asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	06ff84b549	To be able to achieve the things that we would like to achieve with the Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	79ff4ebbdf	Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	ff6aacc1e8	A small clarification to keep bugs from being filed, and confusion from rising, if clearglobalvars is set, and globals are set in the AEL file. (9419) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	fe446989eb	Fix unescaped semicolon (reported via -dev list) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	31a7b4aceb	fix a couple SLA documentation references git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	78d178173f	By default, don't attempt to do any CallerID handling at all with SLA because it is known to not work properly in some situations. However, add an option to enable it for those that would like to use it anyway. The short story behind this is that to properly handle CallerID with SLA, we need the ability to change the CallerID on an existing call, and we are not ready to handle that. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	d93c20ac9d	fix the reference to the SLA documentation git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	fa866efb5c	Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	dd920562ee	Clarify the documentation of the dialout and sendvoicemail options. (issue #9000, caio1982 and serge-v) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3b6dc39807	add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	31cf37519f	Merge changes from svn/asterisk/team/russell/sla_updates * Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	65915e679a	minor tweaks to the sla docs git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	447561d7a2	Merge more changes from svn/asterisk/team/russell/sla_updates * Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	9d3ff33b25	Merge changes from svn/asterisk/team/russell/sla_updates * Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	9021d3c3b2	Merge current set of changes from svn/asterisk/team/russell/sla_updates * Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks\|stations)" CLI commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	f314685447	Merge changes from team/russell/sla_updates. This batch of changes to the SLA code does a few different things. * I made the SLA code event driven instead of having to act in a lot of busy loops while dialing things to wait for state changes. This makes the code more efficient and readable at the same time. * I have implemented a couple of new features. The first is inbound trunk ringing timeouts. This is an option that defines how long to let an incoming call on a trunk to ring. * I have also implemented ring timeouts for stations. They may be specified for the entire station, meaning it is how long to let the station ring before giving up. You can also specify a ring timeout for a specific trunk on a station. So, you can say that you only want a specific station to ring 5 seconds if it is line1 ringing, but otherwise, there is no timeout. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	6bdc40358a	Change the formatting of sla.conf.sample to make it more readable. (issue #9112, blitzrage) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	960b4de2de	Merged revisions 55005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55005 \| russell \| 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) \| 9 lines Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a conference is created from meetme.conf, it is acceptable behavior that the pin can not be changed until the conference goes away. I also added a note in meetme.conf to describe this behavior. We still have another issue in 1.4 and trunk where some conferences with no users don't go away. That is the real bug that needs to be addressed here. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	2123a1bf02	Fix a typo where "vmpassword" should be "vmsecret" git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	7ee02f585d	Merge team/russell/sla_rewrite This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	90a4b844a9	Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	d7cde47f06	Add explanation of port= in combination with defaultip= (thanks jsmith) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	96beb30159	By suggestion from kpfleming last week, change "vmpassword" to "vmsecret". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	c9a898c665	Fix Italian numeral support in say.conf for "_[2-9]00" case. "2131" would've translated to something along the lines of (pardon my..Italian {or lack thereof}) "duecentocentotrentuno", which makes no sense at all. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	2e9e873c09	Fix German language support in say.conf Properly support 21, 31, 41, 51, 61, 71, 81, and 91. einundzwanzig has the same format as zweiundzwanzig (as do all other "_ZX" spoken numerals) Fix support for numbers in the 10,000,000 to 99,999,999 range. Add support for numbers in the 100,000,000 to 999,999,999 range. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Matt O'Gorman	cc003179d4	Patch allows for changing voicemail password in users.conf from voicemail main, written by AnthonyL bug #8436 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@51030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	6de5768987	Update documentation to state that you shouldn't use realtime static with voicemail.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@50647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	fb52698667	Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48319 \| crichter \| 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) \| 1 line changed a few debugs to higher debug levels ........ r48321 \| crichter \| 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) \| 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 \| crichter \| 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) \| 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 \| crichter \| 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) \| 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 \| crichter \| 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) \| 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 \| crichter \| 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) \| 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 \| crichter \| 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) \| 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	ab6ee2376a	Adding note on effect of applicationmap features on re-invites git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@49145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	d2b7e8b247	Be a bit more politically correct git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48985 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	bfe4bb0f1e	Issue #8575 - Buggy cisco MWI support. Normally we try not to change our software for bugs in other devices. But in this case, the Cisco phones are so widespread so we try to implement a fix while waiting for a bugfix from Cisco. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	4ee818eb8f	Merged revisions 48322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48322 \| russell \| 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) \| 3 lines Fix the name of the rtignoreregexpire option in the sample configuration file. (issue #8526, arkadia) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	7945d4ca35	Add missing s from another repository. (thanks jcmoore!) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	027096b3a3	Updating sip.conf.sample with information about T38 not working when chan_local or chan_agent is involved in the call. I don't know how big a fix that would be to solve, but this is the current state of affairs. (Chan_sip currently checks if the other side of the bridge has a SIP tech. We could/should implement another check, possibly for udptl_write or some flag in the ast_channel structure). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	56c03478ab	Add documentation to voicemail.conf.sample for ODBC storage. Issue 8499 - patch by blitzrage. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	f89143bd13	- Disable RTP hold timers while T.38 fax transmission happens - Encapsulate RTP timers in the rtp structure so we have one for video and one for audio The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send something that video phones support in the RTP stream. I now this is a big architectual change at this stage for 1.4, but decided it was needed to avoid future bug reports. - Document the RTP NAT keepalive option in sip.conf.sample Issue 7679 in the bug tracker. Please test. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	8cbe6025b6	Merged revisions 48183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48183 \| qwell \| 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) \| 2 lines Fix a small typo - issue 8848, reported by pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48186 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	98d3fb64ed	- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states - Remove support for T.38 early media, since it's impossible. (Two patches in one - extra friday evening offer due to being off line from svn today... :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	802c3c3ecf	Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 \| file \| 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) \| 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	a68edf400f	Explain the use device status system implemented in SIP for subscriptions, queues and manager a bit better. Like in 1.2, you will get more detailed information if you set a call limit for a device. When the call limit is reached, the status system will report a device as busy. For queues, setting a call limit per SIP device is propably a requirement. In most cases, it will work much better if you only use type=peer and not type=friend. We might decide to backport the new setting from trunk to apply all call limits to the peer part of a friend only. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	3fe8e34039	Clarify RTP timers. Sorry, grandma. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	7da1a54fe6	Explain properly how videosupport works. Committ from Asterisk Video Task Force meeting in Paris! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	e1e6a1b2a8	Make the HOLD notification optional, in order to avoid a lot of extra database lookups for all those realtime users out there. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	5bd53e3588	- CANCEL is never authenticated (according to the RFC) - Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	4fd3b973bf	clean up sample config, and make native file playback the more obvious default choice git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	9ab1cc22a4	Support ;rport when we're supposed to support ;rport. Issue #7473 . git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Christian Richter	6964f148ba	Merged revisions 46176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46176 \| crichter \| 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) \| 1 line added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@46351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	8273d95be3	update entry to reboot a snom phone (issue #7850 , pnlarsson) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	590698e583	Adding information about Marks direct-RTP hack to the docs... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	45fc0eaba4	Now, remove all traces of the option that we did not need :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago

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780 Commits (50d606bbf8f64dea9c65acdd993f2c406c8ca801)