28598 Commits (1a3e699316b672ade3890f6055429f132849c0ac)

 

All Branches   

Search

Author	SHA1	Message	Date
zuul	330ea54a92	Merge "sip_to_pjsip: Add cert_file." into 14	9 years ago
zuul	1866fa9d06	Merge "res_format_attr_g729: Add annexb=no format parameter to SDPs" into 14	9 years ago
zuul	b1600bccf5	Merge "res_pjsip: Add contact_user to endpoint" into 14	9 years ago
zuul	7952f7f478	Merge "ari: Add documentation that path parameters are case-sensitive" into 14	9 years ago
Alexander Traud	9ca84fa2e8	sip_to_pjsip: Add cert_file. ... When using the migration script sip_to_pjsip.py, cert_file was not migrated to pjsip.conf. A previous change regarding this contained a copy/paste error. ASTERISK-22374 Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b	9 years ago
Alexander Traud	d192cd125c	sip.conf: tlsclientmethod is using sslv23 as default. ... When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL SSLv23_method. This was documented incorrectly in the file sip.conf.sample. SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that function should have been called 'secure_method' or 'automatic_method' back in the 90s. Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically. ASTERISK-24425 Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3	9 years ago
Joshua Colp	4437db8d89	Merge "sip_to_pjsip: Write cos and tos." into 14	9 years ago
Kevin Harwell	d2bee6b535	res_format_attr_g729: Add annexb=no format parameter to SDPs ... Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0	9 years ago
Kevin Harwell	03c7e5e1ea	rest-api: Swagger scripts were not replacing format variable in file brief ... Given resource paths did not have 'json' substituted in for the '{format}'. For some auto generated documentation/comment strings it resulted in something like the following: "... REST handler for /api-docs/sounds.{format}" This patch makes sure the resource api's path is properly substituted. ASTERISK-25472 #close Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23	9 years ago
George Joseph	85a1de9f72	res_odbc: Correct the dependency relationship with res_odbc_transaction ... The MODULEINFO dependencies between these 2 modules was reversed. res_odbc should depend on res_odbc_transaction, not the other way around. ASTERISK-25984 #close Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f	9 years ago
Kevin Harwell	81e3b8f141	sip_to_pjsip: Set correct tls transport method ... A recent update had a copy/paste error where the unused variable 'val' was being passed to the set_value function instead of the 'method' value itself. This patch passes in the right variable. ASTERISK-22374 Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06	9 years ago
Joshua Colp	e251482d88	Merge "sip_to_pjsip: Parse register even with transport." into 14	9 years ago
Joshua Colp	f9e2be625b	Merge "sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit." into 14	9 years ago
Joshua Colp	0c315e58e6	Merge "sip_to_pjsip: Map (session-)timers correctly." into 14	9 years ago
Joshua Colp	78ffb5d89c	Merge "sip_to_pjsip: Add cert_file and ca_list_path." into 14	9 years ago
Joshua Colp	122749fe1a	Merge "sip_to_pjsip: Write username even without authname." into 14	9 years ago
Joshua Colp	c61dbc6318	Merge "sip_to_pjsip: Map the TLS method correctly." into 14	9 years ago
Joshua Colp	55a81c3a12	Merge "sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent." into 14	9 years ago
Joshua Colp	21d927000e	Merge "sip_to_pjsip: Write media_encryption." into 14	9 years ago
Joshua Colp	62a59d6c20	Merge "sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry." into 14	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 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	9 years ago
zuul	a50303cfc4	Merge "chan_sip: Fix lastrtprx always updated" into 14	9 years ago
zuul	72b102bab8	Merge "core: Entity ID is not set or invalid" into 14	9 years ago
Joshua Colp	1ae4b3c289	Merge "res_sorcery_config.c: Cleanup ao2 container usage idioms." into 14	9 years ago
Joshua Colp	9140f52df4	Merge "sorcery.c: Minor optimizations." into 14	9 years ago
Joshua Colp	7ec6a26dd6	Merge "sorcery.c: Tweak some container declaration formatting." into 14	9 years ago
Joshua Colp	251113408a	Merge "manager: Add <see-also> tags to relate AoC events and actions" into 14	9 years ago
Joshua Colp	f21701b4a4	Merge "res_agi: Improve documentation" into 14	9 years ago
Joshua Colp	8de930b593	Merge "manager: Add <see-also> links between related events" into 14	9 years ago
Joshua Colp	5d32617a25	Merge "pbx.c: Additional fixes to ast_context_remove_extension_callerid2." into 14	9 years ago
zuul	01083e87e2	Merge "func_channel: Reorganize documentation" into 14	9 years ago
zuul	b8b9cee272	Merge "manager: Add <see-also> tags to relate UserEvent actions/apps/events" into 14	9 years ago
zuul	4b46e440df	Merge "manager: Add <see-also> tags to relate Bridge related events,actions, and apps" into 14	9 years ago
zuul	cb6b69b53e	Merge "manager: Add <see-also> tags to relate interrelated events/actions together" into 14	9 years ago