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2016-08-25res_fax: Fix deadlock setting FAXMODE channel variable.Richard Mudgett
ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. Unfortunately, it also introduced a deadlock potential because set_channel_variables() which sets FAXMODE can be called during a masquerade. The ast_channel_get_t38_state() which gets the value used to set FAXMODE cannot be called with the channel locked. As a result, local channels can deadlock because of how they must acquire the locks necessary to operate. The intent of FAXMODE is for dialplan to know how a fax was transferred after the fax completes. However, the previous patch sets FAXMODE to the channel's current T.38 state AFTER the fax has completed and where T.38 may have already disconnected. * Set FAXMODE based upon T.38 negotiations exchanged either with the fax applications or the fax framehooks. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1
2016-08-25res_fax.c: Fix deadlock in fax_gateway_indicate_t38().Richard Mudgett
fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be called with any channel locks already held. A deadlock can happen if the function is operating on a local channel. * Made fax_gateway_indicate_t38() unlock the channel before calling ast_indicate_data() since fax_gateway_indicate_t38() is always called with the channel locked. * Made fax_gateway_indicate_t38() return void since nothing cared about its return value. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407
2016-08-25res_fax.c: Add chan locked precondition comments.Richard Mudgett
Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7
2016-08-25ast_framehook_detach() must be called with the channel locked.Richard Mudgett
The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584
2016-08-25ast_framehook_attach() must be called with the channel locked.Richard Mudgett
The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438
2016-08-25chan_iax2: Set plaintext auth to deprecated as per ASTERISK-22820varnav
Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth is not supported in IAX2 protocol. Please refer to section 8.6.13 of RFC 5456. But plaintext auth is still supported by Asterisk implementation of IAX2. This support should be dropped. Patch, based on asterisk-dev discussion, adds deprecation warning on startup if 'auth' is set to 'plaintext', changes default values of 'auth' from 'md5, plaintext' to 'md5'. Patch is safe in terms of backwards compatibility, will work even if remote peers have auth=plaintext and we have defaults. auth=plaintext setting will remain deprecated in Asterisk 14 and 15, and IAX2 plaintext support will be removed in Asterisk 16. ASTERISK-22820 #close Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf
2016-08-24res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_optionsGeorge Joseph
ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362
2016-08-23ConfBridge: Rework announcer channel methodologyMark Michelson
NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. 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: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0
2016-08-23Merge "Revert "ConfBridge: Rework announcer channel methodology""Joshua Colp
2016-08-23Revert "ConfBridge: Rework announcer channel methodology"Joshua Colp
This reverts commit 5aa877305223faab5a1119276a934893ab9dc138. Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491
2016-08-22Merge "ConfBridge: Rework announcer channel methodology"zuul
2016-08-22Merge "followme: initialize all config items on reload"zuul
2016-08-22Merge "compilation failed with -Werror=maybe-uninitialized"zuul
2016-08-22Merge "res_odbc_transaction: add dep on generic_odbc"zuul
2016-08-22Merge "pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations."Joshua Colp
2016-08-22compilation failed with -Werror=maybe-uninitializedAlexei Gradinari
The compilation failed for devmode --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable DO_CRASH --enable TEST_FRAMEWORK res_pjsip/pjsip_configuration.c: In function dtls_handler: res_pjsip/pjsip_configuration.c:974:20: error: back may be used uninitialized in this function [-Werror=maybe-uninitialized] int size = strlen(front); ^ cc1: all warnings being treated as errors Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580
2016-08-22Merge "res_ari: Add http prefix to generated docs"zuul
2016-08-21res_odbc_transaction: add dep on generic_odbcDavid M. Lee
When res_odbc_transaction depended on res_odbc, it got the generic_odbc headers and libs implicitly. Now that it no longer depends on res_odbc, its dependency on generic_odbc must be explicit. Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911
2016-08-20pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations.Alexander Traud
PJProject supports a lot of platforms even Windows, some with different defaults when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS, "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows. Because of this, if configured with just an IPv6 address/transport, PJProject listens to both IPv4 and IPv6. However, this is not supported by the PJProject team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP, incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only server which accepts incoming connections on IPv4. If you try to configure two transports, one with IPv4 and one with IPv6 on the same interface, as expected by the PJProject team, the IPv4 transport is not able to bind because the IPv6 transport listens to both already. One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide. Then, you are able to configure two transports, one for each IP version on the same interface. That way, you get a server which works with IPv4 clients and IPv6 clients at the same time over the same interface. Here, this change sets this parameter directly within PJProject to match the expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack servers out of the box like in chan_sip. This change was accepted by the PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected to arrive in the next version, PJProject 2.6.0. Until then, this change is incorporated in the bundled PJProject of Asterisk. ASTERISK-26309 Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae
2016-08-19Merge "sip_to_pjsip: Map externhost/ip to Transports."zuul
2016-08-19res_ari: Add http prefix to generated docsTorrey Searle
updated the uri handler to include the url prefix of the http server this enables res_ari to add it to the uris when generating docs Change-Id: I279335a2625261a8492206c37219698f42591c2e (cherry picked from commit 6f448f32fe9b7379e2630fab7b06205f901f2ded)
2016-08-19Merge "res_odbc: Correct the dependency relationship with res_odbc_transaction"zuul
2016-08-19Merge "sip.conf: tlsclientmethod is using sslv23 as default."zuul
2016-08-19Merge "rest-api: Swagger scripts were not replacing format variable in file ↵zuul
brief"
2016-08-19Merge "sip_to_pjsip: Add cert_file."zuul
2016-08-19Merge "res_format_attr_g729: Add annexb=no format parameter to SDPs"zuul
2016-08-19Merge "res_pjsip: Add contact_user to endpoint"zuul
2016-08-19Merge "ari: Add documentation that path parameters are case-sensitive"zuul
2016-08-19sip_to_pjsip: Add cert_file.Alexander Traud
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
2016-08-19sip.conf: tlsclientmethod is using sslv23 as default.Alexander Traud
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
2016-08-18Merge "sip_to_pjsip: Write cos and tos."Joshua Colp
2016-08-18res_format_attr_g729: Add annexb=no format parameter to SDPsKevin Harwell
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
2016-08-18rest-api: Swagger scripts were not replacing format variable in file briefKevin Harwell
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
2016-08-18res_odbc: Correct the dependency relationship with res_odbc_transactionGeorge Joseph
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
2016-08-18sip_to_pjsip: Set correct tls transport methodKevin Harwell
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
2016-08-18Merge "res_pjsip_session.c: Fix unbound srv failover tests."zuul
2016-08-18Merge "sip_to_pjsip: Parse register even with transport."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Write local_net, contact_acl, contact_deny, and ↵Joshua Colp
contact_permit."
2016-08-18Merge "sip_to_pjsip: Map (session-)timers correctly."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Add cert_file and ca_list_path."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Write username even without authname."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Map the TLS method correctly."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Write media_encryption."Joshua Colp
2016-08-18Merge "sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry."Joshua Colp
2016-08-18ConfBridge: Rework announcer channel methodologyMark Michelson
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
2016-08-18sip_to_pjsip: Map the TLS method correctly.Alexander Traud
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
2016-08-18sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent.Alexander Traud
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
2016-08-18sip_to_pjsip: Map (session-)timers correctly.Alexander Traud
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
2016-08-18sip_to_pjsip: Write username even without authname.Alexander Traud
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
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-04 19:24:05 +0000