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2006-04-01Flesh out the remainder of the manager + http changes and create a sample ↵markster2-5/+17
application to partially demonstrate the capability of manager over http. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16850 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-31 Provide warning about current behavior of autofill = yes bweschke1-1/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16673 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-30Typooej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16305 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-29whitespace "fixes", and general cleanupnorth1-13/+11
It's nice to have consistency in sample configs too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16268 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-29the comment character is ';' not '#' ...rizzo1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16235 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-29Added more "valid" phone types to skinny sample config.north1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15938 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-28update example filerizzo1-42/+50
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15543 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-28disable the http server by default at the request of people on IRCrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15518 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-28support subscription-based MWI, and use proper Call-ID on NOTIFY messages ↵kpfleming1-0/+2
(issue #6390) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15476 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-28improve IP TOS support for SIP and IAX2 (issue #6355, code from jcollie plus ↵kpfleming3-15/+10
modifications) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15435 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-27Issue #5427oej1-1/+7
- Enable videosupport per device - Implement maxcallbitrate setting for video calls Patch by John Martin, thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15148 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-27Issue #6705 (oej)oej1-1/+5
- Implement option for allow/disallow subscriptions - Implement option for allow/disallow overlap dialling - Set default to disable overlap dialling in sip.conf.sample for new installations - Remove overlap dialling from subscription logic git-svn-id: http://svn.digium.com/svn/asterisk/trunk@15107 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-25Add micro-http server and abstract manager interface, make snmp not die markster1-0/+23
on reload. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14953 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-23Allow channels to be moved if channel change is requested in SETUP_ACK, also ↵mattf1-1/+1
add a WAY cool new field to the nsf option git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14521 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-21Added separate outsignalling specification, and fixed FEATDMF to allow forjdixon1-0/+9
international inbound calls. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14079 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-21add a CLI command that allows converting files to other formats usingrussell1-10/+10
the Asterisk file format and codec translator modules (issue #6062) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14027 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-21Merged revisions 13964 via svnmerge from russell1-1/+6
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r13964 | russell | 2006-03-21 13:59:29 -0500 (Tue, 21 Mar 2006) | 3 lines add a note explaining how to set the DYNAMIC_FEATURES variable to allow the use of custom features (issue #6747) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13967 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-21add indications for Malaysia (issue #6758)russell1-0/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13738 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-20removed dynamic switching from transparent to hdlc mode. Instead we've got a ↵crichter1-0/+6
config option hdlc=yes now which enables the hdlc controller for a data call git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13637 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-20these traceing option do not exist anymorecrichter1-9/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13633 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-19Fix reference to README filesoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13549 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-17Add reference to examples for files and custom, too make it more obiousoej1-3/+4
that you're required to read on... (hello xrobau) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13323 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-16Clarify documentation for "progressinband" - imported from 1.2oej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@13246 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-14deprecate the mailboxdetail option and always use its behavior, instead ↵russell1-7/+0
(issue #6665) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12923 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-14add an option to cdr.conf that enables ending CDRs before executingrussell1-0/+7
the "h" extension as opposed to afterwards (issue #6193) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12896 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-09added option to change the connected party number dialplan (ton)crichter1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12481 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-08allows the table field to be configurable formogorman1-0/+2
cdr_tds.conf. patch provided by bug 6629 with minor change. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12475 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-07added a bit more detailed description for the echotraining parameter, also ↵crichter1-5/+9
changed the default from 1 to 2000. The default for the upper_threshold is now 0 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@12287 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-06Merged revisions 11946 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r11946 | russell | 2006-03-05 21:32:35 -0500 (Sun, 05 Mar 2006) | 2 lines fix a typo in the description of the ringtimeout option ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11947 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-02 cdr_csv logging parameters in cdr.confmogorman1-0/+5
usegmtime, log date/time in GMT loguniqueid log uniqueid loguserfield log user field patch provided by bug 5015 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11586 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-28Whitespace changesoej1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11455 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-28better default values for jitterbuffer in code and configcrichter1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11334 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-26Add SNMP support (bug #6439)markster1-0/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11193 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-26Make IAX2 multithreadedmarkster1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11192 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-22Merged revisions 10511,10535,10736 via svnmerge from kpfleming1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r10736 | kpfleming | 2006-02-22 11:32:15 -0600 (Wed, 22 Feb 2006) | 2 lines add comment warning people about trying to use hostnames/IPs in the sample config ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10737 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-20Changing syntax once again slightly and standardizingmogorman1-22/+16
config to other asterisk samples , bug note 6530 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10559 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15adde incoming_early_audio option, to avoid sending tone indications to the ↵crichter1-0/+11
remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10227 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15added pmp_l1_check option, to avoid l1 checking for group calls on PMP portscrichter1-0/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10225 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15add option to avoid calling members whose channels are 'in use' (issue ↵kpfleming1-0/+6
#6315, plus documentation) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10163 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14changed naming scheme for variables so it matchesmogorman1-1/+1
asterisk standard, changed it so it can take frames of sizes other than 20ms, allowed for the app to reload properly, and finalyl changed sample to general section as to follow standards. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10086 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14set properties for new files (i need to get this documented)kpfleming1-24/+24
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9960 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-14Commiting 5959 with minor formatting and typomogorman1-0/+24
fixes. Thanks to all those involved. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9918 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-13Bug 6477 - minor syntax error, plus a few other syntax fixestilghman1-4/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9783 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-10Add smdi support for asterisk (see doc/smdi.txt for config info) (#5945)mattf3-5/+63
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9423 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-10restore 'rfc2833' naming for DTMF mode in chan_sipkpfleming1-7/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9391 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-09- Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.oej1-6/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9294 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-09- Adding example on using european time zones in voicemail.conf oej1-5/+7
- Moving the timezone section to below the documentation section Example written and contributed by Henrik Olsen, Astricon Training student. Thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9292 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-07default values of jitterbuffer and jitterbuffer_upper_threshold should be > ↵crichter1-1/+1
0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9186 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-02* removed unnecessary struct elements and functionscrichter1-36/+30
* fixed "RETRIEVE does not work" bug * fixed some NT Mode bugs * removed some // comments * added configureable jitterbuffer * removed own tone-generator, and use asterisks instead, to support asterisks indications * added more support for hw-bridging, we bridge now every possible call * fixed some hdlc mode issues, with a patch for chan_zap we can make data calls between chan_zap and chan_misdn now * completely reworked the config engine, works like a charm now * fixed SetCallerPres - bug * added Progress and Proceeding passing * optimized Ringing Indication handling * added full ast_send_text support (you can setup nice menus with the dialplan now) * added support to read /etc/misdn-init.conf to clarify the NT+PTP Problem * we compile now channels/misdn if mISDNuser is installed systemwide git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9114 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-01- Adding a doc/00README.1st with an INDEX over README filesoej1-0/+26
- Moving files from / to /doc or /configs - Renaming some documentation files Thank you for the initiative, manxpower! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9046 f38db490-d61c-443f-a65b-d21fe96a405b
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