495 Commits (d108bee11bcc5b529cc118fd423dcd31cdfa2ed0)

Author	SHA1	Message	Date
George Joseph	471ff375fd	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
George Joseph	b8b7c2e428	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	ba8adb4ce3	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	9b13ab6a63	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	52b29f9b4c	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	10 years ago
Richard Mudgett	a4dcbdf50f	Fix alembic branches on master. ... Change-Id: I64ed21fec50eb833641ca49d92184f6aaabd86e8	10 years ago
Joshua Colp	2cdbd4d711	Merge "pjsip/alembic: Fix qualify_timeout column definition"	10 years ago
Daniel Journo	c60d6c0162	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	10 years ago
Daniel Journo	8182146e85	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	10 years ago
Richard Mudgett	1fffe71f77	res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts". ... PJPROJECT has a function available to dump the compile time options used when building the library. * Add CLI "pjsip show buildopts" command. * Update contrib/scripts/autosupport to get pjproject information. Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748	10 years ago
George Joseph	a41aab477a	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	10 years ago
Joshua Colp	e106292a92	Merge "Alembic: Increase column size of PJSIP AOR "contact"."	10 years ago
Joshua Colp	8bd9e2dcc0	Merge "Alembic: Add PJSIP global keep_alive_interval."	10 years ago
Mark Michelson	c5e16fe33a	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
Walter Doekes	64b2046f3d	Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts. ... The spandspflow2pcap.py creates pcap files from fax.log files, generated through 'fax set debug on' when receiving a fax. An example fax.log is included as spandspflow2pcap.log. The sipp-sendfax.xml SIPp scenario can be used to replay that fax with a recent version of SIPp. ASTERISK-25660 #close Change-Id: I4de8f28b084055b482ab8a5b28d28b605b0ed526	10 years ago
Mark Michelson	cfb34adb83	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
Corey Farrell	cd5ae02812	Increase account code maximum length to 80. ... This increases the maximum length of account code's to match extensions. This ensures it is always possible to set an accountcode to ${EXTEN} without truncation. ASTERISK-23904 Reported by: Ben Merrills Change-Id: If122602304ce03362722eb213a3111b32da5eeb9	10 years ago
Joshua Colp	17be8b9c20	Merge "install_prereq: Update repositories before install on Debian systems"	10 years ago
Rodrigo Ramírez Norambuena	88f3dbaec9	install_prereq: Update repositories before install on Debian systems ... When to install packages the indexed local is more old of the version of software on the repository they have been upgraded by security update then get the package will give 404 not found. The patch prevent by update local index to repository for aptitude before install. ASTERISK-25495 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I645959e553aac542805ced394cac2dca964051fa	10 years ago
Kevin Harwell	691c0e0b31	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
Matt Jordan	b9bd249a85	contrib/scripts/autosupport: Update for Asterisk 13 ... This patch adds some minor tweaks for autosupport to update it for Asterisk 13. This includes: * Finally removing most references to Zaptel * Adding support for some additional 'core' commands, and fixing nomenclature that generally hasn't been used for some time * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 (cherry picked from commit 9fc9777fa3)	10 years ago
Joshua Colp	96c26f497a	Merge "Scripts: check file versions of Asterisk and dependencies"	10 years ago
Scott Griepentrog	d6472d96b3	Scripts: check file versions of Asterisk and dependencies ... To help in diagnosing mismatched modules and libraries, this script scans for version, repository, and source information and reports what is found. ASTERISK-25376 #close Reported by: Ashley Sanders Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6	10 years ago
Mark Michelson	993ae9a669	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
Scott Griepentrog	53e2a6a829	contrib: script install_prereq should install sqlite3 ... Asterisk needs the sqlite 3 library, which is package sqlite-devel in CentOS. By adding this package to the script, a problem with configure failing is resolved. ASTERISK-25331 #close Reported by: Kevin Harwell Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec	10 years ago
Joshua Colp	309dd2a409	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	2b42264e66	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: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d	10 years ago
Kevin Harwell	93ac45d3bd	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
Rodrigo Ramírez Norambuena	6737ded058	install_prereq: Check if is installed aptitude otherwise to install. ... If in Debian or system based, dont have aptitude installed the script do nothing. This patch checked if aptitude installed, if not installed. Also, if execute script with all packages installed yet, the script not show nothing and return exit 1 because the command 'grep' get nothing from pipe from 'awk'. ASTERISK-25113 #close Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f	10 years ago
Corey Farrell	478fb4a388	MALLOC_DEBUG: Replace WRAP_LIBC_MALLOC with ASTMM_LIBC. ... There are 3 ways that calls directly to standard allocator functions can be dealt with: 1. Block their use, cause them to generate an error. This is the default. 2. Replace them with the Asterisk equivalent function calls. 3. Leave them alone. This change allows one of these 3 options to be selected by any source. The source just needs to define ASTMM_LIBC to ASTMM_BLOCK, ASTMM_REDIRECT, or ASTMM_IGNORE to use option 1, 2 or 3 respectively. Normally ASTMM_BLOCK is the correct option, so it is default when ASTMM_LIBC is not defined. In some cases when building 3rd party code it is desirable to have it use Asterisk functions, without changing the whole source - ASTMM_REDIRECT accomplishes this. When using 3rd party libraries sometimes a static inline function will make use of malloc or free. In these cases it may be unsafe to replace the allocator in the header, as it's possible the memory could be freed by the library using standard allocators. For those cases ASTMM_IGNORE is needed. Change-Id: I8afef4bc7f3b93914263ae27d3a5858b69663fc7	10 years ago
George Joseph	cf637f2510	doc: Make progdocs play nice with git ... Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in Changed /Makefile to copy asterisk-ng-doxygen.in to asterisk-ng-doxygen then modify it with version instead of modifying asterisk-ng-doxygen directly. Updated clean targets as well. Updated /.gitignore and doc/.gitignore. Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622	10 years ago
Ivan Poddubny	b885f719bf	contrib/editors: Fix vim syntax highlighting of comments in config files ... * Added a lookbehind to one-line comment matcher to skip escaped semicolons. * Added support for block comments. Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7	10 years ago
Matt Jordan	75c0aa6979	contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update ... The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755 failed to add ENUM support for Postgres databases. This requires a specific import from the sqlalchemy.dialects.postgresql package. This patch corrects this error, which allows for Postgres update scripts to be generated. ASTERISK-24706 Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015	10 years ago
Joshua Colp	7c18edaa10	Merge "Example script for scan-build (the llvm static analyzer)"	10 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. ... * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	10 years ago
Diederik de Groot	1e74793061	Example script for scan-build (the llvm static analyzer) ... - Added Pre-amble (Options / Flags / Usage Example / GNU License) - Extended Configurability - Made Executable ASTERISK-24917 Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8	10 years ago
George Joseph	c6ed681638	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
George Joseph	51886c68dc	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matthew Jordan	8bae18ab93	res_pjsip: Add an 'auto' option for DTMF Mode ... This patch adds support for automatically detecting the type of DTMF that a PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', the channel created for an endpoint will attempt to determine if RFC 4733 DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type for the channel will be set to inband. Review: https://reviewboard.asterisk.org/r/4438 ASTERISK-24706 #close Reported by: yaron nahum patches: yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) ........ Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	cb1c639817	Add missing file. ASTERISK-24781 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	79a81fed59	alemebic scripts: endpoint identifier order option ... The script was added in 13, but when committed to trunk it caused a branch to occur due to some trunk only alemebic changes. This fixes it so that the new 'add_pjsip_endpoint_identifier_order script points to the correct down revision. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	aef7278af6	res_pjsip: Allow configuration of endpoint identifier query order ... This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ ........ Merged revisions 433028 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	d42c6adb1a	Revert - res_pjsip: Allow configuration of endpoint identifier query order ... Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson ........ Merged revisions 432868 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	1ce529d30e	res_pjsip: allow configuration of endpoint identifier query order ... It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ ........ Merged revisions 432638 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	55709bc1f7	install_prereq: Tweak flags when configuring pjproject. ... This change does two things: 1. Disables debugging so assertions which can return an error do, instead of asserting. 2. Enables IPv6 support. ASTERISK-24632 #close Reported by: Rusty Newton ........ Merged revisions 431843 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Walter Doekes	49cbfa7de6	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	14b8e03dad	contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts ... On Debian based systems, the install_prereq tool uses a search command on Debian that results in selecting both 64-bit and 32-bit packages. Besides the waste of disk space, this can actually cause aptitude use 100% of memory on a VM with 1GB of RAM as it tried to work out all of the 32-bit package dependencies. This patch filters out the 32-bit packages on a 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048 #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876) ........ Merged revisions 430798 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430799 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	fba836cc02	sip_to_pjsip: improve ability to parse input files ... General improvements to SIP to PJSIP conversion utility: 1) track default section of input file to allow parsing an include file that doesn't specify a [section] 2) informatively handle case of assignment without [section] 3) correctly handle getting sections from included files - [section]'s are inherited by included file 4) provide null string as default transport bind ip 5) gracefully handle missing portions of registration string 6) denote steps of operation during conversion and confirm top level files as a convenience ASTERISK-24474 #close Review: https://reviewboard.asterisk.org/r/4280/ Reported by: John Kiniston ........ Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7f8b7ace72	res_pjsip_sdp_rtp: Add support for optimistic SRTP. ... Optimistic SRTP is the ability to enable SRTP but not have it be a fatal requirement. If SRTP can be used it will be, if not it won't be. This gives you a better chance of using it without having your sessions fail when it can't be. Encrypt all the things! Review: https://reviewboard.asterisk.org/r/3992/ ........ Merged revisions 428222 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b2e766a6b7	alembic: Fix alembic migration for 'moh_passthrough' option in res_pjsip. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago