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2007-10-01Remove chan_usbradio config file from tree, it is not present in here.file1-54/+0
(closes issue #10839) Reported by: casper git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@84163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-18Correct the allowexternaldomains option in SIP sample config.qwell1-1/+1
Issue 10753 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82751 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-14Add a note to help clarify the value set with the echocancel option.russell1-0/+3
(inspired by Malcolm's blog post on blogs.digium.com about HPEC) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82435 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-14Added channel driver for USB Radio device andjdixon1-0/+54
support thereof. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82366 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-10Removing non-existent options from misdn configuration sample.mmichelson1-10/+0
(closes issue #10678, reported and patched by IgorG) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@82091 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-07Moving the explanation for joinempty to a more appropriate placemmichelson1-10/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@81886 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-04Change default followme config file to point to the correct files.qwell1-3/+3
Issue 10644, patch by pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@81453 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-30Fix a typo, update a reload command, and remove an unused configuration file.russell3-95/+3
(closes issue #10606, casper) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@81379 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-28Add Russian tones. (closes issue #7953, hanabana)russell1-7/+12
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@81226 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-21(issue #10510)qwell1-17/+15
Reported by: casper Patches: cdr.conf.diff uploaded by casper (license 55) Fix a few errors in sample cdr config file. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80130 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20(issue #10499)qwell1-8/+8
Reported by: casper Patches: extensions.conf.sample.diff uploaded by casper (license 55) Update CLI examples in extensions.conf.sample to reflect command changes. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80047 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-10(closes issue #10422)file1-0/+4
Reported by: bhowell Add note to sample configuration about module load order and how it can cause perfectly good queue members to be marked as invalid. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@78951 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08(closes issue #10335)file1-1/+2
Reported by: adamgundy Update sip.conf to include another scenario where directrtpsetup will fail. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@78569 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-02Make sure we actually allow 6 chars to be sent.qwell1-1/+3
Also make note of the "A" option of date format. Issue 9779, modifications by DEA, wedhorn, and myself. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77996 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Add a sample configuration file and example tables for use with ↵russell2-1/+20
res_config_pgsql. (issue #9676, suretec) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@63329 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04explanation for httptimeout in manager.confpari1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@63047 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the ↵murf1-4/+37
channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@62689 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-01Merged revisions 62496 via svnmerge from russell1-0/+21
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62496 | russell | 2007-05-01 11:26:23 -0500 (Tue, 01 May 2007) | 3 lines Add indications.conf information for the Philippines. (issue #9525, reported and patched by loloski) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@62497 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Remove unused (and potentially confusing) jitterbuffer options from sample ↵qwell1-5/+1
config. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@62371 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06To be able to achieve the things that we would like to achieve with therussell1-0/+12
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: http://svn.digium.com/svn/asterisk/branches/1.4@60603 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-05Added some clarification to the example configs for CDRs, on how to select a ↵murf2-6/+79
backend. Also, made cdr-csv the default if you 'make samples', and no other changes. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@60323 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-30A small clarification to keep bugs from being filed, and confusion from ↵murf1-0/+10
rising, if clearglobalvars is set, and globals are set in the AEL file. (9419) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59452 f38db490-d61c-443f-a65b-d21fe96a405b
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
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