86 Commits (b82f58d2bfa2b64efca087b66f7f6c916ef29c78)

Author	SHA1	Message	Date
Joshua Colp	b3f4a6365e	pjsip: Add Alembic for PUBLISH support. ... This change adds database tables for the PUBLISH support so it can be configured using realtime. A minor fix to the res_pjsip_publish_asterisk module was done so that it read the sorcery configuration from the correct section. Finally the sample configuration files have been updated. ASTERISK-26928 Change-Id: I81991ae5c75af98d247f7eacd1c0b0a763675952	8 years ago
Sean Bright	864dda07f3	alembic: Turn off execute bit on non-executable python scripts ... Change-Id: I744c986da4a38aeff8c00837eb89de7841fbc86c	8 years ago
Richard Begg	398e5ec16c	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	8 years ago
George Joseph	9b756662a8	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	8 years ago
Mark Michelson	7bc69753bc	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	8 years ago
Matt Jordan	b3c2c996f1	res_pjsip_endpoint_identifier_ip: Add an option to match requests by header ... This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit 30f52d79d7)	8 years ago
Joshua Colp	37aaaa2da2	res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. ... This change implements SRV support for the IP based endpoint identifier module. All possible addresses through SRV are looked up and added as matches. If no SRV records are available a fallback to normal host resolution is done. If an IP address is provided then no SRV lookup occurs. This is configured using the "srv_lookups" option on the identify section and defaults to "yes". ASTERISK-26693 Change-Id: I6b641e275bf96629320efa8b479737062aed82ac	8 years ago
Joshua Colp	e0bc17edff	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. ... When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
Rodrigo Ramírez Norambuena	29b7a5b00f	Add text of cdr directory into README.md for ast-db-manage ... Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636	9 years ago
George Joseph	86550f9c17	alembic: Allow cdr, config and voicemail to exist in the same schema ... cdr, config and voicemail are all separate alembic trees. Because alembic's default is to use a table named 'alembic_version' to store the current tree revision, the 3 trees can't exist in the same schema without stepping on each other. Now each tree uses 'alembic_version_<tree_name>' as the version table. Each tree's env.py script now first checks for 'alembic_version'. If it finds it AND its revision is in the tree's history, the script renames it to 'alembic_version_<tree_name>'. Regardless, the script then continues with the migration using 'alembic_version_<tree_name>' and creates that table if it's not found. The result is that if an existing 'alembic_version' table was found but it didn't belong to this tree, it's left alone and 'alembic_version_<tree_name>' is used or created. WARNING: If multiple trees are using the same schema, they MUST NOT CRU or D any objects with names that might exist in the other trees. An example would be 'yesno_values' type. If two trees perform operations on it, one tree could pull it out from under the other. Thankfully we currently don't share any names among cdr, config and voicemail. NOTE: Since the env.py scripts in each tree were identical, a common env.py has been placed in the ast-db-manage directory and a symlink to it has been placed in each tree directory. ASTERISK-24311 #close Reported-by: Dafi Ni Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898	9 years ago
Richard Mudgett	30af92e78d	res_pjsip: Add ignore_uri_user_options option. ... This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	9 years ago
George Joseph	329507fe20	res_pjsip: Add contact_user to endpoint ... contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	9 years ago
Kevin Harwell	5913929d31	alembic: add auth_username to endpoint's identify_by enum ... A new identify_by option was added recently, auth_username. However, this setting was not added as an allowable choice in the database enumeration value. This patch updates the current enumeration, adding in the new setting. ASTERISK-26268 #close Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8	9 years ago
Kevin Harwell	f6ec94cca6	alembic/sqlalchemy: auto increment only allowed on a single column ... The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b	9 years ago
Alexei Gradinari	a06a1af0eb	res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack ... The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a	9 years ago
Mark Michelson	eda95236d1	Fix sqlalchemy error regarding identifier length. ... sqlalchemy was complaining: sqlalchemy.exc.IdentifierError: Identifier 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 characters This fixes the problem by changing the index name to be "ps_contacts_qualifyfreq_exp" instead. ASTERISK-26227 #close Reported by Mark Michelson Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9	9 years ago
Richard Mudgett	851b1c3a17	res_pjsip: Add fax_detect_timeout endpoint option. ... The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	9 years ago
Alexei Gradinari	2c16a81dd5	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	7ed680cf92	Merge "res_pjsip: improve realtime performance #2" into 13	9 years ago
Mark Michelson	1b79e2deff	Fix Alembic upgrades. ... A non-existent constraint was being referenced in the upgrade script. This patch corrects the problem by removing the reference. 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 updates Sorcery to use "yes" and "no" ASTERISK-26128 #close Change-Id: I366dbbf91418a9cb160b3ca74b0e59b5ac284bec	9 years ago
Alexei Gradinari	b3c787d1dd	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
Joshua Colp	cda3385409	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	230686f4ec	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	524a302974	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
zuul	a01ce2b889	Merge "res_pjsip: improve realtime performance" into 13	9 years ago
Alexei Gradinari	9c2032240e	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	fe38d21c2a	pjsip: Added "reg_server" to contacts (fixed alembic) ... ASTERISK-25931 Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549	9 years ago
Alexei Gradinari	3cb8934de0	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	38bed4515d	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	df3639700a	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	d312fdeb1b	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
George Joseph	d8f0bc3572	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	5aa5c49413	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
George Joseph	86d6e44cc1	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	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	9 years ago
George Joseph	953d1cc11a	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	a85fab7c44	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	d604a9afc8	Fix alembic branches on v13. ... Change-Id: I313449b609ede18ad1e1763a655dd23b9210a8e0	9 years ago
Joshua Colp	e7b83fa76e	Merge "pjsip/alembic: Fix qualify_timeout column definition" into 13	9 years ago
Joshua Colp	236896f391	Merge "pjsip: Add option global/regcontext" into 13	9 years ago
Daniel Journo	46f21df302	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
George Joseph	219c204a41	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
Daniel Journo	22801a06ee	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
Mark Michelson	53bd5a539a	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
Mark Michelson	da17dc4d75	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	10 years ago
Kevin Harwell	c58091737d	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
Mark Michelson	ac62928d6b	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
Joshua Colp	2749721791	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	d9094ddd73	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: I06660ba672c0a343814af4cec838e6025cafd54b	10 years ago
Kevin Harwell	31c77b157b	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. ... Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	10 years ago