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Author SHA1 Message Date
tilghman 9f974d96fa Enhance ExternalIVR with new options and commands.
(closes issue #12705)
 Reported by: ctooley
 Patches: 
       new_externalivr_argument_format-v2.diff uploaded by ctooley (license 136)
       new_externalivr_documentation.diff uploaded by ctooley (license 136)
       and a few additional fixes by me


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117725 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22 05:10:01 +00:00
mmichelson cc2d2ea926 Add a new manager event, AgentRingNoAnswer to
app_queue.

(closes issue #12591)
Reported by: CCHAsteria
Patches:
      app_queue_RNA_event.diff uploaded by CCHAsteria (license 477)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117625 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-21 20:27:45 +00:00
oej 0feeda5c8f Add support for playing an audio file for caller and callee at start and stop of monitoring (one-touch monitor).
Keep messages short, since the other party is waiting while one party hear the message...



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115784 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-12 18:39:09 +00:00
russell 995531248a Merge changes from team/russell/smdi-msg-searching
This commit adds some new features to the SMDI_MSG_RETRIEVE() dialplan function.
Previously, this function only allowed searching by the forwarding station.
I have added some options to allow you to also search for messages in the queue
by the message desk terminal ID, as well as the message desk number.

This originally came up as a suggestion on the asterisk-dev mailing list.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115021 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01 19:05:36 +00:00
murf 137c1d8d9e (closes issue #12467)
Reported by: atis
Tested by: murf

This upgrade adds the ~~ (concatenation) string operator to expr2.
While not needed in normal runtime pbx operation, it is needed when
raw exprs are being syntax checked. This plays into future syntax-
unification plans. By permission of atis, this addition in trunk 
and the reason of why things are as they are will suffice to close
this bug.

I also added a short note about the previous addition of "sip show sched"
to the CLI in CHANGES, which I discovered I forgot in a previous commit.




git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114423 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21 21:13:02 +00:00
russell e2341e6bee Add a simple janitor project
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114325 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21 15:01:04 +00:00
tilghman 084bd8a463 fileio.h does not exist; io.h does, though.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114202 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-17 15:12:52 +00:00
murf 993e45a63b This is the scariest commit I've done in a long time. This is the astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114190 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-16 23:53:27 +00:00
russell ac86bfbc55 Make some notes about common usage of pbx_builtin_getvar_helper() that is not
thread-safe.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111909 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28 22:50:46 +00:00
tilghman 5dc9edd7f5 Merged revisions 111605 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines

Update debugging text, since Valgrind eliminated the --log-file-exactly option.
(Closes issue #12320)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111606 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-28 14:37:28 +00:00
juggie fdce67d61a update documentation to reflect the changes in the way configure detects net-snmp.
(closes issue #12067)
Reported by: juggie
Patches:
      12067_snmp_doc.patch uploaded by juggie (license 24)
Tested by: juggie


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110911 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26 17:24:54 +00:00
jpeeler d7f3722fa5 documenting changes as a result of adding TCP functionality to ExternalIVR
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108639 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-13 23:12:59 +00:00
qwell e1e65e86e8 Merged revisions 107826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines

Update documentation for pgsql ODBC voicemail.

(closes issue #12186)
Reported by: jsmith
Patches:
      vm_pgsql_doc_update.patch uploaded by jsmith (license 15)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107827 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11 23:38:00 +00:00
russell 20154045f6 fix example usage
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106684 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 16:31:48 +00:00
russell b1028d3f65 minor text changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106518 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 01:19:02 +00:00
russell fb7c8e41c3 Add updated SMDI documentation that I had only sitting in my email ... oops
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106507 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07 01:15:36 +00:00
mmichelson f523ddafbb Adding the Atxfer manager command. With this, you may initiate
an attended transfer over AMI

(closes issue #10585)
Reported by: ornati
Patches:
      atxfer-trunk-r90428.diff uploaded by ornati (license 210)
	  (with modifications from me)
Tested by: putnopvut



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106236 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 22:33:05 +00:00
mvanbaak c0a0a7bccc document var_metric usage to prevent bugreports that are actually configuration issues
(closes issue #12151)
Reported by: caio1982
Patches:
      DB_metric3.diff uploaded by caio1982 (license 22)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106186 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05 21:19:06 +00:00
russell 82c0c67eac note that the chan_sip conversion is already in progress
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104473 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 04:42:59 +00:00
russell 755d0faedb add another janitor project
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104444 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 04:33:30 +00:00
russell 0b84675fcf Add the stuff from the janitor projects page that is still relevant. I figure
that if we keep this in the tree, it will be much easier to keep up to date.
The page on asterisk.org just links to this on svn.digium.com/view


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104419 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 04:14:54 +00:00
qwell 3f03885586 Create placeholder file...for now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104418 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-27 03:52:18 +00:00
tilghman 14f9996743 1) Make braces mandatory for if/for/while, even around single statements.
2) Revise the argument parsing section, showing use of the standard macros.
3) Fix a typo.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104176 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-26 18:40:26 +00:00
bbryant 85bdf7bf13 Adding more tls configuration details to sip.conf sample, with a list of valid ciphers provided in both files. .. First commit since July, woot
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25 19:00:16 +00:00
tilghman 92539559f8 Move Originate to a separate privilege and require the additional System privilege to call out to a subshell.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104039 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-22 22:55:35 +00:00
mmichelson e92a79cd30 Merged revisions 103722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb 2008) | 8 lines

Final round of changes for configure script logic for IMAP

Now if a directory is specified, then we will search that directory for
a source installation of the IMAP toolkit. If none is found, then we will
use that directory as the basis for detecting a package installation of
the IMAP c-client. If that check fails, then configure will fail.


........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103725 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-15 17:32:43 +00:00
mmichelson 0142926ab0 Same changes as made to 1.4 in revision 103710
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103711 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-15 00:59:21 +00:00
mmichelson 4b5327af0c Trunk version of 1.4's imap documentation updates
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103705 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-14 23:51:49 +00:00
qwell 34fa07e0ee Reintroduce more chan_vpb stuff that was removed in r100421 and r100422
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100679 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-28 21:11:24 +00:00
qwell 76bbd4b67a Get rid of that last little bit.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100422 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-25 22:54:01 +00:00
tilghman 533d426fef Add res_config_ldap for realtime LDAP engine.
(closes issue #5768)
 Reported by: mguesdon
 Patches: 
       res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121)
       res_ldap.conf.sample uploaded by suretec (license 70)
       asterisk-v3.1.4.ldif uploaded by suretec (license 70)
       asterisk-v3.1.4.schema uploaded by suretec (license 70)
 Tested by: oej, mguesdon, suretec, cthorner


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99696 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22 22:33:20 +00:00
oej 555cacbd74 Documentation updates for BRIDGEPVTCALLID
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22 20:44:56 +00:00
oej e71421ff23 Small fixes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99482 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22 09:46:28 +00:00
russell d6e19bdc91 Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip.  There are various
new options in configs/sip.conf.sample that are used to enable these features.  Also,
there is a document, doc/siptls.txt that describes some things in more detail.

This code was implemented by Brett Bryant and James Golovich.  It was reviewed
by Joshua Colp and myself.  A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names.  If you were one of them, thanks a lot for the help!

(closes issue #4903, but with completely different code that what exists there.)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-18 22:04:33 +00:00
twilson 7918f534be Update res_phoneprov to default to setting the SERVER variable to the IP
the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf.  I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.

Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport.  Tested on Linux and OS X.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98988 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17 03:09:32 +00:00
file b43ac58b68 Update documentation.
(closes issue #11763)
Reported by: IgorG
Patches:
      docupd.v1.diff uploaded by IgorG (license 20)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98695 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14 14:33:17 +00:00
russell 397f602159 Add some extra checking to help out with a potential error when trying to
run "make asterisk.pdf" when not all of the right packages are installed.

(closes issue #10763)
Reported by: Corydon76
Patches:
      20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98454 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 23:57:01 +00:00
tilghman 56dc662118 Documentation updates
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98152 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 15:12:33 +00:00
twilson 11f6af8c7b Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution.  It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09 21:37:26 +00:00
mmichelson f6d4739457 Adding user-configurable TCP timeout settings to IMAP voicemail. This could
go a long way towards preventing unexplainable hangs experienced by people. In the
case of MWI hangs, this also will mean that the SIP port isn't blocked anymore.

(closes issue #11665, reported by yehavi)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96934 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07 21:04:09 +00:00
qwell eaa6eea1b7 Fix -s socket option, and document it as well.
Closes issue #11645, patch by Laureano.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95070 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27 23:28:01 +00:00
mmichelson 1c3afe8876 Adding documentation for new manager actions and events in app_queue
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94903 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27 17:18:39 +00:00
mmichelson f18c0943a0 Merging the queue-penalty branch. In short, this allows one to dynamically adjust
the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending
on the amount of time passed. The purpose is to allow the call to open up to more (or maybe
just different) members without the caller's losing his place in the queue. See 
configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample
for how to associate a rule with a queue.

Along with the functional changes, new CLI and manager commands exist to show the rules defined and
there is an additional CLI command to reload the queue rules.

Future enhancements that may be made: support for realtime queue rules and support for dynamically adding
a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write
this myself very soon).



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94370 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21 00:44:17 +00:00
oej 82835dcbcd Add option for starting remote Asterisk by naming the actual runtime socket instead of pointing
to configuration file with -C

Reported by: sobomax
Patches: 
      asterisk.c.diff.trunk uploaded by sobomax (license 359)
      doc changes by committer
(closes issue #11598)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93854 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19 07:01:40 +00:00
oej d170d5e82b A minor update, caused by a recent bug report ;-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18 07:56:29 +00:00
rizzo b0d3c91caf small documentation update (nothing important).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-18 07:22:26 +00:00
oej b9b03966fb HUGE improvements to QoS/CoS handling by IgorG
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings

Minor modifications by me, a big effort from IgorG.
Thanks!


Reported by: IgorG
Patches: 
      qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16 10:51:53 +00:00
oej ab9c1114a7 Add a few extra headers in the voicemail users listing in
manager 1.1. Update documentation too.

(closes issue #11495)
Reported by: caio1982
Patches: 
      extra_vm_manager_info1.diff uploaded by caio1982 (license 22)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92140 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10 13:29:57 +00:00
rizzo 58ddf93b1d add a bit of info on the build infrastructure
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92084 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10 04:38:49 +00:00
qwell 20476ae522 Add count of total number of calls processed by asterisk during it's lifetime.
Add number of total calls and current calls to SNMP.

Closes issue #10057, patch by jcmoore.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91779 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07 16:11:05 +00:00