mirror of
https://github.com/asterisk/asterisk.git
synced 2026-07-14 23:54:29 -07:00
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@151019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
+182
-65
@@ -31,41 +31,8 @@
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* ********** IMPORTANT *
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* \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
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* settings, dialplan commands and dialplans apps/functions
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* See \ref sip_tcp_tls
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*
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* ******** TCP implementation changes needed
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* \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
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* \todo Save TCP/TLS sessions in registry
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* If someone registers a SIPS uri, this forces us to set up a TLS connection back.
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* \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
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* \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
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* The tcpbindaddr config option should only be used to open ADDITIONAL ports
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* So we should propably go back to
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* bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
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* if tlsenable=yes, open TLS port (provided we also have cert)
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* tcpbindaddr = extra address for additional TCP connections
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* tlsbindaddr = extra address for additional TCP/TLS connections
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* udpbindaddr = extra address for additional UDP connections
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* These three options should take multiple IP/port pairs
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* Note: Since opening additional listen sockets is a *new* feature we do not have today
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* the XXXbindaddr options needs to be disabled until we have support for it
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*
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* \todo Be prepared for one outbound and another incoming socket per pvt. This applies
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* specially to communication with other peers (proxies).
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* \todo We need to test TCP sessions with SIP proxies and in regards
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* to the SIP outbound specs.
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* \todo transport=tls was deprecated in RFC3261 and should not be used at all. See section 22.2.2.
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*
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* \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
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* message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
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* \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
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* multiple domains in our TLS implementation, meaning one socket and one cert per domain
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* \todo Selection of transport for a request needs to be done after we've parsed all route headers,
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* also considering outbound proxy options.
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* First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
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* Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
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* DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
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* Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
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*
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*
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* ******** General TODO:s
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* \todo Better support of forking
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@@ -120,6 +87,84 @@
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* the sip_hangup() function
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*/
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/*! \page sip_tcp_tls SIP TCP and TLS support
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* The TCP and TLS support is unfortunately implemented in a way that is not
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* SIP compliant and tested in a SIP infrastructure. We hope to fix this for
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* at least release 1.6.2. This code was new in 1.6.0 and won't be fixed for
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* that release, due to the current release policy. Only bugs compared with
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* the working functionality in 1.4 will be fixed. Bugs in new features will
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* be fixed in the next release. As 1.6.1 is already in release
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* candidate mode, there will be a buggy SIP channel in that release too.
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*
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* If you have opinions about this release policy, send mail to the asterisk-dev
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* mailing list.
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*
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* \par tcpfixes TCP implementation changes needed
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* \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
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* \todo Save TCP/TLS sessions in registry
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* If someone registers a SIPS uri, this forces us to set up a TLS connection back.
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* \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
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* \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
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* The tcpbindaddr config option should only be used to open ADDITIONAL ports
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* So we should propably go back to
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* bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
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* if tlsenable=yes, open TLS port (provided we also have cert)
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* tcpbindaddr = extra address for additional TCP connections
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* tlsbindaddr = extra address for additional TCP/TLS connections
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* udpbindaddr = extra address for additional UDP connections
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* These three options should take multiple IP/port pairs
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* Note: Since opening additional listen sockets is a *new* feature we do not have today
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* the XXXbindaddr options needs to be disabled until we have support for it
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*
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* \todo re-evaluate the transport= setting in sip.conf. This is right now not well
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* thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
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* even if udp is the configured first transport.
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*
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* \todo Be prepared for one outbound and another incoming socket per pvt. This applies
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* specially to communication with other peers (proxies).
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* \todo We need to test TCP sessions with SIP proxies and in regards
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* to the SIP outbound specs.
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* \todo transport=tls was deprecated in RFC3261 and should not be used at all. See section 22.2.2.
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*
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* \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
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* message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
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* \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
|
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* multiple domains in our TLS implementation, meaning one socket and one cert per domain
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* \todo Selection of transport for a request needs to be done after we've parsed all route headers,
|
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* also considering outbound proxy options.
|
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* First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
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* Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
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* DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
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* Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
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* \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
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* devices directly from the dialplan. UDP is only a fallback if no other method works,
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* in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
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* MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
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*
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* When dialling unconfigured peers (with no port number) or devices in external domains
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* NAPTR records MUST be consulted to find configured transport. If they are not found,
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* SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
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* If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
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* \note this only applies if there's no outbound proxy configured for the session. If an outbound
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* proxy is configured, these procedures might apply for locating the proxy and determining
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* the transport to use for communication with the proxy.
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* \par Other bugs to fix ----
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* __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
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* - sets TLS port as default for all TCP connections, unless other port is given in contact.
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* parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
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* - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
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* a bad guess.
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* - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
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* get_destination(struct sip_pvt *p, struct sip_request *oreq)
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* - Doesn't store the information that we got an incoming SIPS request in the channel, so that
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* we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
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* fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
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* channel variable in the dialplan.
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* get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
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* - As above, if we have a SIPS: uri in the refer-to header
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* - Does not check transport in refer_to uri.
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*/
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/*** MODULEINFO
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<depend>chan_local</depend>
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***/
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@@ -507,7 +552,7 @@ struct sip_proxy {
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char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
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struct sockaddr_in ip; /*!< Currently used IP address and port */
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time_t last_dnsupdate; /*!< When this was resolved */
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enum sip_transport transport;
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enum sip_transport transport;
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int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
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/* Room for a SRV record chain based on the name */
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};
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@@ -620,7 +665,8 @@ static const struct cfsip_methods {
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#define SIP_OPT_FROMCHANGE (1 << 17)
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#define SIP_OPT_RECLISTINV (1 << 18)
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#define SIP_OPT_RECLISTSUB (1 << 19)
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#define SIP_OPT_UNKNOWN (1 << 20)
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#define SIP_OPT_OUTBOUND (1 << 20)
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#define SIP_OPT_UNKNOWN (1 << 21)
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/*! \brief List of well-known SIP options. If we get this in a require,
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@@ -646,6 +692,8 @@ static const struct cfsip_options {
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{ SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
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/* Disable the REFER subscription, RFC 4488 */
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{ SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
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/* SIP outbound - the final NAT battle - draft-sip-outbound */
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{ SIP_OPT_OUTBOUND, NOT_SUPPORTED, "outbound" },
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/* RFC3327: Path support */
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{ SIP_OPT_PATH, NOT_SUPPORTED, "path" },
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/* RFC3840: Callee preferences */
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@@ -690,7 +738,7 @@ static const struct cfsip_options {
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/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
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#define STANDARD_SIP_PORT 5060
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/*! \brief Standard SIP TLS port for sips: from RFC 3261. DO NOT CHANGE THIS */
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/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
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#define STANDARD_TLS_PORT 5061
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/*! \note in many SIP headers, absence of a port number implies port 5060,
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@@ -1801,11 +1849,11 @@ static struct sip_auth *authl = NULL;
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/* --- Sockets and networking --------------*/
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/*! \brief Main socket for SIP communication.
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/*! \brief Main socket for UDP SIP communication.
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*
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* sipsock is shared between the SIP manager thread (which handles reload
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* requests), the io handler (sipsock_read()) and the user routines that
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* issue writes (using __sip_xmit()).
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* requests), the udp io handler (sipsock_read()) and the user routines that
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* issue udp writes (using __sip_xmit()).
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* The socket is -1 only when opening fails (this is a permanent condition),
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* or when we are handling a reload() that changes its address (this is
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* a transient situation during which we might have a harmless race, see
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@@ -2021,6 +2069,7 @@ static int sip_refer_allocate(struct sip_pvt *p);
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static void ast_quiet_chan(struct ast_channel *chan);
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static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
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static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
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/*!
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* \brief generic function for determining if a correct transport is being
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* used to contact a peer
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@@ -3666,6 +3715,8 @@ static char *get_in_brackets(char *tmp)
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* \verbatim
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* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
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* \endverbatim
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*
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* \todo This function needs to look for ;transport= too
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*/
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static int parse_uri(char *uri, char *scheme,
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char **ret_name, char **pass, char **domain, char **port, char **options)
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@@ -7845,7 +7896,15 @@ static void add_route(struct sip_request *req, struct sip_route *route)
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add_header(req, "Route", r);
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}
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/*! \brief Set destination from SIP URI */
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/*! \brief Set destination from SIP URI
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*
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* Parse uri to h (host) and port - uri is already just the part inside the <>
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* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
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* If there's a port given, turn NAPTR/SRV off. NAPTR might indicate SIPS preference even
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* for SIP: uri's
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*
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* If there's a sips: uri scheme, TLS will be required.
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*/
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static void set_destination(struct sip_pvt *p, char *uri)
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{
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char *h, *maddr, hostname[256];
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@@ -7853,9 +7912,8 @@ static void set_destination(struct sip_pvt *p, char *uri)
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struct hostent *hp;
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struct ast_hostent ahp;
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int debug=sip_debug_test_pvt(p);
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/* Parse uri to h (host) and port - uri is already just the part inside the <> */
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/* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...] */
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int tls_on = FALSE;
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int use_dns = global_srvlookup;
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if (debug)
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ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);
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@@ -7866,10 +7924,12 @@ static void set_destination(struct sip_pvt *p, char *uri)
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++h;
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else {
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h = uri;
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if (!strncasecmp(h, "sip:", 4))
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if (!strncasecmp(h, "sip:", 4)) {
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h += 4;
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else if (!strncasecmp(h, "sips:", 5))
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} else if (!strncasecmp(h, "sips:", 5)) {
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h += 5;
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||||
tls_on = TRUE;
|
||||
}
|
||||
}
|
||||
hn = strcspn(h, ":;>") + 1;
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if (hn > sizeof(hostname))
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@@ -7883,9 +7943,9 @@ static void set_destination(struct sip_pvt *p, char *uri)
|
||||
/* Parse port */
|
||||
++h;
|
||||
port = strtol(h, &h, 10);
|
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}
|
||||
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? */
|
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maddr = strstr(h, "maddr=");
|
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@@ -7896,6 +7956,8 @@ static void set_destination(struct sip_pvt *p, char *uri)
|
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hn = sizeof(hostname);
|
||||
ast_copy_string(hostname, maddr, hn);
|
||||
}
|
||||
|
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/*! \todo XXX If we have use_dns on, then look for NAPTR/SRV, otherwise, just look for A records */
|
||||
|
||||
hp = ast_gethostbyname(hostname, &ahp);
|
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if (hp == NULL) {
|
||||
@@ -9707,8 +9769,8 @@ static int transmit_state_notify(struct sip_pvt *p, int state, int full, int tim
|
||||
|
||||
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 @@ static int transmit_refer(struct sip_pvt *p, const char *dest)
|
||||
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 @@ static int transmit_refer(struct sip_pvt *p, const char *dest)
|
||||
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 @@ static int __set_address_from_contact(const char *fullcontact, struct sockaddr_i
|
||||
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 @@ static int __set_address_from_contact(const char *fullcontact, struct sockaddr_i
|
||||
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 @@ static int __set_address_from_contact(const char *fullcontact, struct sockaddr_i
|
||||
|
||||
/* 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 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
|
||||
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 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
|
||||
|
||||
/* 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 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
|
||||
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 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
|
||||
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 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
|
||||
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 @@ static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *r
|
||||
* 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@host:port;uri-parameters?headers
|
||||
* sips:user:password@host:port;uri-parameters?headers
|
||||
* sip:user:password;user-parameters@host:port;uri-parameters?headers
|
||||
* sips:user:password;user-parameters@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 @@ static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq)
|
||||
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 @@ static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq)
|
||||
\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 @@ static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *t
|
||||
}
|
||||
|
||||
/*! \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 @@ static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoi
|
||||
}
|
||||
|
||||
|
||||
/*! \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 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
|
||||
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 @@ static struct ast_custom_function sipchaninfo_function = {
|
||||
"- 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 @@ static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
|
||||
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);
|
||||
@@ -15951,6 +16058,16 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
|
||||
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
|
||||
This transaction is already scheduled to be killed by sip_hangup().
|
||||
|
||||
Reference in New Issue
Block a user