@ -11943,7 +12040,9 @@
}
-/*! \brief Call transfer support (old way, deprecated by the IETF)--*/
+/*! \brief Call transfer support (old way, deprecated by the IETF)
+ * \note does not account for SIPS: uri requirements, nor check transport
+ */
static int get_also_info(struct sip_pvt *p, struct sip_request *oreq)
{
char tmp[256] = "", *c, *a;
@@ -12399,6 +12498,8 @@
of2 = ast_strdupa(of);
/* ignore all fields but name */
+ /*! \todo Samme logical error as in many places above. Need a generic function for this.
+ */
if (p->socket.type == SIP_TRANSPORT_TLS) {
if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy)) {
if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy))
@@ -15550,7 +15651,12 @@
"- t38passthrough 1 if T38 is offered or enabled in this channel, otherwise 0\n"
};
-/*! \brief Parse 302 Moved temporalily response */
+/*! \brief Parse 302 Moved temporalily response
+ \todo XXX Doesn't redirect over TLS on sips: uri's.
+ If we get a redirect to a SIPS: uri, this needs to be going back to the
+ dialplan (this is a request for a secure signalling path).
+ Note that transport=tls is deprecated, but we need to support it on incoming requests.
+*/
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
{
char tmp[SIPBUFSIZE];
@@ -15576,6 +15682,7 @@
else {
if (strncasecmp(trans, "udp", 3))
ast_debug(1, "received contact with an invalid transport, '%s'\n", s);
+ /* This will assume UDP for all unknown transports */
transport = SIP_TRANSPORT_UDP;
}
} while(0);
@@ -15950,6 +16057,16 @@
ast_string_field_set(p, theirtag, NULL);
proc_422_rsp(p, req);
break;
+
+ case 428: /* Use identity header - rfc 4474 - not supported by Asterisk yet */
+ xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
+ append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
+ ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
+ if (p->owner)
+ ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+ break;
+
+
case 487: /* Cancelled transaction */
/* We have sent CANCEL on an outbound INVITE
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