asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	dcebcaa3da	Merge "build: Add configure check for proto field of PJSIP TLS transport setting." into 13	9 years ago
Joshua Colp	6d309cd2cd	Merge "res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100" into 13	9 years ago
Richard Mudgett	9ae21b510f	chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full(). Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12	9 years ago
Richard Mudgett	56bcb97a3c	chan_sip.c: Simplify sip_pvt destructor call levels. Remove destructor calling destroy_it calling really_destroy_it for no benefit. Just make the destructor the really_destroy_it function. Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a	9 years ago
Joshua Colp	677a65fcbb	build: Add configure check for proto field of PJSIP TLS transport setting. Older versions of PJSIP do not have the proto field on the TLS transport setting structure. This change adds a configure check so even if it is not present we will still be able to build. Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9	9 years ago
George Joseph	32f0a3d52a	build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE I can't ever recall actually needing the intermediate files or the checking that a double compile produces. What I CAN remember is every DONT_OPTIMIZE build needing 3 invocations of gcc instead of 1 just to do the checks and produce those intermediate files. Having said that, Richard pointed out that the reason for the double compile was that there were cases in the past where a submitted patch failed to compile because the submitter never tried it with the optimizations turned on. To get the best of both worlds, COMPILE_DOUBLE has been split into its own option. If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected BUT you can then turn it off if all you need are the debugging symbols. This way you have to make an informed decision about disabling COMPILE_DOUBLE. To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature was added to menuselect. The <use> element can now contain an "autoselect" attribute which will turn the used member on but not create a hard dependency. The cflags.xml implementation for COMPILE_DOUBLE looks like this... <member name="DONT_OPTIMIZE" displayname="Disable Optimizations ..."> <use autoselect="yes">COMPILE_DOUBLE</use> <support_level>core</support_level> </member> <member name="COMPILE_DOUBLE" displayname="Pre-compile with ...> <depend>DONT_OPTIMIZE</depend> <support_level>core</support_level> </member> When DONT_OPTIMIZE is turned on, COMPILE_DOUBLE is turned on because of the use. When DONT_OPTIMIZE is turned off, COMPILE_DOUBLE is turned off because of the depend. When COMPILE_DOUBLE is turned on, DONT_OPTIMIZE is turned on because of the depend. When COMPILE_DOUBLE is turned off, DONT_OPTIMIZE is left as is because it only uses COMPILE_DOUBLE, it doesn't depend on it. I also made a few tweaks to the ncurses implementation to move things left a bit to allow longer descriptions. Change-Id: Id49ca930ac4b5ec4fc2d8141979ad888da7b1611	9 years ago
George Joseph	38499e7125	pjproject: Pass (dont_)optimize flags to pjproject and fix pjsua The pjproject Makefile now uses the Asterisk optimization flags which are determined by the setting of the DONT_OPTMIZE menuselect flag. The Makefile was also restructured so a change to the top level menuselect.makeopts will result in a rebuild of pjproject. Also, "--disable-resample" was removed from the pjproject configure options. Without resample, pjsua (which is used by the testsuite) can't make audio calls. When it can't, it segfaults. Change-Id: I24b0a4d0872acef00ed89b3c527a713ee4c2ccd4	9 years ago
Walter Doekes	336cae73cc	app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. Channel masquerading had a conflict with autochannel locking. When locking autochannel->channel, the channel is fetched from the autochannel and then locked. During the fetch, the autochannel -- which has no locks itself -- can be modified by someone who owns the channel lock. That means that the value of autochan->channel cannot be trusted until you hold the lock. In practice, this caused problems with Local channels getting masqueraded away while the ChanSpy attempted to get info from that channel. The old channel which was about to get removed got locked, but the new (replaced) channel got unlocked (no-op). Because the replaced channel was now locked (and would never get unlocked), it couldn't get removed from the channel list in a timely manner, and would now cause deadlocks when iterating over the channel list. This change checks the autochannel after locking the channel for changes to the autochannel. If the channel had been changed, the lock is reobtained on the new channel. In theory it seems possible that after this fix, the lock attempt on the old (wrong) channel can be on an already destroyed lock, maybe causing a crash. But that hasn't been observed in the wild and is harder induce than the current deadlock. Thanks go to Filip Frank for suggesting a fix similar to this and especially to IRC user hexanol for pointing out why this deadlock was possible and testing this fix. And to Richard for catching my rookie while loop mistake ;) ASTERISK-25321 #close Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def	9 years ago
Joshua Colp	ed34bbdf9b	Merge "install_prereq: Add packages for bundled pjproject" into 13	9 years ago
zuul	9263ea0b13	Merge "res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited" into 13	9 years ago
zuul	f74ac9d5b6	Merge "pjproject_bundled: Remove --with-external-pa from configure options." into 13	9 years ago
George Joseph	875d5e9872	pjproject_bundled: Remove --with-external-pa from configure options. Not sure why it was there in the first place as we already specify --disable-sound. Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9	9 years ago
George Joseph	530cff5f5f	res_pjsip: Strip spaces from items parsed from comma-separated lists Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(&copy, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2	9 years ago
George Joseph	3c8076a83b	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
zuul	6e58f83d8d	Merge "third_party/Makefile.rules: Replace unsupported != operator with $(shell ...)" into 13	9 years ago
Joshua Colp	cb70ed7214	Merge "config_transport: Fix objects returned by ast_sip_get_transport_states" into 13	9 years ago
zuul	772036b525	Merge "alembic: Fix downgrade and tweak for sqlite" into 13	9 years ago
George Joseph	27f32cd0a6	res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited Per RFC3325, the 'From' header is now anonymized on outgoing calls when caller id presentation is prohibited. TID = trust_id_outbound PRO = Set(CALLERID(pres)=prohib) USR = endpoint/from_user DOM = endpoint/from_domain PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) Conditions \|Result --------------------\|---------------------------------------------------- TID PRO USR DOM \|PAI FROM --------------------\|---------------------------------------------------- Y Y abc def.ghi \|PRI "Anonymous" <sip:abc@def.ghi> Y Y abc \|PRI "Anonymous" <sip:abc@anonymous.invalid> Y Y def.ghi \|PRI "Anonymous" <sip:anonymous@def.ghi> Y Y \|PRI "Anonymous" <sip:anonymous@anonymous.invalid> Y N abc def.ghi \|YES <sip:abc@def.ghi> Y N abc \|YES <sip:abc@<ip_address>> Y N def.ghi \|YES "Caller Name" <sip:<caller_exten>@def.ghi> Y N \|YES "Caller Name" <sip:<caller_exten>@<ip_address>> N Y abc def.ghi \|NO "Anonymous" <sip:abc@def.ghi> N Y abc \|NO "Anonymous" <sip:abc@anonymous.invalid> N Y def.ghi \|NO "Anonymous" <sip:anonymous@def.ghi> N Y \|NO "Anonymous" <sip:anonymous@anonymous.invalid> N N abc def.ghi \|YES <sip:abc@def.ghi> N N abc \|YES <sip:abc@<ip_address>> N N def.ghi \|YES "Caller Name" <sip:<caller_exten>@def.ghi> N N \|YES "Caller Name" <sip:<caller_exten>@<ip_address>> ASTERISK-25791 #close Reported-by: Anthony Messina Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9	9 years ago
zuul	9b0a96b947	Merge "loader: Retry dlopen when loading fails" into 13	9 years ago
George Joseph	7cf7b0a4f9	third_party/Makefile.rules: Replace unsupported != operator with $(shell ...) Apparently the != operator is fairly new so I've replaced it with the old $(shell ...) syntax. Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479 Reported-by: Richard Mudgett	9 years ago
zuul	d2be16472e	Merge "bridge.c: Crash during attended transfer when missing a local channel half" into 13	9 years ago
George Joseph	53f57001f2	loader: Retry dlopen when loading fails Although we use the RTLD_LAZY flag when calling dlopen the first time on a module, this only defers resolution for function calls. Pointer references to functions are determined at link time so dlopen expects them to be there. Since we don't cross-module link, pointers to functions in other modules won't be available and dlopen will fail. Doing a "hardened" build also causes problems because it typically sets "-z now" on the ld command line which overrides RTLD_LAZY at run time. If the failing module isn't a GLOBAL_SYMBOLS module, then dlopen will be called again after all the GLOBAL_SYMBOLS modules have been loaded and they'll eventually resolve. If the calling module IS a GLOBAL_SYMBOLS module itself and a third module depends on it, then there's an issue because the second time through the dlopen loop, GLOBAL_SYMBOLS modules aren't given any special treatment and since the order in which dlopen is called isn't deterministic, the dependent may again be tried before the module it needs is loaded. Simple solution: Save modules that fail load_resource because of a dlopen error in a list and retry them immediately after the first pass. Keep retrying until the failed list is empty or we reach a #defined max retries. Error messages are suppressed until the final pass which also gets rid of those confusing error messages about module failures that are later corrected. Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb	9 years ago
zuul	039ea76a8a	Merge "res_pjsip_dtmf_info: NULL terminate the message body." into 13	9 years ago
Kevin Harwell	40d9e9e238	bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880	9 years ago
Kevin Harwell	ff3da61c35	res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 During the transfer process, some phones (okay it was the Jitsi softphone, but maybe others are out there) send a "bye" immediately after receiving a SIP Notify. When a "bye" is received early for some types of transfers the transferer channel may no longer be available during late stage transfer processing. For instance, during an attended transfer involving stasis bridging at one point the created local channel looks for an associated swap channel in order to retrieve the stasis application name. If the transferer has hung up then the local channel will fail to find it. The local channel then has no way to know which stasis app to enter, so it fails and hangs up as well. Thus the transfer does not complete as expected. This patch delays the sending of the initial notify in order to give the transfer process enough time to gather the necessary data for a successful transfer. ASTERISK-25771 Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16	9 years ago
zuul	9e896540c8	Merge "build-system: Allow building with static pjproject" into 13	9 years ago
Joshua Colp	26b8f2692e	res_pjsip_dtmf_info: NULL terminate the message body. PJSIP does not ensure that when printing the message body the buffer will be NULL terminated. This is problematic when searching for the signal and duration values of the DTMF. This change ensures the buffer is always NULL terminated. Change-Id: I52653a1a60c93092d06af31a27408d569cc98968	9 years ago
Joshua Colp	2dc79e13be	Merge "func_callerid.c: Update REDIRECTING reason documentation." into 13	9 years ago
Joshua Colp	86124f63c8	Merge "SIP diversion: Fix REDIRECTING(reason) value inconsistencies." into 13	9 years ago
Joshua Colp	3b6b164f2e	Merge "res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref." into 13	9 years ago
zuul	d6e274b97d	Merge "res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason." into 13	9 years ago
Joshua Colp	f4670c6a76	Merge "CHAOS: cleanup possible null vars on msg alloc failure" into 13	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	6f0d7ce9db	config_transport: Fix objects returned by ast_sip_get_transport_states ast_sip_get_transport_states was returning a container of internal_state objects instead of ast_sip_transport_state objects. This was causing transport lookups to fail, most noticably in res_pjsip_nat, which couldn't find the correct external addresses. This was causing contacts to go out with internal ip addresses. ASTERISK-25830 #close Reported-by: Sean Bright Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e	9 years ago
Scott Griepentrog	1ea7a5a774	CHAOS: cleanup possible null vars on msg alloc failure In message.c, if msg_alloc fails to init the string field, vars may be null, so use a null tolerant cleanup. In res_pjsip_messaging.c, if msg_data_create fails, mdata will be null, so use a null tolerant cleanup. ASTERISK-25323 Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56	9 years ago
Scott Griepentrog	3c37c7071f	CHAOS: prevent crash on failed strdup This patch avoids crashing on a null pointer if the strdup() allocation fails. ASTERISK-25323 Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5	9 years ago
Richard Mudgett	9633be9d25	func_callerid.c: Update REDIRECTING reason documentation. Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386	9 years ago
Richard Mudgett	4165ea7778	SIP diversion: Fix REDIRECTING(reason) value inconsistencies. Previous chan_sip behavior: Before this patch chan_sip would always strip any quotes from an incoming reason and pass that value up as the REDIRECTING(reason). For an outgoing reason value, chan_sip would check the value against known values and quote any it didn't recognize. Incoming 480 response message reason text was just assigned to the REDIRECTING(reason). Previous chan_pjsip behavior: Before this patch chan_pjsip would always pass the incoming reason value up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip would send the reason value as passed down. With this patch: Both channel drivers match incoming reason values with values documented by REDIRECTING(reason) and values documented by RFC5806 regardless of whether they are quoted or not. RFC5806 values are mapped to the equivalent REDIRECTING(reason) documented value and is set in REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a quoted string version ('"unconditional"') is converted to REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal with 'cfu' instead of any of the aliases. The incoming 480 response reason text supported by chan_sip checks for known reason values and if not matched then puts quotes around the reason string and assigns that to REDIRECTING(reason). Both channel drivers send outgoing known REDIRECTING(reason) values as the unquoted RFC5806 equivalent. User custom values are either sent as is or with added quotes if SIP doesn't allow a character within the value as part of a RFC3261 Section 25.1 token. Note that there are still limitations on what characters can be put in a custom user value. e.g., embedding quotes in the middle of the reason string is silly and just going to cause you grief. * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the 'cfu' value. * Added missing malloc() NULL return check in res_pjsip_diversion.c set_redirecting_reason(). * Fixed potential read from a stale pointer in res_pjsip_diversion.c add_diversion_header(). The reason string needed to be copied into the tdata memory pool to ensure that the string would always be available. Otherwise, if the reason string returned by reason_code_to_str() was a user's reason string then the string could be freed later by another thread. Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87	9 years ago
Richard Mudgett	41f4af4ce5	res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd	9 years ago
Richard Mudgett	4c5998ff55	res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. * Fix double unref of other_party channel in off nominal path. * This is unlikely to be a real problem. However, for safety, in handle_incoming_request() keep the datastore ref with the other_party channel ref until we are finished with the other_party channel. Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821	9 years ago
George Joseph	b59956a875	build-system: Allow building with static pjproject Background here: http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html From CHANGES: * To help insure that Asterisk is compiled and run with the same known version of pjproject, a new option (--with-pjproject-bundled) has been added to ./configure. When specified, the version of pjproject specified in third-party/versions.mak will be downloaded and configured. When you make Asterisk, the build process will also automatically build pjproject and Asterisk will be statically linked to it. Once a particular version of pjproject is configured and built, it won't be configured or built again unless you run a 'make distclean'. To facilitate testing, when 'make install' is run, the pjsua and pjsystest utilities and the pjproject python bindings will be installed in ASTDATADIR/third-party/pjproject. The default behavior remains building with the shared pjproject installation, if any. Building: All you have to do is include the --with-pjproject-bundled option on the ./configure command line (and remove any existing --with-pjproject option if specified). Everything else is automatic. Behind the scenes: The top-level Makefile was modified to include 'third-party' in the list of MOD_SUBDIRS. The third-party directory was created to contain any third party packages that may be needed in the future. Its Makefile automatically iterates over any subdirectories passing on targets. The third-party/pjproject directory was created to house the pjproject source distribution. Its Makefile contains targets to download, patch configure, generate dependencies, compile libs, apps and python bindings, sanitized build.mak and generate a symbols list. When bootstrap.sh is run, it automatically includes the configure.m4 file in third-party/pjproject. This file has a macro to download and conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR and PJPROJECT_BUNDLED. It also tests for the capabilities like PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to trying to compile. Of course, bootstrap.sh is only run once and the configure file is incldued in the patch. When configure is run with the new options, the macro in configure.m4 triggers the download, patch, conifgure and tests. No compilation is performed at this time. The downloaded tarball is cached in /tmp so it doesn't get downloaded again on a distclean. When make is run in the top-level Asterisk source directory, it will automatically descend all the subdirectories in third_party just as it does for addons, apps, etc. The top-level Makefile makes sure that the 'third-party' is built before 'main' so that dependencies from the other directories are built first. When main does build, a new shared library (libasteriskpj) is created that links statically to the pjproject .a files and exports all their symbols. The asterisk binary links to that, just as it does with libasteriskssl. When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject python bindings are installed in ASTDATADIR/third-party/pjproject. This will facilitate testing, including running the testsuite which will be updated to check that directory for the pjsua module ahead of the system python library. Modules should continue to depend on pjproject if they use pjproject APIs directly. They should not care about the implementation. No changes to any res_pjsip modules were made. Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103	9 years ago
zuul	d1495bc0eb	Merge "chan_sip.c: Fix T.38 issues caused by leaving a bridge." into 13	9 years ago
zuul	84d1036205	Merge "res_pjsip_t38.c: Back out part of an earlier fix attempt." into 13	9 years ago
zuul	5393a4963a	Merge "bridge core: Add owed T.38 terminate when channel leaves a bridge." into 13	9 years ago
zuul	dbf52dd7d7	Merge "channel api: Create is_t38_active accessor functions." into 13	9 years ago
zuul	4c85d7b612	Merge "bridge_channel: Don't settle owed events on an optimization." into 13	9 years ago
zuul	94f3198c90	Merge "channel.c: Route all control frames to a channel through the same code." into 13	9 years ago
zuul	414e297648	Merge "res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s." into 13	9 years ago
Richard Mudgett	18a323e542	chan_sip.c: Fix T.38 issues caused by leaving a bridge. chan_sip could not handle AST_T38_TERMINATED frames being sent to it when the channel left the bridge. The action resulted in overlapping outgoing reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not happy. * Force T.38 to be remembered as locally bridged. Now when the channel leaves the native RTP bridge after T.38, the channel remembers that it has already reINVITEed the media back to Asterisk. It just needs to terminate T.38 when the AST_T38_TERMINATED arrives. * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if they happen before the T.38 state changes to disabled. Now the T.38 state is set to disabled before the reINVITE is sent. ASTERISK-25582 #close Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce	9 years ago
Richard Mudgett	263a39f2cc	res_pjsip_t38.c: Back out part of an earlier fix attempt. This backs out item 4 of the `4875e5ac32` commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame that it puts into the bridge may or may not be processed by the time the bridged peer is kicked out of the bridge. If it is processed then all is well. However, if it is not processed then that channel is stuck in fax mode until it hangs up or maybe if it joins another bridge for T.38 faxing. ASTERISK-25582 Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7	9 years ago

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