asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	21d927000e	Merge "sip_to_pjsip: Write media_encryption." into 14	10 years ago
Joshua Colp	62a59d6c20	Merge "sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry." into 14	10 years ago
Mark Michelson	d974988736	ConfBridge: Rework announcer channel methodology One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5	10 years ago
Alexander Traud	94d1076e69	sip_to_pjsip: Map the TLS method correctly. When using the migration script sip_to_pjsip.py and tlsclientmethod is not set in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is offering/using not just TLSv1.0 but TLSv1.2 as well. ASTERISK-22374 Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f	10 years ago
Alexander Traud	9f86d27f60	sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent. When using the migration script sip_to_pjsip.py, no section of type=system or type=general were created. Therefore the keys compactheaders, timerb, timert1, and useragent were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1	10 years ago
Alexander Traud	019eb0454a	sip_to_pjsip: Map (session-)timers correctly. When using the migration script sip_to_pjsip.py, session-timers=accept and session-timers=refuse were mapped to wrong values. ASTERISK-22374 Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092	10 years ago
Alexander Traud	e054acc723	sip_to_pjsip: Write username even without authname. When using the migration script sip_to_pjsip.py, now the (mandatory) username is written to pjsip.conf, even if there was no (optional) authname in the register string in sip.conf. ASTERISK-22374 Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f	10 years ago
Alexander Traud	d0c1a4fd33	sip_to_pjsip: Parse register even with transport. When using the migration script sip_to_pjsip.py and the register string started with a transport in sip.conf - like tls://... - register was not parsed correctly and therefore not migrated correctly to pjsip.conf. ASTERISK-22374 Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2	10 years ago
Alexander Traud	1a556c7618	sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit. When using the migration script sip_to_pjsip.py, those keys got missing. These keys might appear several times and the function "merge_value" tried to collect those. However, because these keys have different names in sip.conf and pjsip.conf, "merge_value" was not able to find the new key name in sip.conf. This change lets "merge_value" search with the old key name in sip.conf and write with the new key name in pjsip.conf. ASTERISK-22374 Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2	10 years ago
Alexander Traud	32fa73d108	sip_to_pjsip: Map externhost/ip to Transports. When using the migration script sip_to_pjsip.py, the externhost or externip of sip.conf were erroneously written to Endpoints instead to Transports. ASTERISK-22374 Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4	10 years ago
Alexander Traud	db1715a7e7	sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry. When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and minexpiry were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b	10 years ago
Alexander Traud	527ac18ce7	sip_to_pjsip: Write media_encryption. When using the migration script sip_to_pjsip.py, encryption=yes got missing and media_encryption=sdes was not written to pjsip.conf, because of a typo. ASTERISK-22374 Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05	10 years ago
Alexander Traud	6078499b43	sip_to_pjsip: Write cos and tos. When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got missed, because of a typo. Therefore, cos and tos were not written to pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused by a copy-and-paste error. ASTERISK-22374 Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2	10 years ago
Alexander Traud	8c63306001	sip_to_pjsip: Add cert_file and ca_list_path. When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825	10 years ago
George Joseph	f87af89396	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	10 years ago
Richard Mudgett	d833ef6523	res_pjsip_session.c: Fix unbound srv failover tests. Commit `1b666549f3` broke the srv failover functionality if a TCP connection gets disconnected. Under these conditions, session_inv_on_state_changed() gets a PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new transport. Unfortunately, session_inv_on_tsx_state_changed() also gets the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates the session. * Made session_inv_on_tsx_state_changed() complete terminating the session on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still PJSIP_INV_STATE_DISCONNECTED. ASTERISK-26305 #close Reported by: Richard Mudgett Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d	10 years ago
Alexander Traud	41e9bf60aa	BuildSystem: Detect ca_list_path capabilities in external PJProject. Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the bundled PJProject but not in an external PJProject. Therefore, ca_list_path could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2 is detected again to enable ca_list_path again. ASTERISK-26303 #close Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0	10 years ago
George Joseph	d938c18733	ari: Add documentation that path parameters are case-sensitive Added to api.wiki.mustache so that the generated object pages have the notation in the table header as well as under each method that has path parameters. ASTERISK-25492 #close Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf	10 years ago
Corey Farrell	ba91710bc4	Refactor usage pattern of xmldoc info tag. This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea	10 years ago
zuul	a50303cfc4	Merge "chan_sip: Fix lastrtprx always updated" into 14	10 years ago
zuul	72b102bab8	Merge "core: Entity ID is not set or invalid" into 14	10 years ago
Joshua Colp	1ae4b3c289	Merge "res_sorcery_config.c: Cleanup ao2 container usage idioms." into 14	10 years ago
Joshua Colp	9140f52df4	Merge "sorcery.c: Minor optimizations." into 14	10 years ago
Joshua Colp	7ec6a26dd6	Merge "sorcery.c: Tweak some container declaration formatting." into 14	10 years ago
Joshua Colp	251113408a	Merge "manager: Add <see-also> tags to relate AoC events and actions" into 14	10 years ago
Joshua Colp	f21701b4a4	Merge "res_agi: Improve documentation" into 14	10 years ago
Joshua Colp	8de930b593	Merge "manager: Add <see-also> links between related events" into 14	10 years ago
Joshua Colp	5d32617a25	Merge "pbx.c: Additional fixes to ast_context_remove_extension_callerid2." into 14	10 years ago
zuul	01083e87e2	Merge "func_channel: Reorganize documentation" into 14	10 years ago
zuul	b8b9cee272	Merge "manager: Add <see-also> tags to relate UserEvent actions/apps/events" into 14	10 years ago
zuul	4b46e440df	Merge "manager: Add <see-also> tags to relate Bridge related events,actions, and apps" into 14	10 years ago
zuul	cb6b69b53e	Merge "manager: Add <see-also> tags to relate interrelated events/actions together" into 14	10 years ago
zuul	709f5f5bd4	Merge "app_dial: Improve documentation" into 14	10 years ago
cjack	19e323a6aa	chan_sip: Fix lastrtprx always updated Packets are read regulary, when there is no data in buffer fr->frametype is AST_FRAME_NULL. There was no check of frametype and lastrtprx always updated and, therefore, rtptimeout did not work at all. ASTERISK-25270 #close Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d	10 years ago
Alexei Gradinari	811e7d7788	core: Entity ID is not set or invalid The Exchanging Device and Mailbox States could not working if the Entity ID (EID) is not set manually and can't be obtained from ethernet interface. This patch replaces debug message to warning and addes missing description about option 'entityid' to asterisk.conf.sample. With this patch the asterisk also: (1) decline loading the modules which won't work without EID: res_corosync and res_pjsip_publish_asterisk. (2) warn if EID is empty on loading next modules: pbx_dundi, res_xmpp Starting with v197 systemd/udev will automatically assign "predictable" names for all local Ethernet interfaces. This patch also addes some new ethernet prefixes "eno" and "ens". ASTERISK-26164 #close Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6	10 years ago
Richard Mudgett	840d274f4e	res_sorcery_config.c: Cleanup ao2 container usage idioms. Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a	10 years ago
Richard Mudgett	2f8f063d0a	sorcery.c: Minor optimizations. * Remove some unused parameters from internal functions: sorcery_wizard_create() sorcery_wizard_update() sorcery_wizard_delete() * Created the struct sorcery_observer_invocation ao2 object without a lock since it is not needed in sorcery_observer_invocation_alloc(). * Cleanup generic ao2 container sorcery object id hash, sort, and cmp functions. Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e	10 years ago
Richard Mudgett	aed60171f1	sorcery.c: Tweak some container declaration formatting. * Tweak sorcery_object_type_alloc() formatting. * Tweak ast_sorcery_init() formatting. Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944	10 years ago
Corey Farrell	f7f9081f7a	pbx.c: Additional fixes to ast_context_remove_extension_callerid2. Do not check registrar of the first extension head. We should only check the registrar when we match the priority. Additionally fix a couple calls to strcmp which used the input callerid instead of the clean version ex.cidmatch. ASTERISK-26233 Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1	10 years ago
Matt Jordan	7c46e3bfbb	app_dial: Improve documentation * Add some helpful <literal> and other embedded paragraph tags * Document some of the lesser known channel variables set by Dial * Add examples for some common Dial uses, along with some more challenging but useful options Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1	10 years ago
Matt Jordan	2e77b973fb	manager: Add <see-also> tags to relate interrelated events/actions together Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4	10 years ago
Matt Jordan	18065140d5	manager: Add <see-also> tags to relate Bridge related events,actions, and apps Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85	10 years ago
Matt Jordan	f69ae95889	manager: Add <see-also> tags to relate AoC events and actions Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e	10 years ago
Matt Jordan	c5574667c7	manager: Add <see-also> tags to relate UserEvent actions/apps/events Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4	10 years ago
Matt Jordan	1a44ebe8ec	res_agi: Improve documentation * Groups of AGI commands that have similar functionality now reference each other, and all reference the AGI application for ease of wiki reference. * The documentation for the AGI application has been improved, in particular noting the various AGI types and how they are invoked. * A warning message has been added to DeadAGI, noting that it is deprecated. Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d	10 years ago
Matt Jordan	3b3198333a	manager: Add <see-also> links between related events This patch adds some see-also references between related AMI events. It focuses primarily on those events that are guaranteed to come in pairs, such as DTMFBegin/DTMFEnd, as well as those that occur during the life cycle of an Asterisk channel, such as Newchannel/Hangup. Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3	10 years ago
Matt Jordan	89dccb04bc	func_channel: Reorganize documentation * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b	10 years ago
Joshua Colp	2ce487585e	manager: Clarify that dialplan manipulation actions are under system class. ASTERISK-26246 #close Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657	10 years ago
zuul	0bb8366580	Merge "res_pjsip: Fail global load if debug or default_from_user are empty" into 14	10 years ago
zuul	f771e284eb	Merge "res_pjsip_caller_id: Copy header name to short header name" into 14	10 years ago

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