3396 Commits (2600a42e800a5c6c4eb18fb41cd172ab29bdd537)

Author	SHA1	Message	Date
Joshua Colp	9462a08aee	Merge "musiconhold: Only warn if music class is not found in memory and database."	10 years ago
Gianluca Merlo	894071ea2c	config: fix flags in uint option handler ... The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e	10 years ago
Walter Doekes	13cdf3e8a1	musiconhold: Only warn if music class is not found in memory and database. ... The log message when a MusicOnHold music class was not found was changed from debug level to WARNING level in Asterisk 11.19 and 13.5. For those using realtime musiconhold, this message is wrong because it warns before checking the database. This changeset delays the warning until after the database has been checked. Reported-by: Conrad de Wet ASTERISK-25444 #close Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf	10 years ago
Matthew Jordan	22e2340813	res/res_http_media_cache: Add an HTTP(S) backend for the core media cache ... This patch adds a bucket backend for the core media cache that interfaces to a remote HTTP server. When a media item is requested in the cache, the cache will query its bucket backends to see if they can provide the media item. If that media item has a scheme of HTTP or HTTPS, this backend will be invoked. The backend provides callbacks for the following: * create - this will always retrieve the URI specified by the provided bucket_file, and store it in the file specified by the object. * retrieve - this will pull the URI specified and store it in a temporary file. It is then up to the media cache to move/rename this file if desired. * delete - destroys the file associated with the bucket_file. * stale - if the bucket_file has expired, based on received HTTP headers from the remote server, or if the ETag on the server no longer matches the ETag stored on the bucket_file, the resource is determined to be stale. Note that the backend respects the ETag, Expires, and Cache-Control headers provided by the HTTP server it is querying. ASTERISK-25654 Change-Id: Ie201c2b34cafc0c90a7ee18d7c8359afaccc5250	10 years ago
Sergio Medina Toledo	bdccb81157	res_pjsip_refer.c: Fix seg fault in process of Refer-to header. ... The "Refer-to" header of an incoming REFER request is parsed by pjsip_parse_uri(). That function requires the URI parameter to be NULL terminated. Unfortunately, the previous code added the NULL terminator by overwriting memory that may not be safe. The overwritten memory results could be benign, memory corruption, or a segmentation fault. Now the URI is NULL terminated safely by copying the URI to a new chunk of memory with the correct size to be NULL terminated. ASTERISK-25814 #close Change-Id: I32565496684a5a49c3278fce06474b8c94b37342	10 years ago
zuul	d804b904c5	Merge "build: Add configure check for proto field of PJSIP TLS transport setting."	10 years ago
Joshua Colp	abe893725b	Merge "res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100"	10 years ago
Joshua Colp	4df7b3ae80	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	f0799da3ac	Merge "res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited"	10 years ago
zuul	7329f2ab97	Merge "res_pjsip: Strip spaces from items parsed from comma-separated lists"	10 years ago
Joshua Colp	5cf2226e01	Merge "main/cli.c: Refactor function to print seconds formatted"	10 years ago
zuul	760444a1f5	Merge "res_odbc_transaction: fix some format tab"	10 years ago
George Joseph	d2eb65f71e	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
Rodrigo Ramírez Norambuena	f690c105f3	res_odbc_transaction: fix some format tab ... Change-Id: I265e4ac47c629c9a63dd86b59df82a7ab3c64384	10 years ago
Rodrigo Ramírez Norambuena	0ec9fe5421	main/cli.c: Refactor function to print seconds formatted ... Refactor and created function ast_cli_print_timestr_fromseconds to print seconds formatted: year(s) week(s) day(s) hour(s) second(s) This function now is used in addons/cdr_mysql.c,cdr_pgsql.c, main/cli.c, res_config_ldap.c, res_config_pgsql.c. Change-Id: Ibeb8634102cd11d3f8623398b279cb731bcde36c	10 years ago
zuul	64c03d4d19	Merge "config_transport: Fix objects returned by ast_sip_get_transport_states"	10 years ago
George Joseph	2b9849625c	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
Kevin Harwell	0d2ccbca62	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	6af7fc4c37	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	d7fe2becdd	Merge "SIP diversion: Fix REDIRECTING(reason) value inconsistencies."	10 years ago
Joshua Colp	8140d7a8ef	Merge "res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason."	10 years ago
zuul	71427f1454	Merge "res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref."	10 years ago
zuul	53f84a4670	Merge "CHAOS: cleanup possible null vars on msg alloc failure"	10 years ago
George Joseph	7b71bca8a4	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	0a3f0e85ac	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	60aa871be3	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	25de01f301	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	8c8ef4efb0	res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. ... Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd	10 years ago
Richard Mudgett	75ec137e91	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
Joshua Colp	91f8763452	Merge "res_pjsip_t38.c: Back out part of an earlier fix attempt."	10 years ago
zuul	c47d15fda7	Merge "res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s."	10 years ago
Richard Mudgett	bf29a4e2e6	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	acf329a3c7	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
Joshua Colp	62d98b5a7f	Merge "res_pjsip/config_transport: Allow reloading transports."	10 years ago
George Joseph	7e3e1ddf7e	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
zuul	d6b2c1d98e	Merge "res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables"	10 years ago
zuul	062857ece3	Merge "res_pjsip_config_wizard: Add command to export primitive objects"	10 years ago
Joshua Colp	def3fb4634	Merge "res_pjproject: Add ability to map pjproject log levels to Asterisk log levels"	10 years ago
Christof Lauber	b7970cabfa	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
George Joseph	ba8adb4ce3	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	f8767a8804	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	14886643c6	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	62282bb8ce	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	49203628f9	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	4f08e9fb64	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	be811c4be1	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
zuul	23b2b7747d	Merge "Fix creation race of contact_status structures."	10 years ago
Mark Michelson	13b6c02945	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	5c400a0fed	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
zuul	1783edd181	Merge "res_pjsip: Refactor load_module/unload_module"	10 years ago