518 Commits (0d487b53b13e9debb51ad8fe84b3767de5d25d75)

Author	SHA1	Message	Date
Joshua Colp	c2a72e6aa6	Merge "install_prereq: Checkout of libSRTP 1.5.x."	9 years ago
Alexander Traud	8476a9332f	install_prereq: Checkout of libSRTP 1.5.x. ... Since 5th November 2014, the master branch of libSRTP changed the prefix of several member names and is not compatible with the source code in Asterisk anymore. Therefore instead, this change checks out the latest version of the libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as backend. This makes AES-GCM and AES-IN possible. ASTERISK-22131 #close Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6	9 years ago
Alexei Gradinari	1c949eea6c	res_pjsip: Added "subscribe_context" to endpoint ... If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514	9 years ago
Joshua Colp	040a11cecd	Merge "res_pjsip: improve realtime performance #2"	9 years ago
Alexei Gradinari	6fa3ed0679	res_pjsip: improve realtime performance #2 ... The patch removes updating all Endpoints' status on startup. Instead, only non-qualified aors with static contact and non-qualified non-expired contacts are retrieved from the realtime to update the endpoint status to ONLINE. The endpoint name was added to the contact object to simply find the endpoint that created this contact. The status of endpoints with qualified aors will be updated by 'qualify' functions. ASTERISK-26061 #close Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df	9 years ago
Mark Michelson	b6bd97eea2	Fix Alembic upgrades. ... A non-existent constraint was being referenced in the upgrade script. This patch corrects the problem by removing the reference. In addition, the head of the alembic branch referred to a non-existent revision. This has been fixed by referring to the proper revision. This patch fixes another realtime problem as well. Our Alembic scripts store booleans as yes or no values. However, Sorcery tries to insert "true" or "false" instead. This patch introduces a new boolean type that translates to "yes" or "no" instead. ASTERISK-26128 #close Change-Id: I51574736a881189de695a824883a18d66a52dcef	9 years ago
Joshua Colp	4505a59dc9	alembic: Fix migration. ... The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting to use UniqueConstraint and failing. It was not imported and after importing it also continued to fail. I've changed the script to use the explicit name of the constraint instead. Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9	9 years ago
Alexei Gradinari	31f17abe44	res_pjsip: add "via_addr", "via_port", "call_id" to contact ... As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576	9 years ago
Alexei Gradinari	69a85a519f	res_pjsip: Endpoint IP Access Controls ... With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02	9 years ago
Joshua Colp	87787bb889	Merge "res_pjsip: improve realtime performance"	9 years ago
Alexei Gradinari	cc4c5f5693	res_pjsip: improve realtime performance ... This patch modified pjsip_options to retrieve only permament contacts for aor if the qualify_frequency is > 0 and persisted contacts if the qualify_frequency is > 0. This patch also fixed a bug in res_sorcery_astdb. res_sorcery_astdb doesn't save object data retrived from astdb. ASTERISK-25826 Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05	9 years ago
Alexei Gradinari	4df48581f1	pjsip: Added "reg_server" to contacts (fixed alembic) ... ASTERISK-25931 Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549	9 years ago
zuul	d9ed4e3153	Merge "pjproject_bundled: Various fixes discovered during testing of OSes"	9 years ago
George Joseph	e61716b774	pjproject_bundled: Various fixes discovered during testing of OSes ... For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c	9 years ago
Alexei Gradinari	2b1edee772	pjsip: Added "reg_server" to contacts. ... If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8	9 years ago
George Joseph	4ebf9a938d	res_pjsip: Add ability to identify by Authorization username ... A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	9 years ago
Alexei Gradinari	860b135c88	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	9 years ago
George Joseph	751d7a5a49	alembic: Remove batch operations (and sqlite support) ... Because SQLite doesn't support full ALTER capabilities, alembic scripts require batch operations. However, that capability wasn't available until 0.7.0 which some distributions haven't reached yet. Therefore, the batch operations introduced in commit 86d6e44cc (review 2319) have been reverted and SQLite is unsupported again, for now anyway. Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql. ASTERISK-25890 #close Reported-by: Harley Peters Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80	9 years ago
Joshua Colp	86ccd1bd2b	Merge "install_prereq: Fix check_installed_debs remove subversion"	9 years ago
George Joseph	060b7b83bc	install_prereq: Fix check_installed_debs remove subversion ... check_installed_debs wasn't handling virtual packages like libsrtp-dev and libresample-dev and on multiarch systems it was accidentally filtering out all packages if any :i386 packages were found instead of just filtering out the :i386 packages themselves. Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda	9 years ago
George Joseph	e2524fcee3	res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited ... res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea	9 years ago
George Joseph	c948ce9651	sorcery/res_pjsip: Refactor for realtime performance ... There were a number of places in the res_pjsip stack that were getting all endpoints or all aors, and then filtering them locally. A good example is pjsip_options which, on startup, retrieves all endpoints, then the aors for those endpoints, then tests the aors to see if the qualify_frequency is > 0. One issue was that it never did anything with the endpoints other than retrieve the aors so we probably could have skipped a step and just retrieved all aors. But nevermind. This worked reasonably well with local config files but with a realtime backend and thousands of objects, this was a nightmare. The issue really boiled down to the fact that while realtime supports predicates that are passed to the database engine, the non-realtime sorcery backends didn't. They do now. The realtime engines have a scheme for doing simple comparisons. They take in an ast_variable (or list) for matching, and the name of each variable can contain an operator. For instance, a name of "qualify_frequency >" and a value of "0" would create a SQL predicate that looks like "where qualify_frequency > '0'". If there's no operator after the name, the engines add an '=' so a simple name of "qualify_frequency" and a value of "10" would return exact matches. The non-realtime backends decide whether to include an object in a result set by calling ast_sorcery_changeset_create on every object in the internal container. However, ast_sorcery_changeset_create only does exact string matches though so a name of "qualify_frequency >" and a value of "0" returns nothing because the literal "qualify_frequency >" doesn't match any name in the objset set. So, the real task was to create a generic string matcher that can take a left value, operator and a right value and perform the match. To that end, strings.c has a new ast_strings_match(left, operator, right) function. Left and right are the strings to operate on and the operator can be a string containing any of the following: = (or NULL or ""), !=, >, >=, <, <=, like or regex. If the operator is like or regex, the right string should be a %-pattern or a regex expression. If both left and right can be converted to float, then a numeric comparison is performed, otherwise a string comparison is performed. To use this new function on ast_variables, 2 new functions were added to config.c. One that compares 2 ast_variables, and one that compares 2 ast_variable lists. The former is useful when you want to compare 2 ast_variables that happen to be in a list but don't want to traverse the list. The latter will traverse the right list and return true if all the variables in it match the left list. Now, the backends' fields_cmp functions call ast_variable_lists_match instead of ast_sorcery_changeset_create and they can now process the same syntax as the realtime engines. The realtime backend just passes the variable list unaltered to the engine. The only gotcha is that there's no common realtime engine support for regex so that's been noted in the api docs for ast_sorcery_retrieve_by_fields. Only one more change to sorcery was done... A new config flag "allow_unqualified_fetch" was added to reg_sorcery_realtime. "no": ignore fetches if no predicate fields were supplied. "error": same as no but emit an error. (good for testing) "yes": allow (the default); "warn": allow but emit a warning. (good for testing) Now on to res_pjsip... pjsip_options was modified to retrieve aors with qualify_frequency > 0 rather than all endpoints then all aors. Not only was this a big improvement in realtime retrieval but even for config files there's an improvement because we're not going through endpoints anymore. res_pjsip_mwi was modified to retieve only endpoints with something in the mailboxes field instead of all endpoints then testing mailboxes. res_pjsip_registrar_expire was completely refactored. It was retrieving all contacts then setting up scheduler entries to check for expiration. Now, it's a single thread (like keepalive) that periodically retrieves only contacts whose expiration time is < now and deletes them. A new contact_expiration_check_interval was added to global with a default of 30 seconds. Ross Beer reports that with this patch, his Asterisk startup time dropped from around an hour to under 30 seconds. There are still objects that can't be filtered at the database like identifies, transports, and registrations. These are not going to be anywhere near as numerous as endpoints, aors, auths, contacts however. Back to allow_unqualified_fetch. If this is set to yes and you have a very large number of objects in the database, the pjsip CLI commands will attempt to retrive ALL of them if not qualified with a LIKE. Worse, if you type "pjsip show endpoint <tab>" guess what's going to happen? :) Having a cache helps but all the objects will have to be retrieved at least once to fill the cache. Setting allow_unqualified_fetch=no prevents the mass retrieve and should be used on endpoints, auths, aors, and contacts. It should NOT be used for identifies, registrations and transports since these MUST be retrieved in bulk. Example sorcery.conf: [res_pjsip] endpoint=config,pjsip.conf,criteria=type=endpoint endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error ASTERISK-25826 #close Reported-by: Ross Beer Tested-by: Ross Beer Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67	9 years ago
Leif Madsen	0da36fca6b	Add initial support to build Docker images ... This work-in-progress is the first step to being able to reliably build Asterisk containers from the Asterisk source. I'm submitting this based on feedback gained at AstriDevCon 2015. Information about how to use this is provided in contrib/docker/README.md and will result in a local Asterisk container being built right from your source. I believe this can eventually be automated via hub.docker.com. Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1	9 years ago
George Joseph	471ff375fd	install_prereq: Add packages for bundled pjproject ... RedHat/CentOS needs python-devel Debian/Ubuntu needs automake, libsrtp-dev and python-dev Ubuntu also needed libncurses5-dev for cmenuselect so while not needed for pjproject, I adedd it anyway. Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089	9 years ago
George Joseph	b8b7c2e428	alembic: Fix downgrade and tweak for sqlite ... Downgrade had a few issues. First there was an errant 'update' statement in add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we weren't cleaning up the ENUMs so subsequent upgrades on postgres failed because the types already existed. For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly. Fortunately alembic batch_operations takes care of this for us if we use it so the alter and drops were converted to use batch operations. Here's an example downgrade: with op.batch_alter_table('ps_endpoints') as batch_op: batch_op.drop_column('tos_audio') batch_op.drop_column('tos_video') batch_op.add_column(sa.Column('tos_audio', yesno_values)) batch_op.add_column(sa.Column('tos_video', yesno_values)) batch_op.drop_column('cos_audio') batch_op.drop_column('cos_video') batch_op.add_column(sa.Column('cos_audio', yesno_values)) batch_op.add_column(sa.Column('cos_video', yesno_values)) with op.batch_alter_table('ps_transports') as batch_op: batch_op.drop_column('tos') batch_op.add_column(sa.Column('tos', yesno_values)) # Can't cast integers to YESNO_VALUES, so dropping and adding is required batch_op.drop_column('cos') batch_op.add_column(sa.Column('cos', yesno_values)) Upgrades from base to head and downgrades from head to base were tested repeatedly for postgresql, mysql/mariadb, and sqlite3. Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8	9 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	9 years ago
George Joseph	9b13ab6a63	pjsip/alembic: Add missing columns to system and registration ... ps_systems needed disable_tcp_switch ps_registrations needed line and endpoint ASTERISK-25737 #close Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19	9 years ago
George Joseph	52b29f9b4c	pjsip/alembic: Fix definition of qualify_timeout ... A recent commit set qualify_timeout to Decimal which isn't supported. This path corrects it to Float. Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf	9 years ago
Richard Mudgett	a4dcbdf50f	Fix alembic branches on master. ... Change-Id: I64ed21fec50eb833641ca49d92184f6aaabd86e8	9 years ago
Joshua Colp	2cdbd4d711	Merge "pjsip/alembic: Fix qualify_timeout column definition"	9 years ago
Daniel Journo	c60d6c0162	pjsip/alembic: Fix qualify_timeout column definition ... Corrects the qualify_timeout column type from Integer to Decimal ASTERISK-25686 #close Reported-by: Marcelo Terres Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8	9 years ago
Daniel Journo	8182146e85	pjsip: Add option global/regcontext ... Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62	9 years ago
Richard Mudgett	1fffe71f77	res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts". ... PJPROJECT has a function available to dump the compile time options used when building the library. * Add CLI "pjsip show buildopts" command. * Update contrib/scripts/autosupport to get pjproject information. Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748	9 years ago
George Joseph	a41aab477a	pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address ... On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88	9 years ago
Joshua Colp	e106292a92	Merge "Alembic: Increase column size of PJSIP AOR "contact"."	9 years ago
Joshua Colp	8bd9e2dcc0	Merge "Alembic: Add PJSIP global keep_alive_interval."	9 years ago
Mark Michelson	c5e16fe33a	Alembic: Add PJSIP global keep_alive_interval. ... The keep_alive_interval option was added about a year ago, but no alembic revision was created to add the appropriate column to the database. This commit fixes the problem and adds the column. This was discovered by running the testsuite with automatic conversion to realtime enabled. Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a	9 years ago
Walter Doekes	64b2046f3d	Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts. ... The spandspflow2pcap.py creates pcap files from fax.log files, generated through 'fax set debug on' when receiving a fax. An example fax.log is included as spandspflow2pcap.log. The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a recent version of SIPp. ASTERISK-25660 #close Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526	9 years ago
Mark Michelson	cfb34adb83	Alembic: Increase column size of PJSIP AOR "contact". ... When running the PJSIP AMI "show_endpoint" test with automatic conversion to realtime, the test would fail. This was because the AOR "contact" column was sized at 40, and the configured contact was larger than that. This commit increases the size of the contact column to 255 characters. Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1	10 years ago
Corey Farrell	cd5ae02812	Increase account code maximum length to 80. ... This increases the maximum length of account code's to match extensions. This ensures it is always possible to set an accountcode to ${EXTEN} without truncation. ASTERISK-23904 Reported by: Ben Merrills Change-Id: If122602304ce03362722eb213a3111b32da5eeb9	10 years ago
Joshua Colp	17be8b9c20	Merge "install_prereq: Update repositories before install on Debian systems"	10 years ago
Rodrigo Ramírez Norambuena	88f3dbaec9	install_prereq: Update repositories before install on Debian systems ... When to install packages the indexed local is more old of the version of software on the repository they have been upgraded by security update then get the package will give 404 not found. The patch prevent by update local index to repository for aptitude before install. ASTERISK-25495 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I645959e553aac542805ced394cac2dca964051fa	10 years ago
Kevin Harwell	691c0e0b31	res_pjsip_outbound_registration: registration stops due to fatal 4xx response ... During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2	10 years ago
Matt Jordan	b9bd249a85	contrib/scripts/autosupport: Update for Asterisk 13 ... This patch adds some minor tweaks for autosupport to update it for Asterisk 13. This includes: * Finally removing most references to Zaptel * Adding support for some additional 'core' commands, and fixing nomenclature that generally hasn't been used for some time * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 (cherry picked from commit 9fc9777fa3)	10 years ago
Joshua Colp	96c26f497a	Merge "Scripts: check file versions of Asterisk and dependencies"	10 years ago
Scott Griepentrog	d6472d96b3	Scripts: check file versions of Asterisk and dependencies ... To help in diagnosing mismatched modules and libraries, this script scans for version, repository, and source information and reports what is found. ASTERISK-25376 #close Reported by: Ashley Sanders Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6	10 years ago
Mark Michelson	993ae9a669	res_pjsip: Change default from user value. ... When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190	10 years ago
Scott Griepentrog	53e2a6a829	contrib: script install_prereq should install sqlite3 ... Asterisk needs the sqlite 3 library, which is package sqlite-devel in CentOS. By adding this package to the script, a problem with configure failing is resolved. ASTERISK-25331 #close Reported by: Kevin Harwell Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec	10 years ago
Joshua Colp	309dd2a409	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Mark Michelson	2b42264e66	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d	10 years ago