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2007-03-19Fix unescaped semicolon (reported via -dev list)tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59040 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-16fix a couple SLA documentation referencesrussell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58957 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-14By default, don't attempt to do any CallerID handling at all with SLA becauserussell1-1/+5
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: http://svn.digium.com/svn/asterisk/branches/1.4@58894 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-13fix the reference to the SLA documentationrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58870 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-12Add matchexterniplocally setting which only substitutes your ↵file1-0/+4
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: http://svn.digium.com/svn/asterisk/branches/1.4@58779 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Clarify the documentation of the dialout and sendvoicemail options.russell1-5/+6
(issue #9000, caio1982 and serge-v) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58119 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03add missing configuration template. Thanks to Lacy Moore on asterisk-users ↵russell1-0/+10
for pointing this out\! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57591 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merge changes from svn/asterisk/team/russell/sla_updatesrussell1-2/+8
* 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: http://svn.digium.com/svn/asterisk/branches/1.4@57364 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28minor tweaks to the sla docsrussell1-5/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge more changes from svn/asterisk/team/russell/sla_updatesrussell1-40/+61
* 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: http://svn.digium.com/svn/asterisk/branches/1.4@57203 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge changes from svn/asterisk/team/russell/sla_updatesrussell1-0/+4
* 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: http://svn.digium.com/svn/asterisk/branches/1.4@57144 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merge current set of changes from svn/asterisk/team/russell/sla_updatesrussell1-2/+10
* 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: http://svn.digium.com/svn/asterisk/branches/1.4@57089 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22Merge changes from team/russell/sla_updates.russell1-5/+12
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: http://svn.digium.com/svn/asterisk/branches/1.4@56277 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-20Change the formatting of sla.conf.sample to make it more readable. russell1-63/+59
(issue #9112, blitzrage) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@55553 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Merged revisions 55005 via svnmerge from russell1-0/+5
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: http://svn.digium.com/svn/asterisk/branches/1.4@55006 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12Fix a typo where "vmpassword" should be "vmsecret"russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@54002 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-10Merge team/russell/sla_rewriterussell1-21/+77
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: http://svn.digium.com/svn/asterisk/branches/1.4@53810 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-02Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, ↵oej1-0/+6
but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53109 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Add explanation of port= in combination with defaultip= (thanks jsmith)oej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53062 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-25By suggestion from kpfleming last week, change "vmpassword" to "vmsecret".russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52160 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-20Fix Italian numeral support in say.conf for "_[2-9]00" case.qwell1-1/+1
"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: http://svn.digium.com/svn/asterisk/branches/1.4@51350 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-20Fix German language support in say.confqwell1-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.4@51348 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-16Patch allows for changing voicemail password in users.conf from voicemail ↵mogorman1-0/+5
main, written by AnthonyL bug #8436 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51030 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-12Update documentation to state that you shouldn't use realtime static with ↵qwell1-1/+2
voicemail.conf git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@50647 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-03Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from crichter1-2/+2
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: http://svn.digium.com/svn/asterisk/branches/1.4@49313 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-02Adding note on effect of applicationmap features on re-invitesoej1-1/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49145 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Be a bit more politically correctoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48985 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27Issue #8575 - Buggy cisco MWI support.oej1-0/+3
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: http://svn.digium.com/svn/asterisk/branches/1.4@48982 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-06Merged revisions 48322 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@48323 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Add missing s from another repository. (thanks jcmoore!)oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48268 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-05Updating sip.conf.sample with information about T38 not workingoej1-0/+2
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: http://svn.digium.com/svn/asterisk/branches/1.4@48264 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-04Add documentation to voicemail.conf.sample for ODBC storage.qwell1-0/+6
Issue 8499 - patch by blitzrage. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48230 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-02- Disable RTP hold timers while T.38 fax transmission happensoej1-8/+18
- 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: http://svn.digium.com/svn/asterisk/branches/1.4@48199 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-01Merged revisions 48183 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/branches/1.4@48186 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-01- Backport of the "limitonpeers" patch from trunk, to fix a lot of issues ↵oej1-0/+6
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: http://svn.digium.com/svn/asterisk/branches/1.4@48177 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-30Merged revisions 48142 via svnmerge from file1-2/+4
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: http://svn.digium.com/svn/asterisk/branches/1.4@48143 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Explain the use device status system implemented in SIP for subscriptions,oej1-0/+9
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: http://svn.digium.com/svn/asterisk/branches/1.4@48113 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-29Clarify RTP timers. Sorry, grandma.oej1-3/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@48105 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-20Explain properly how videosupport works. oej1-1/+5
Committ from Asterisk Video Task Force meeting in Paris! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47845 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16Make the HOLD notification optional, in order to avoid a lot of extra ↵oej1-1/+4
database lookups for all those realtime users out there. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47755 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16- CANCEL is never authenticated (according to the RFC)oej1-0/+6
- Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47733 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07clean up sample config, and make native file playback the more obvious ↵kpfleming1-27/+28
default choice git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@47279 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-31Support ;rport when we're supposed to support ;rport. Issue #7473.oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@46628 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-27Merged revisions 46176 via svnmerge from crichter1-0/+7
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: http://svn.digium.com/svn/asterisk/branches/1.4@46351 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-18update entry to reboot a snom phone (issue #7850, pnlarsson)russell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45439 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Adding information about Marks direct-RTP hack to the docs...oej1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45329 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-17Now, remove all traces of the option that we did not need :-)oej1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45314 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45265 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45265 | file | 2006-10-16 15:59:54 -0400 (Mon, 16 Oct 2006) | 2 lines Use responses rather then replies even though they mean the same thing. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45280 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-16Merged revisions 45260 via svnmerge from file1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45260 | file | 2006-10-16 15:28:18 -0400 (Mon, 16 Oct 2006) | 2 lines Add 'ignoreoodreplies' option which will not create a pvt structure on a SIP response but instead basically drop it. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45262 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-06Merged revisions 44334 via svnmerge from crichter1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r44334 | crichter | 2006-10-04 17:13:58 +0200 (Mi, 04 Okt 2006) | 1 line added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@44561 f38db490-d61c-443f-a65b-d21fe96a405b
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