3334 Commits (7a74971771d9bf302cb26238996eeb3cb1df7af8)

Author	SHA1	Message	Date
zuul	dcebcaa3da	Merge "build: Add configure check for proto field of PJSIP TLS transport setting." into 13	10 years ago
Joshua Colp	6d309cd2cd	Merge "res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100" into 13	10 years ago
Joshua Colp	677a65fcbb	build: Add configure check for proto field of PJSIP TLS transport setting. ... Older versions of PJSIP do not have the proto field on the TLS transport setting structure. This change adds a configure check so even if it is not present we will still be able to build. Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9	10 years ago
zuul	9263ea0b13	Merge "res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited" into 13	10 years ago
George Joseph	530cff5f5f	res_pjsip: Strip spaces from items parsed from comma-separated lists ... Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2	10 years ago
Joshua Colp	cb70ed7214	Merge "config_transport: Fix objects returned by ast_sip_get_transport_states" into 13	10 years ago
George Joseph	27f32cd0a6	res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited ... Per RFC3325, the 'From' header is now anonymized on outgoing calls when caller id presentation is prohibited. TID = trust_id_outbound PRO = Set(CALLERID(pres)=prohib) USR = endpoint/from_user DOM = endpoint/from_domain PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) Conditions |Result --------------------|---------------------------------------------------- TID PRO USR DOM |PAI FROM --------------------|---------------------------------------------------- Y Y abc def.ghi |PRI "Anonymous" <sip:abc@def.ghi> Y Y abc |PRI "Anonymous" <sip:abc@anonymous.invalid> Y Y def.ghi |PRI "Anonymous" <sip:anonymous@def.ghi> Y Y |PRI "Anonymous" <sip:anonymous@anonymous.invalid> Y N abc def.ghi |YES <sip:abc@def.ghi> Y N abc |YES <sip:abc@<ip_address>> Y N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi> Y N |YES "Caller Name" <sip:<caller_exten>@<ip_address>> N Y abc def.ghi |NO "Anonymous" <sip:abc@def.ghi> N Y abc |NO "Anonymous" <sip:abc@anonymous.invalid> N Y def.ghi |NO "Anonymous" <sip:anonymous@def.ghi> N Y |NO "Anonymous" <sip:anonymous@anonymous.invalid> N N abc def.ghi |YES <sip:abc@def.ghi> N N abc |YES <sip:abc@<ip_address>> N N def.ghi |YES "Caller Name" <sip:<caller_exten>@def.ghi> N N |YES "Caller Name" <sip:<caller_exten>@<ip_address>> ASTERISK-25791 #close Reported-by: Anthony Messina Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9	10 years ago
zuul	039ea76a8a	Merge "res_pjsip_dtmf_info: NULL terminate the message body." into 13	10 years ago
Kevin Harwell	ff3da61c35	res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 ... During the transfer process, some phones (okay it was the Jitsi softphone, but maybe others are out there) send a "bye" immediately after receiving a SIP Notify. When a "bye" is received early for some types of transfers the transferer channel may no longer be available during late stage transfer processing. For instance, during an attended transfer involving stasis bridging at one point the created local channel looks for an associated swap channel in order to retrieve the stasis application name. If the transferer has hung up then the local channel will fail to find it. The local channel then has no way to know which stasis app to enter, so it fails and hangs up as well. Thus the transfer does not complete as expected. This patch delays the sending of the initial notify in order to give the transfer process enough time to gather the necessary data for a successful transfer. ASTERISK-25771 Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16	10 years ago
Joshua Colp	26b8f2692e	res_pjsip_dtmf_info: NULL terminate the message body. ... PJSIP does not ensure that when printing the message body the buffer will be NULL terminated. This is problematic when searching for the signal and duration values of the DTMF. This change ensures the buffer is always NULL terminated. Change-Id: I52653a1a60c93092d06af31a27408d569cc98968	10 years ago
Joshua Colp	86124f63c8	Merge "SIP diversion: Fix REDIRECTING(reason) value inconsistencies." into 13	10 years ago
Joshua Colp	3b6b164f2e	Merge "res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref." into 13	10 years ago
zuul	d6e274b97d	Merge "res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason." into 13	10 years ago
Joshua Colp	f4670c6a76	Merge "CHAOS: cleanup possible null vars on msg alloc failure" into 13	10 years ago
George Joseph	6f0d7ce9db	config_transport: Fix objects returned by ast_sip_get_transport_states ... ast_sip_get_transport_states was returning a container of internal_state objects instead of ast_sip_transport_state objects. This was causing transport lookups to fail, most noticably in res_pjsip_nat, which couldn't find the correct external addresses. This was causing contacts to go out with internal ip addresses. ASTERISK-25830 #close Reported-by: Sean Bright Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e	10 years ago
Scott Griepentrog	1ea7a5a774	CHAOS: cleanup possible null vars on msg alloc failure ... In message.c, if msg_alloc fails to init the string field, vars may be null, so use a null tolerant cleanup. In res_pjsip_messaging.c, if msg_data_create fails, mdata will be null, so use a null tolerant cleanup. ASTERISK-25323 Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56	10 years ago
Scott Griepentrog	3c37c7071f	CHAOS: prevent crash on failed strdup ... This patch avoids crashing on a null pointer if the strdup() allocation fails. ASTERISK-25323 Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5	10 years ago
Richard Mudgett	4165ea7778	SIP diversion: Fix REDIRECTING(reason) value inconsistencies. ... Previous chan_sip behavior: Before this patch chan_sip would always strip any quotes from an incoming reason and pass that value up as the REDIRECTING(reason). For an outgoing reason value, chan_sip would check the value against known values and quote any it didn't recognize. Incoming 480 response message reason text was just assigned to the REDIRECTING(reason). Previous chan_pjsip behavior: Before this patch chan_pjsip would always pass the incoming reason value up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip would send the reason value as passed down. With this patch: Both channel drivers match incoming reason values with values documented by REDIRECTING(reason) and values documented by RFC5806 regardless of whether they are quoted or not. RFC5806 values are mapped to the equivalent REDIRECTING(reason) documented value and is set in REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a quoted string version ('"unconditional"') is converted to REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal with 'cfu' instead of any of the aliases. The incoming 480 response reason text supported by chan_sip checks for known reason values and if not matched then puts quotes around the reason string and assigns that to REDIRECTING(reason). Both channel drivers send outgoing known REDIRECTING(reason) values as the unquoted RFC5806 equivalent. User custom values are either sent as is or with added quotes if SIP doesn't allow a character within the value as part of a RFC3261 Section 25.1 token. Note that there are still limitations on what characters can be put in a custom user value. e.g., embedding quotes in the middle of the reason string is silly and just going to cause you grief. * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the 'cfu' value. * Added missing malloc() NULL return check in res_pjsip_diversion.c set_redirecting_reason(). * Fixed potential read from a stale pointer in res_pjsip_diversion.c add_diversion_header(). The reason string needed to be copied into the tdata memory pool to ensure that the string would always be available. Otherwise, if the reason string returned by reason_code_to_str() was a user's reason string then the string could be freed later by another thread. Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87	10 years ago
Richard Mudgett	41f4af4ce5	res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. ... Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd	10 years ago
Richard Mudgett	4c5998ff55	res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. ... * Fix double unref of other_party channel in off nominal path. * This is unlikely to be a real problem. However, for safety, in handle_incoming_request() keep the datastore ref with the other_party channel ref until we are finished with the other_party channel. Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821	10 years ago
zuul	84d1036205	Merge "res_pjsip_t38.c: Back out part of an earlier fix attempt." into 13	10 years ago
zuul	414e297648	Merge "res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s." into 13	10 years ago
Richard Mudgett	263a39f2cc	res_pjsip_t38.c: Back out part of an earlier fix attempt. ... This backs out item 4 of the 4875e5ac32 commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame that it puts into the bridge may or may not be processed by the time the bridged peer is kicked out of the bridge. If it is processed then all is well. However, if it is not processed then that channel is stuck in fax mode until it hangs up or maybe if it joins another bridge for T.38 faxing. ASTERISK-25582 Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7	10 years ago
George Joseph	ee947d4a7a	res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s. ... There are a few cases where we're emitting notices or warnings for things that really need neither, like a client retrying to subscribe to mwi when they're not conifgured for it. They get a 404 so there's no need for non-debug messages. Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f	10 years ago
zuul	d35c494df1	Merge "res_pjsip/config_transport: Allow reloading transports." into 13	10 years ago
George Joseph	6e70e8ccdb	res_sorcery_memory_cache: Fix SEGV in some CLI commands ... A few of the CLI commands weren't checking for enough arguments and were SEGVing. Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413	10 years ago
Christof Lauber	ea9deff996	res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables ... Introduced realloaction of ast_str buf in sqlite3_escape functions in case the returned buffer from threadstorage was actually too small. Change-Id: I3c5eb43aaade93ee457943daddc651781954c445	10 years ago
zuul	aa637f0a91	Merge "res_pjsip_config_wizard: Add command to export primitive objects" into 13	10 years ago
Joshua Colp	56561c386a	Merge "res_pjproject: Add ability to map pjproject log levels to Asterisk log levels" into 13	10 years ago
George Joseph	d2a1457e0b	res_pjsip/config_transport: Allow reloading transports. ... The 'reload' mechanism actually involves closing the underlying socket and calling the appropriate udp, tcp or tls start functions again. Only outbound_registration, pubsub and session needed work to reset the transport before sending requests to insure that the pjsip transport didn't get pulled out from under them. In my testing, no calls were dropped when a transport was changed for any of the 3 transport types even if ip addresses or ports were changed. To be on the safe side however, a new transport option was added (allow_reload) which defaults to 'no'. Unless it's explicitly set to 'yes' for a transport, changes to that transport will be ignored on a reload of res_pjsip. This should preserve the current behavior. Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf	10 years ago
George Joseph	6b921f706d	res_pjproject: Add ability to map pjproject log levels to Asterisk log levels ... Warnings and errors in the pjproject libraries are generally handled by Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings or errors so the messages emitted by pjproject directly are either superfluous or misleading. A good exampe of this are the level-0 errors pjproject emits when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS client be treated any differently? A config file for res_pjproject has bene added (pjproject.conf) and a new log_mappings object allows mapping pjproject levels to Asterisk levels (or nothing). The defaults if no pjproject.conf file is found are the same as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR, 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG<level> Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898	10 years ago
Alexei Gradinari	f295088764	res_pjsip_outbound_publish: Fix processing 412 response ... When Asterisk receives a 412 (Conditional Request Failed) response it has to recreate publish session. There is bug in res_pjsip_outbound_publish.c The function sip_outbound_publish_client_alloc is called with wrong object while processing 412 (Conditional Request Failed) response. This patch fixes it. ASTERISK-25229 #close Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359	10 years ago
George Joseph	87ab65c557	res_odbc: Fix exports.in for missing symbols ... res_odbc.exports.in was missing a few symbols. Changed to wildcards. Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c	10 years ago
George Joseph	c0f3062031	res_statsd: Fix exports.in for missing symbols ... res_statsd.export.in was missing the _va variations of the log functions causing Asterisk to crash in res_pjsip if OPTIONAL_API wasn't enabled. ASTERISK-25727 #close Reported-by: Gergely Dömsödi Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b	10 years ago
George Joseph	5e848dae7b	res_pjsip_config_wizard: Add command to export primitive objects ... A new command (pjsip export config_wizard primitives) has been added that will export all the pjsip objects it created to the console or a file suitable for reuse in a pjsip.conf file. ASTERISK-24919 #close Reported-by: Ray Crumrine Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b	10 years ago
George Joseph	34c64707d1	res_pjsip_caller_id: Fix segfault when replacing rpid or pai header ... If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify the header added by the dialplan function. Since the header added by the dialplan function is generic string, there are no virtual functions to parse the uri and we get a segfault when we try. Since the modify, was really only an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER and recreate it. This raises a question for another time though: What should happen with duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups so if it's session supplement is loaded after res_pjsip_caller_id's (or any other module that adds headers), there'll be dups in the message. ASTERISK-25337 #close Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa	10 years ago
Joshua Colp	9c28acd9db	Merge "Fix creation race of contact_status structures." into 13	10 years ago
Mark Michelson	ebe167f792	Fix creation race of contact_status structures. ... It is possible when processing a SIP REGISTER request to have two threads end up creating contact_status structures in sorcery. contact_status is created using a "find or create" function. If two threads call into this at the same time, each thread will fail to find an existing contact_status, and so both will end up creating a new contact status. During testing, we would see sporadic failures because the PJSIP_CONTACT() dialplan function would operate on a different contact_status than what had been updated by res_pjsip/pjsip_options. The fix here is two-fold: 1) The "find or create" function for contact_status now has a lock around the entire operation. This way, if two threads attempt the operation simultaneously, the first to get there will create the object, and the second will find the object created by the first thread. 2) res_sorcery_memory has had its create callback updated so that it will not allow for objects with duplicate IDs to be created. Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97	10 years ago
Joshua Colp	1c4f2a920d	res_pjsip_pubsub: Move where the subscription is stored to after initialized. ... A problem arose when testing the AMI subscription listing actions where it was possible for a subscription that had not been fully initialized to be listed. This was problematic as the underlying listing code would crash. This change makes it so the subscription tree is fully set up before it is added to the list of subscriptions. This ensures that when the listing actions get the subscription it is valid. ASTERISK-25738 #close Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48	10 years ago
Joshua Colp	1cddd954f9	Merge "res_pjsip: Refactor load_module/unload_module" into 13	10 years ago
zuul	953ba9da88	Merge "res_pjsip: Handle pjsip_dlg_create_uas deprecation" into 13	10 years ago
George Joseph	b1b797e0e7	res_pjsip: Refactor load_module/unload_module ... load_module was just too hairy with every step having to clean up all previous steps on failure. Some of the pjproject init calls have now been moved to a separate load_pjsip function and the unload_pjsip function was enhanced to clean up everything if an error happened at any stage of the load process. In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns and ast_threadpool_shutdowns were also corrected. Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302	10 years ago
zuul	8c15f309c2	Merge "Resources/res_phoneprov: fix memory leak and heap-use-after-free" into 13	10 years ago
Badalyan Vyacheslav	20e9792fbc	Resources/res_phoneprov: fix memory leak and heap-use-after-free ... * heap-use-after-free happens when we free "cfg" but then use "value" which refers to it * A memory leak occurs because in some cases it is not released "defaults" ASTERISK-25721 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469	10 years ago
George Joseph	c1bf014ea0	res_pjsip: Handle pjsip_dlg_create_uas deprecation ... Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically increments the lock on the returned dialog. To account for this, configure.ac now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use the original call or the new one. If the new one was used, the ref count is decremented before returning. ASTERISK-25751 #close Reported-by Josh Colp Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8	10 years ago
George Joseph	2451d4e455	res_pjsip: Fix infinite recursion when loading transports from realtime ... Attempting to load a transport from realtime was forcing asterisk into an infinite recursion loop. The first thing transport_apply did was to do a sorcery retrieve by id for an existing transport of the same name. For files, this just returns the previous object from res_sorcery_config's internal container, if any. For realtime, the res_sourcery_realtime driver looks in the database and finds the existing row but now it has to rehydrate it into a sorcery object which means calling... transport_apply. And so it goes. The main issue with loading from realtime (apart from the loop) was that transport stores structures and pointers directly in the ast_sip_transport structure instead of the separate ast_transport_state structure. This patch separates those items into the ast_sip_transport_state structure. The pattern is roughly the same as res_pjsip_outbound_registration. Although all current usages of ast_sip_transport and ast_sip_transport_state were modified to use the new ast_sip_get_transport_state API, the original items are left in ast_sip_transport and kept updated to maintain ABI compatability for third-party modules. They are marked as deprecated and noted that they're now in ast_sip_transport_state. ASTERISK-25606 #close Reported-by: Martin Moučka Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19	10 years ago
Mark Michelson	c5e7d5c105	Merge "logging: Remove/fix some message annoyances" into 13	10 years ago
Joshua Colp	6cac364284	Merge "res_stasis_device_state: Fix refcounting error." into 13	10 years ago
Joshua Colp	014fc9ef65	Merge "res_xmpp: Does not connect in component mode" into 13	10 years ago
Mark Michelson	23829b3253	res_stasis_device_state: Fix refcounting error. ... Device state subscription lifetimes were governed by when the subscription was established and unsubscribed from. However, it is possible that at the time of unsubscription, there could be device state events still in flight. When those device state events occur, the device state callback could attempt to dereference a freed pointer. Crash. This change ensures that the lifetime of the device state subscription does not end until the underlying stasis subscription has confirmed that its final message has been sent. Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2	10 years ago