3855 Commits (2473b791b9002f1041e4c1f95d7e9f58733ff89b)

Author	SHA1	Message	Date
Sean Bright	2473b791b9	Replace calls to strtok() with strtok_r() ... strtok() uses a static buffer, making it not thread safe. Also add a #define to cause a compile failure if strtok is used. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	6 years ago
Giuseppe Sucameli	0bde3751a0	chan_sip: Fix leak using contact ACL ... Free old peer's contactacl before overwrite it within build_peer. ASTERISK-28194 Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c	6 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	3077ad0c24	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
Jasper Hafkenscheid	2cf5079205	chan_sip: Attempt ast_do_pickup in handle_invite_replaces ... When a call pickup is performed using and invite with replaces header the ast_do_pickup method is attempted and a PICKUP stasis message is sent. ASTERISK-28081 #close Reported-by: Luit van Drongelen Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a	7 years ago
Corey Farrell	90a11c4ae7	chan_sip deprecation. ... This officially deprecates chan_sip in Asterisk 17+. A warning is printed upon startup or module load to tell users that they should consider migrating. chan_sip is still built by default but the default modules.conf skips loading it at startup. Very important to note we are not scheduling a time where chan_sip will be removed. The goal of this change is to accurately inform end users of the current state of chan_sip and encourage movement to the fully supported chan_pjsip. Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93	7 years ago
Corey Farrell	687ab7aeee	astobj2: Eliminate legacy container allocation macros. ... These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are also removed. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
pk16208	6627c56b3d	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI ... With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago
hajekd	406be41f21	chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads ... Fixes random asterisk crash on start or reload with TLS phones. ASTERISK-28034 #close Reported-by: David Hajek Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac	7 years ago
Jaco Kroon	9680790531	chan_sip: improved ip:port finding of peers for non-UDP transports. ... Also remove function peer_ipcmp_cb since it's not used (according to rmudgett). Prior to b2c4e8660a (ASTERISK_27457) insecure=port was the defacto standard. That commit also prevented insecure=port from being applied for sip/tcp or sip/tls. Into consideration there are three sets of behaviour: 1. "previous" - before the above commit. 2. "current" - post above commit, pre this one. 3. "new" - post this commit. The problem that the above commit tried to address was guests over TCP. It succeeded in doing that but broke transport!=udp with host!=dynamic. This commit attempts to restore sane behaviour with respect to transport!=udp for host!=dynamic whilst still retaining the guest users over tcp. It should be noted that when looking for a peer, two passes are made, the first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer, thus looking for full matches (IP + Port), the second pass sets SIP_INSECURE_PORT, thus expecting matches on IP only where the matched peer allows for that (in the author's opinion: UDP with insecure=port, or any TCP based, non-dynamic host). In previous behaviour there was special handling for transport=tcp|tls whereby a peer would match during the first pass if the utilized transport was TCP|TLS (and the peer allowed that specific transport). This behaviour was wrong, or dubious at best. Consider two dynamic tcp peers, both registering from the same IP (NAT), in this case either peer could match for connections from an IP. It's also this behaviour that prevented SIP guests over tcp. The above referenced commit removed this behaviour, but kept applying the SIP_INSECURE_PORT only to WS|WSS|UDP. Since WS and WSS is also TCP based, the logic here should fall into the TCP category. This patch updates things such that the previously non-explicit (TCP behaviour) transport test gets performed explicitly (ie, matched peer must allow for the used transport), as well as the indeterministic source-port nature of the TCP protocol is taken into account. The new match algorithm now looks like: 1. As per previous behaviour, IP address is matched first. 2. Explicit filter with respect to transport protocol, previous behaviour was semi-implied in the test for TCP pure IP match - this now made explicit. 3. During first pass (without SIP_INSECURE_PORT), always match on port. 4. If doing UDP, match if matched against peer also has SIP_INSECURE_PORT, else don't match. 5. Match if not a dynamic host (for non-UDP protocols) 6. Don't match if this is WS|WSS, or we can't trust the Contact address (presumably due to NAT) 7. Match (we have a valid Contact thus if the IP matches we have no choice, this will likely only apply to non-NAT). To logic-test this we need a few different scenarios. Towards this end, I work with a set number of peers defined in sip.conf: [peer1] host=1.1.1.1 transport=tcp [peer2] host=1.1.1.1 transport=udp [peer3] host=1.1.1.1 port=5061 insecure=port transport=udp [peer4] host=1.1.1.2 transport=udp,tcp [peer5] host=dynamic transport=udp,tcp Test cases for UDP: 1 - incoming UDP request from 1.1.1.1: - previous: - pass 1: * peer1 or peer2 if from port 5060 (indeterminate, depends on peer ordering) * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 - current: as per previous. - new: - pass 1: * peer2 if from port 5060 * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 2 - incoming UDP request from 1.1.1.2: - previous: - pass 1: * peer5 if registered from 1.1.1.2 and port matches * peer4 if source port is 5060 - pass 2: * no match (guest) - current: as previous. - new as previous (with the variation that if peer5 didn't have udp as allowed transport it would not match peer5 whereas previous and current code could). 3 - incoming UDP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address and source port matches. - pass 2: * peer5 if insecure=port is additionally set. * no match (guest) - current - as per previous - new - as per previous Test cases for TCP based transports: 4 - incoming TCP request from 1.1.1.1 - previous: - pass 1 (indeterministic, depends on ordering of peers in memory): * peer1; or * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or * peer2 if the source port happens to be 5060; or * peer3 if the source port happens to be 5061. - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer1 or peer2 if from source port 5060 * peer3 if from source port 5060 * peer5 if registered as 1.1.1.1 and source port matches - pass 2: * no match (guest) - new: - pass 1: * peer 1 if from port 5060 * peer 5 if registered and source port matches - pass 2: * peer 1 5 - incoming TCP request from 1.1.1.2 - previous (indeterminate, depends on ordering): - pass 1: * peer4; or * peer5 if peer5 registered from 1.1.1.2 - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * no match (guest). - new: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * peer4 6 - incoming TCP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer5 if registered from that address and port matches. - pass 2: * no match (guest) - new: as per current. It should be noted the test cases don't make explicit mention of TLS, WS or WSS. WS and WSS previously followed UDP semantics, they will now enforce source port matching. TLS follow TCP semantics. The previous commit specifically tried to address test-case 6, but broke test-cases 4 and 5 in the process. ASTERISK-27881 #close Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2	7 years ago
Joshua Colp	25764691b0	Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"	7 years ago
Matthew Fredrickson	9f9dce05b2	netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API ... This function originally was used in chan_sip to enable some simplifying assumptions and eventually was copy and pasted into res_pjsip_logger and res_hep. Since it's replicated in three places, it's probably best to move it into the public netsock2 API for these modules to use. Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04	7 years ago
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	7 years ago
Jenkins2	fabfe701bb	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge"	7 years ago
Richard Mudgett	0c03eab962	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge ... There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	7 years ago
Richard Mudgett	d72a2966da	chan_sip.c: Fix INVITE with replaces channel ref leak. ... Given the below call scenario: A -> Ast1 -> B C <- Ast2 <- B 1) A calls B through Ast1 2) B calls C through Ast2 3) B transfers A to C When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of the channel between Ast1 and Ast2. Channel ref leaks are easily seen in the CLI "core show channels" output. The leaked channels appear in the output but you can do nothing with them and they never go away unless you restart Asterisk. * Properly account for the channel refs when imparting a channel into a bridge when handling an INVITE with replaces in handle_invite_replaces(). The ast_bridge_impart() function steals a channel ref but the code didn't account for how many refs were held by the code at the time and which ref was stolen. * Eliminated RAII_VAR in handle_invite_replaces(). ASTERISK-27740 Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4	7 years ago
Jenkins2	e45e8466f1	Merge "chan_sip: Peers with distinct source ports don't match, regardless of transport."	7 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	7 years ago
Jenkins2	1485719531	Merge "Replace direct checks of option_debug with DEBUG_ATLEAST macro."	7 years ago
Corey Farrell	c8a521b6c8	Replace direct checks of option_debug with DEBUG_ATLEAST macro. ... Checking option_debug directly is incorrect as it ignores file/module specific debug settings. This system-wide change replaces nearly all direct checks for option_debug with the DEBUG_ATLEAST macro. Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0	7 years ago
Jean Aunis	75a35ee5e8	chan_sip: Fix improper RTP framing on outgoing calls ... The "ptime" SDP parameter received in a SIP response was not honoured. Moreover, in the abscence of this "ptime" parameter, locally configured framing was lost during response processing. This patch systematically stores the framing information in the ast_rtp_codecs structure, taking it from the response or from the configuration as appropriate. ASTERISK-27674 Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c	7 years ago
Joshua Colp	259c80675e	chan_sip: Emit a second ringing event to ensure channel is found. ... When constructing a dialog-info+xml NOTIFY message a ringing channel is found if the state is ringing and further information is placed into the message. Due to the migration to the Stasis message bus this did not always work as expected. This change raises a second ringing event in such a way to guarantee that the event is received by chan_sip and another lookup is done to find the ringing channel. ASTERISK-24488 Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c	7 years ago
Richard Mudgett	8372138cce	chan_sip.c: Fix crash processing CANCEL. ... Check if initreq data string exists before using it when processing a CANCEL request. ASTERISK-27666 Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97	7 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. ... * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	7 years ago
Joshua Colp	6d5f4768a4	chan_sip: Check that an iostream exists before accessing. ... Before getting the file descriptor for an iostream check that it is present. ASTERISK-27534 Change-Id: Ie0aa1394007a37c30e337ea1176a6fb3a63bc99c	7 years ago
Sean Bright	15f8b9b8bf	ice: Increase foundation buffer size ... Per RFC 5245, the foundation specified with an ICE candidate can be up to 32 characters but we are only allowing for 31. ASTERISK-27498 #close Reported by: Michele Prà Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf	7 years ago
Corey Farrell	1b80ffa495	Fix Common Typo's. ... Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31	7 years ago
Corey Farrell	3c037ef972	chan_sip: Fix memory leaks. ... In change_redirecting_information variables we use ast_strlen_zero to see if a value should be saved. In the case where the value is not NULL but is a zero length string we leaked. handle_response_subscribe leaked a reference to the ccss monitor instance. Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f	7 years ago
Corey Farrell	6f8b34f9c1	chan_sip: Add security event for calls to invalid extension. ... Log a message to security events when an INVITE is received to an invalid extension. ASTERISK-25869 #close Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f	7 years ago
Joshua Colp	e411b7d112	Merge "chan_sip: 3PCC patch for AMI "SIPnotify""	7 years ago
Jenkins2	58514c7442	Merge "chan_sip: Don't crash in Dial on invalid destination"	7 years ago
Yasuhiko Kamata	cb249b2419	chan_sip: 3PCC patch for AMI "SIPnotify" ... A patch for sending in-dialog SIP NOTIFY message with "SIPnotify" AMI action. ASTERISK-27461 Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4	7 years ago
Sean Bright	0c9cc7e975	chan_sip: Don't send trailing \0 on keep alive packets ... This is a partial fix for ASTERISK~25817 but does not address the comments regarding RFC 5626. Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420	7 years ago
Sean Bright	5039b5741c	chan_sip: Don't crash in Dial on invalid destination ... Stripping the DNID in a SIP dial string can result in attempting to call the argument parsing macros on an empty string, causing a crash. ASTERISK-26131 #close Reported by: Dwayne Hubbard Patches: dw-asterisk-master-dnid-crash.patch (license #6257) patch uploaded by Dwayne Hubbard Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e	7 years ago
Alexander Traud	b2c4e8660a	chan_sip: Peers with distinct source ports don't match, regardless of transport. ... Previously, peers connected via TCP (or TLS) were matched by ignoring their source port. One cannot say anything when protocol:IP:port match, yes (see <http://stackoverflow.com/q/3329641>). However, when the ports do not match, the peers do not match as well. This change allows two peers connected to an Asterisk server via TCP (or TLS) behind a NAT (= same source IP address) to be differentiated via their port as well. ASTERISK-27457 Reported by: Stephane Chazelas Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa	8 years ago
Alexander Traud	491e2eba0d	chan_sip: ICE contained square brackets around IPv6 addresses. ... ASTERISK-27434 Change-Id: Iaeed89b4fa05d94c5f0ec2d3b7cd6e93d2d5a8f7	8 years ago
Sean Bright	04d3785a79	dtls: Add support for ephemeral DTLS certificates. ... This mimics the behavior of Chrome and Firefox and creates an ephemeral X.509 certificate for each DTLS session. Currently, the only supported key type is ECDSA because of its faster generation time, but other key types can be added in the future as necessary. ASTERISK-27395 Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4	8 years ago
Corey Farrell	6474de5f72	chan_sip: Fix SUBSCRIBE with missing "Expires" header. ... When chan_sip receives a SUBSCRIBE request with no "Expires" header it processes the request as an unsubscribe. This is incorrect, per RFC3264 when the "Expires" header is missing a default expiry should be used. ASTERISK-18140 Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5	8 years ago
Alexander Traud	840e08716b	chan_sip: Crypto attribute not last but first on SDP media level. ... This matches the behavior of the other SIP channel driver, chan_pjsip. ASTERISK-27365 Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6	8 years ago
Corey Farrell	c9e19b31f5	chan_sip: Fix output of 'sip set debug off'. ... When sip.conf contains 'sipdebug=yes' it is impossible to disable it using CLI 'sip set debug off'. This corrects the output of that CLI command to instruct the user to turn sipdebug off in the configuration file. ASTERISK-23462 #close Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318	8 years ago
Daniel Tryba	59b6e8467a	res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacy ... Currently privacy requests are only granted if the Privacy header value is exactly "id" (defined in RFC 3325). It ignores any other possible value (or a combination there of). This patch reverses the logic from testing for "id" to grant privacy, to testing for "none" and granting privacy for any other value. "none" must not be used in combination with any other value (RFC 3323 section 4.2). ASTERISK-27284 #close Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56	8 years ago
Jenkins2	68b506caaa	Merge "chan_sip: when getting sip pvt return failure if not found"	8 years ago
Scott Griepentrog	5553644284	chan_sip: when getting sip pvt return failure if not found ... In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612	8 years ago
Vitezslav Novy	67a2ca31f5	chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE ... If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE to both parties to set up media path directly between the endpoints. In this reINVITE msg SDP origin line (o=) contains IP address of endpoint instead of IP of asterisk. This behavior violates RFC3264, sec 8: "When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." This patch assures IP address of Asterisk is always sent in SDP origin line. ASTERISK-17540 Reported by: saghul Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e	8 years ago
kkm	4b58609c33	chan_sip: Access incoming REFER headers in dialplan ... This adds a way to access information passed along with SIP headers in a REFER message that initiates a transfer. Headers matching a dialplan variable GET_TRANSFERRER_DATA in the transferrer channel are added to a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH. The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for headers that should be put into the hash. If not set, no headers are included. If set to a string (perhaps 'X-' in a typical case), all headers starting this string are added. Empty string matches all headers. If there are multiple of the same header, only the latest occurrence in the REFER message is available in the hash. Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the referrer channel, and should be set with the '_' or '__' prefix. I avoided a specific reference to SIP or REFER, as in my mind the mechanism can be generalized to other channel techs. ASTERISK-27162 Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e	8 years ago
kkm	4c0798e91d	chan_sip: Add dialplan function SIP_HEADERS ... Syntax: SIP_HEADERS([prefix]) If the argument is specified, only the headers matching the given prefix are returned. The function returns a comma-separated list of SIP header names from an incoming INVITE message. Multiple headers with the same name are included in the list only once. The returned list can be iterated over using the functions POP() and SIP_HEADER(). For example, '${SIP_HEADERS(Co)}' might return the string 'Contact,Content-Length,Content-Type'. Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional extended headers sent by a peer. ASTERISK-27163 Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267	8 years ago
Corey Farrell	58d032112b	Fix compiler warnings on Fedora 26 / GCC 7. ... GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	8 years ago
Jenkins2	d6c08cc559	Merge "core: Remove 'Data Retrieval API'"	8 years ago
Jenkins2	75022f6b11	Merge "chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support)."	8 years ago
Sean Bright	325eeced6a	core: Remove 'Data Retrieval API' ... This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83	8 years ago