telephony/net/asterisk/files, branch masterMirror of telephony feedhttps://git.openwrt.org/feed/telephony/atom?h=master2026-02-23T19:12:42Zasterisk: add an 'interface' option to trigger reregister events2026-02-23T19:12:42ZAndre Heider2021-11-04T08:16:09Zurn:sha1:04c1b2a250959821fe3e5ba0371a4bc534314284
If set this now automatically re-registers all outbound registrations when
the interface is becoming available. This ensures that no stale IPs are
registered at the SIP trunk.
Fixes #681
Signed-off-by: Andre Heider <a.heider@gmail.com>
asterisk: add a reregister extra command2026-02-23T19:12:42ZAndre Heider2021-11-04T07:57:09Zurn:sha1:1b924c1b50bb5c2a99353a71a26bee79022bb0e5
`/etc/init.d/asterisk reregister` will re-register all outbound registrations.
Currently only pjsip is supported, but, if required, this can be easily
extended in the future.
Signed-off-by: Andre Heider <a.heider@gmail.com>
asterisk: clean up init.d script2026-02-23T19:12:42ZAndre Heider2021-11-04T08:35:18Zurn:sha1:cc89b4b817defd9aa9254ebdd1dc290686c90dfe
Remove redundant empty lines and refactor the logging. It
can now be used for any level.
Signed-off-by: Andre Heider <a.heider@gmail.com>
asterisk: use return in init script2022-11-08T20:08:54ZSebastian Kemper2022-11-08T20:08:16Zurn:sha1:69aebd4f4f4f8cf244276c4967813061412aed21
"start_service()" is a function, hence "return" should be used instead
of "exit".
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
asterisk: don't send stdout to syslog by default2022-02-06T19:16:44ZPhilip Prindeville2022-02-06T18:42:04Zurn:sha1:fb0c2cf2b8d5775e46cd8c2e71f9a20414feeb4a
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
asterisk: init: use daemon facility, not user2021-12-19T11:46:54ZSebastian Kemper2021-12-19T11:46:52Zurn:sha1:992c894c1c3fda14e285ebfbad0bae2ba3c709aa
This facility was suggested in pull request #701.
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
asterisk: send SIGHUP on reload2021-12-19T11:43:35ZAndre Heider2021-11-04T07:15:50Zurn:sha1:c84e2f74140afffd1afdc9d6a0e33dd852b9e82d
asterisk reloads its config upon SIGHUP, use it.
Signed-off-by: Andre Heider <a.heider@gmail.com>
asterisk: don't start with explicit group2021-10-22T22:12:15ZSebastian Kemper2021-10-22T22:12:13Zurn:sha1:9017e04b8755b298649280b3d4491666f1b0c94b
Currently the asterisk init script starts the service with the group
"asterisk". Like this, even if the asterisk user is part of other groups,
asterisk will not be able to make use of them. So for instance if you add the
user to the group "dialout", asterisk will run under group "asterisk", instead
of "asterisk" _and_ "dialout".
Not specifying the group gets rid of this limitation.
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
asterisk: upgrade to Asterisk 18 LTS2020-11-01T08:37:46ZSebastian Kemper2020-11-01T08:37:44Zurn:sha1:b936fade673fca57d1ab51fbfa9323c9fc9880bb
- Bump to new LTS release.
- Move to folder asterisk and remove AST_MAJOR_VERSION variable, as we
only have one version anyway.
- Add new modules.
- Rename voicemail to app-voicemail.
- Remove deps of voicemail on res-adsi and res-smdi as they are
optional.
- Use INSTALL_DATA for headers.
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>