asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	7fd28cefdb	debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c	8 years ago
George Joseph	456bc3c704	debug_utilities: Add ast_logescalator The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	8 years ago
George Joseph	c628a7acac	debug_utilities: Create ast_loggrabber ast_loggrabber gathers log files from customizable search patterns, optionally converts POSIX timestamps to a readable format and tarballs the results. Also a few tweaks were made to ast_coredumper. Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 (cherry picked from commit `5fa1c56d7e`)	8 years ago
George Joseph	47474cfd54	debug_utilities: Create the ast_coredumper utility This utility allows easy manipulation of asterisk coredumps. * Configurable search paths and patterns for existing coredumps * Can generate a consistent coredump from the running instance * Can dump the lock_infos table from a coredump * Dumps backtraces to separate files... - thread apply 1 bt full -> <coredump>.thread1.txt - thread apply all bt -> <coredump>.brief.txt - thread apply all bt full -> <coredump>.full.txt - lock_infos table -> <coredump>.locks.txt * Can tarball corefiles and optionally delete them after processing * Can tarball results files and optionally delete them after processing * Converts ':' in coredump and results file names '-' to facilitate uploading. Jira for instance, won't accept file names with colons in them. Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. [1] For *BSDs, the "devel/gdb" package might have to be installed to get a recent gdb. The utility will check all instances of gdb it finds in $PATH and if one isn't found that can run python, it prints a friendly error. Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd	9 years ago
Joshua Colp	37aaaa2da2	res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. This change implements SRV support for the IP based endpoint identifier module. All possible addresses through SRV are looked up and added as matches. If no SRV records are available a fallback to normal host resolution is done. If an IP address is provided then no SRV lookup occurs. This is configured using the "srv_lookups" option on the identify section and defaults to "yes". ASTERISK-26693 Change-Id: I6b641e275bf96629320efa8b479737062aed82ac	9 years ago
Richard Mudgett	35736d419a	autosupport: Add 'pjproject show buildopts' Change-Id: I8aa55a7c3fb175235ddc7f85e9457d5102d06fa7	9 years ago
Richard Mudgett	580f83dac7	Remove files that got merged in error somehow to the 13 branch. Change-Id: Id79e2226c31084f9252d5aede9050d3cf13322c8	9 years ago
Joshua Colp	8d56016ae4	Merge "tcptls: Use new certificate upon sip reload" into 13	9 years ago
Joshua Colp	e0bc17edff	pjsip: Fix a few media bugs with reinvites and asymmetric payloads. When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc	9 years ago
Joshua Colp	bb982480d8	pjsip: Support dual stack automatically. This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d	9 years ago
Mark Michelson	df75b647da	Update for 13.12.0-rc1	9 years ago
Rodrigo Ramírez Norambuena	29b7a5b00f	Add text of cdr directory into README.md for ast-db-manage Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636	9 years ago
George Joseph	86550f9c17	alembic: Allow cdr, config and voicemail to exist in the same schema cdr, config and voicemail are all separate alembic trees. Because alembic's default is to use a table named 'alembic_version' to store the current tree revision, the 3 trees can't exist in the same schema without stepping on each other. Now each tree uses 'alembic_version_<tree_name>' as the version table. Each tree's env.py script now first checks for 'alembic_version'. If it finds it AND its revision is in the tree's history, the script renames it to 'alembic_version_<tree_name>'. Regardless, the script then continues with the migration using 'alembic_version_<tree_name>' and creates that table if it's not found. The result is that if an existing 'alembic_version' table was found but it didn't belong to this tree, it's left alone and 'alembic_version_<tree_name>' is used or created. WARNING: If multiple trees are using the same schema, they MUST NOT CRU or D any objects with names that might exist in the other trees. An example would be 'yesno_values' type. If two trees perform operations on it, one tree could pull it out from under the other. Thankfully we currently don't share any names among cdr, config and voicemail. NOTE: Since the env.py scripts in each tree were identical, a common env.py has been placed in the ast-db-manage directory and a symlink to it has been placed in each tree directory. ASTERISK-24311 #close Reported-by: Dafi Ni Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898	9 years ago
zuul	139d86c8ab	Merge "sip_to_pjsip.py: Map legacy_useroption_parsing." into 13	9 years ago
zuul	a5f7b07579	Merge "res_pjsip: Add ignore_uri_user_options option." into 13	9 years ago
Richard Mudgett	e9ddab4685	sip_to_pjsip.py: Map legacy_useroption_parsing. Map the sip.conf general section legacy_useroption_parsing to the new pjsip.conf global ignore_uri_user_options. ASTERISK-26316 Reported by: Kevin Harwell Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc	9 years ago
Richard Mudgett	30af92e78d	res_pjsip: Add ignore_uri_user_options option. This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	9 years ago
Walter Doekes	7ed5dc2c58	contrib: Let safe_asterisk script continue without /dev/tty9. If you use the safe_asterisk script, it uses hardcoded defaults before running configurable values from /etc/asterisk/startup.d. The hardcoded default has TTY=9. Some containerized environments don't have such a TTY, and safe_asterisk would stop. The custom configuration from /etc/asterisk/startup.d/* isn't read until after it stopped, so changing TTY in a custom config did not help. This changeset changes safe_asterisk to continue if the TTY setting was untouched and /dev/tty9 and /dev/vc/9 aren't found. Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc	9 years ago
zuul	52335c3fe7	Merge "sip_to_pjsip.py: Map canreinvite as directmedia alias." into 13	9 years ago
zuul	e3f549c2f6	Merge "sip_to_pjsip.py: Fix typo converting outboundproxy registration." into 13	9 years ago
Richard Mudgett	3314e1cec2	sip_to_pjsip.py: Map canreinvite as directmedia alias. Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2	9 years ago
Richard Mudgett	6372f40ba0	sip_to_pjsip.py: Fix typo converting outboundproxy registration. Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15	9 years ago
Richard Mudgett	11eb1afd2d	sip_to_pjsip.py: Fix comment typo and tabs. Change-Id: If35174614545727817d329c60ba4456c028941b5	9 years ago
Alexander Traud	b59d3b48d0	sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. When using the migration script sip_to_pjsip.py, and your sip.conf is configured with bindaddr=::, two transports are written to pjsip.conf, one for 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface like in chan_sip. Furthermore, the script internal functions "build_host" and "split_hostport" did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change makes sure, even such addresses are parsed correctly. ASTERISK-26309 Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48	9 years ago
zuul	8778e22b15	Merge "sip_to_pjsip: Map externhost/ip to Transports." into 13	9 years ago
zuul	f46f88157e	Merge "sip_to_pjsip: Add cert_file." into 13	9 years ago
zuul	504da21d47	Merge "res_pjsip: Add contact_user to endpoint" into 13	9 years ago
Alexander Traud	a628009eb9	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
Joshua Colp	5d4ce33e8c	Merge "sip_to_pjsip: Write cos and tos." into 13	9 years ago
Kevin Harwell	cab6975b02	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	43254c2db7	Merge "sip_to_pjsip: Parse register even with transport." into 13	9 years ago
Joshua Colp	e3d72c7aee	Merge "sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit." into 13	9 years ago
Joshua Colp	e1ef9e697e	Merge "sip_to_pjsip: Map (session-)timers correctly." into 13	9 years ago
Joshua Colp	f5165fd0aa	Merge "sip_to_pjsip: Add cert_file and ca_list_path." into 13	9 years ago
Joshua Colp	9426627452	Merge "sip_to_pjsip: Write username even without authname." into 13	9 years ago
Joshua Colp	02f5d30739	Merge "sip_to_pjsip: Map the TLS method correctly." into 13	9 years ago
Joshua Colp	1ae280ed6c	Merge "sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent." into 13	9 years ago
Joshua Colp	02f2937f48	Merge "sip_to_pjsip: Write media_encryption." into 13	9 years ago
Alexander Traud	2381ddde63	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	6500f5e138	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	21e9c69e56	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	c9a97398f7	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	60275359bc	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	0d479232eb	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	cbc1b2d020	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	5f33e99534	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	231ea0350d	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	23eb065121	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	0b675a208b	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	329507fe20	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

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568 Commits (67b21dc63afb94e4a8b913f149cfc220e16790f7)