97ab07a9e8
Allow a Skinny device to monitor a dialplan hint (w00t!).
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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
9138e53bc9
Merged revisions 56277 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines
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.
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19 years ago
006817c0e7
Merged revisions 55553 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55553 | russell | 2007-02-20 10:41:57 -0600 (Tue, 20 Feb 2007) | 3 lines
Change the formatting of sla.conf.sample to make it more readable.
(issue #9112 , blitzrage)
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19 years ago
6ad66e51ae
Allow both an external application and SMDI to do voicemail notification at the same time. (issue #8625 reported by lters)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f11d0b3d54
Merged revisions 55006 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55006 | russell | 2007-02-16 16:49:42 -0600 (Fri, 16 Feb 2007) | 17 lines
Merged revisions 55005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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19 years ago
b8ab0abb83
Allow the user to specify where to enable the respective features for when a parked call is picked up. (ie: transfers and parking)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ae6898cbe5
Add option to features.conf that enables parking via DTMF on picked up parked calls. (issue #9082 reported by francesco_r)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
1f52d1cc81
Issue #7443 - amdtech - Optionally SIP registrations in another
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realtime family.
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19 years ago
88928f67ed
Make documentation match the source code.
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19 years ago
1bf40c4da3
Merged revisions 54002 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12 Feb 2007) | 2 lines
Fix a typo where "vmpassword" should be "vmsecret"
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19 years ago
32495f91f0
Add support for outbound proxy for peers and [general]
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This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
5715b49c30
Merged revisions 53810 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
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.
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19 years ago
44c6630e4d
rename busy-limit to busy-level, since it is not a limit
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actually parse the busy-limit option from sip.conf, instead of ignoring it
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19 years ago
cfe66e6b26
Patch based on this patch with small changes for trunk...
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Merged revisions 53109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
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.
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19 years ago
0b84b386b9
Implementing "busy-limit".
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If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).
If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled.
This affects SIP subscriptions, call queues and manager applications.
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19 years ago
064e6cff1a
Merged revisions 53062 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb 2007) | 2 lines
Add explanation of port= in combination with defaultip= (thanks jsmith)
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19 years ago
174606b4bd
Merged revisions 52160 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r52160 | russell | 2007-01-24 19:37:16 -0600 (Wed, 24 Jan 2007) | 2 lines
By suggestion from kpfleming last week, change "vmpassword" to "vmsecret".
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19 years ago
34df128519
Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
641f38105a
Merged revisions 51350 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51350 | qwell | 2007-01-20 00:53:49 -0600 (Sat, 20 Jan 2007) | 5 lines
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.
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19 years ago
9e220dfd97
Merged revisions 51348 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51348 | qwell | 2007-01-20 00:16:06 -0600 (Sat, 20 Jan 2007) | 8 lines
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.
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19 years ago
10e3cba61e
Add parkedcalltransfers option for res_features. This basically enables/disables DTMF based transfers. If you want to get former behavior you will have to make sure it is enabled.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
04426fab2c
Add support for G729 passthrough with Sigma Designs boards. (issue #8829 reported by ywalther)
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19 years ago
b7ebcec300
Fix a couple of typos in the sample osp.conf.
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19 years ago
a4640ee9d8
Patch allows for changing voicemail password in users.conf from voicemail main, written by AnthonyL bug #8436
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19 years ago
fea98f6a44
Clarify what the trunkmaxsize value is in (bytes).
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19 years ago
033d849bda
Drop trunkrealloc option and just have the maximum size be a configurable option. This is per Kevin's comments on -dev and my own thoughts after I put the previous option in.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c4b4615dcd
Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by marcodmb, branch by anthonyl)
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19 years ago
cece8001dd
Merged revisions 50647 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r50647 | qwell | 2007-01-12 13:24:40 -0600 (Fri, 12 Jan 2007) | 2 lines
Update documentation to state that you shouldn't use realtime static with voicemail.conf
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
8c4c8b6648
1. Update osp module configuration file.
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19 years ago
1fe0e3d192
Merged revisions 49313 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r49313 | crichter | 2007-01-03 10:06:50 +0100 (Mi, 03 Jan 2007) | 41 lines
Merged revisions 48319,48321,48467,48552,48576,49135,49303 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line
changed a few debugs to higher debug levels
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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.
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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.
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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.
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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
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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.
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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
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19 years ago
0c3298a573
Update sample config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0375227e5c
Added some docs
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19 years ago
94d71436ec
1. Rename 'maxmessage' to 'maxsecs' to differentiate from 'maxmsg'.
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2. Rename 'minmessage' to 'minsecs' for parity.
3. Make 'maxsecs' a per-user option, in addition to global.
(Issue # 8624)
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19 years ago
1e1fd3c3e0
Integrate functionality tested on svncommunity users back into trunk
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19 years ago
29ed493b40
Be politically correct
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19 years ago
da7a35a1cc
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
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19 years ago
850dd4ea61
Use spaces as a separator for the redirect option to improve readability
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19 years ago
2c5071a006
- Convert the list of URI handlers to use the linked list macros. While doing
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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
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19 years ago
9327720c37
As per bug 7978, this version introduces the jittertargetextra option in config files
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19 years ago
437f4288cd
- Generalize the function ssl_setup() so that the certificate info
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are passed as an argument.
- Update the code in main/http.c to use the new interface
(the diff is large but mostly mechanical, due to the name change of
several variables);
- And since now it is trivial, implement "AMI over TLS", and document
the possible options in manager.conf
- And since the test client (openssl s_client -connect host:port )
does not generate \r\n as a line terminator, make get_input()
also accept just a \n as a line terminator (Mac users: do you
also need the \r-only version ?)
The option parsing in manager.conf is not very efficient, and needs
to be cleaned up and made similar to what we have in http.conf
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19 years ago
c7efdf6759
Merged revisions 48323 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48323 | russell | 2006-12-06 11:15:45 -0500 (Wed, 06 Dec 2006) | 11 lines
Merged revisions 48322 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
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19 years ago
d1b621c6a5
Adding docs on t.38
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19 years ago
3e8669595e
Merged revisions 48230 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48230 | qwell | 2006-12-04 11:54:46 -0600 (Mon, 04 Dec 2006) | 4 lines
Add documentation to voicemail.conf.sample for ODBC storage.
Issue 8499 - patch by blitzrage.
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19 years ago
c23bc46089
- Disable RTP timeouts during T.38 transmission
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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19 years ago
97614cb6b4
Merged revisions 48186 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48186 | qwell | 2006-12-01 14:25:51 -0600 (Fri, 01 Dec 2006) | 10 lines
Merged revisions 48183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines
Fix a small typo - issue 8848, reported by pabelanger
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19 years ago
4ce5b7c080
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
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- Small fixes to limitonpeer
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19 years ago
c946e3b3fb
Merged revisions 48143 via svnmerge from
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r48143 | file | 2006-11-30 12:57:35 -0500 (Thu, 30 Nov 2006) | 10 lines
Merged revisions 48142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
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19 years ago
7e46275b51
Clarify some settings for status reports in subscriptions, queues and manager
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19 years ago
e5145bebe4
Explain RTP timeouts
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19 years ago
2e7fd7cbdb
add a new http.conf option, sslbindaddr.
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Because https is more secure than http, it usually
makes sense to keep this service more open than the
one on the unencrypted port.
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19 years ago
4e47ce525b
Update docs for videosupport
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
54d44e9b00
Add ability to notify an external application/script that the voicemail password was,
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while also still changing the password "internally".
Issue 7371, initial patch by pdunkel, with rewrite/config comments by me.
Additional modifications (yay bitmask) by pdunkel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
bfd630682e
Add ability to modify range for dring matching.
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Issue #8369 , patch by ssuehring, modified slightly by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a6f5adefa1
Make it possible to enable/disable onhold tracking, in order to make life easier
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for realtime users.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a427a2a89a
- CANCEL never uses authentication
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- Add docs on canreinvite
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
64d5316a53
Add 'loose' option to joinempty and leavewhenempty which is almost exactly like 'strict' except it does not count paused queue members as unavailable. (issue #8263 reported by gnarf)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f2bc05d1d4
Feature: allow the sanity SQL to be customized per connection class (Issue 6453)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
8b8d8d0b24
Merged revisions 47279 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47279 | kpfleming | 2006-11-07 12:56:21 -0600 (Tue, 07 Nov 2006) | 2 lines
clean up sample config, and make native file playback the more obvious default choice
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19 years ago
729620dd3f
List ss7 with the rest of the valid signalling types. Group SS7 options
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together and comment them out by default.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
d900b47ccf
Adding new config option "limitpeersonly" to only apply call limits
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to the peer side of a type=friend.
This is for trying to support BJ in his quest to solve some issues
with the queue system and type=friend objects.
BJ: Please test!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0b8669b87e
document the "debug" parameter, and the change
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manager list -> manager show
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b136baaff4
Fix rport handling.
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...where did the 1.2 properties come from, really? they're back.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f98f457727
Change name of "contact" setting to "callback" which better reflects what it
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is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
95a4fc7af2
* Added option to run macro when a queue member is connected to a caller,
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see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216 , jmls reported and submitted)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4a523b1b2d
Add the ability to customize some of the prompts used within the voicemail
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application by configuring them in voicemail.conf (issue #7415 , patch by
fkasumovic, with some fixes and documentation updates by myself)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46360 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f19300635f
Merged revisions 46351-46353 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46351 | crichter | 2006-10-27 11:49:20 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46176 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line
fixed not compile issue, which was just introduced
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r46353 | crichter | 2006-10-27 12:03:23 +0200 (Fr, 27 Okt 2006) | 9 lines
Merged revisions 46350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line
fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c
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19 years ago
e85d8e98d1
document the match_auth_username option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
67926b9ac4
Update changes to do US style point code parsing/formatting (xxx.xxx.xxx)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c30f1d12c5
Ok, second attempt...
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19 years ago
25b8f577b8
On the other hand, don't use 1.4 patches for trunk... Sorry.
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19 years ago
13ea5fc0d0
Add ability to adapt the IAX trunk packets to the MTU size, to avoid bad audio
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when the number of channels fill the MTU on a given link.
In the future, this needs to be configurable per peer with trunking enabled.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c15f7953c8
Fix a few issues in the previous (disabled) HTTPS code,
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and support linux as well (using fopencookie(), which should
be available in glibc).
Update configure.ac to check for funopen (BSD) and fopencookie(glibc),
and while we are at it also for gethostbyname_r
(the generated files need to be updated, or you need
to run bootstrap.sh yourself).
Document the new options in http.conf.sample
(names are only tentative, better ones are welcome).
At this point we can safely enable the option.
Anyone willing to try this on Sun and Apple platforms ?
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19 years ago
d171c3d864
remove unused fields and unimplemented options.
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19 years ago
d8e688ece9
Merged revisions 45439 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45439 | russell | 2006-10-17 22:19:07 -0400 (Tue, 17 Oct 2006) | 2 lines
update entry to reboot a snom phone (issue #7850 , pnlarsson)
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19 years ago
a8a26ad389
Update of docs
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19 years ago
c62784c10d
In the course of a data this has been turned into an option to ignore replies, then ignore responses and finally I'm just getting rid of the option altogether and making it the default no matter what. C'est la vie!
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19 years ago
da330feb60
Merged revisions 45280 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45280 | file | 2006-10-16 16:06:18 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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19 years ago
b58cc9e1bd
Merged revisions 45262 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45262 | file | 2006-10-16 15:37:34 -0400 (Mon, 16 Oct 2006) | 10 lines
Merged revisions 45260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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.
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19 years ago
e09ad744af
Merged revisions 44561 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44561 | crichter | 2006-10-06 14:50:25 +0200 (Fr, 06 Okt 2006) | 9 lines
Merged revisions 44334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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19 years ago
77c69dc4ef
Recommend using "sip reload" since it's much easier to learn and
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remember.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b19b4b9764
document a bit the use of templates.
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They are highly convenient for writing configuration files, especially
if you have many similar entries, or want to switch quickly between
different configurations without having to comment/uncomment large
sections of the files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
f94849ca2a
document the "contact" option a bit better.
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19 years ago
ccca5843fd
Two things:
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1. slightly rearrange/simplify the parsing of the argument in sip_register.
This brings in a patch that has been in Mantis (5834) for ages,
and is the larger part of the commit;
2. implement the "contact" option for peers, similar to the one in users.conf:
If you put a "contact" option with a non-empty argument (e.g. contact=123)
in a peer section, asterisk will register with the provider as if you had a
register= username:secret@host/contact
line in the general section.
The latter is a very small is a new feature so i am not putting it
in the 1.4 branch, although the "contact" option in user.conf is
already in the 1.4 branch and so it wouldn't be too strange to
merge it.
Note that the implementation of "contact" is much simpler than
the one in 5834, and limited to a few lines in build_peer().
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19 years ago
2a7ac3f735
update example commands to match current syntax
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(does not apply to 1.4)
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19 years ago
8135d5016a
I've been meaning to add some explanation about muted... here it is
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19 years ago
6e42aa676c
CLI reverbification update to this config file
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19 years ago
8ff3dd273a
Expand setinterfacevar option to also set a variable, MEMBERNAME, which contains the member's name. (issue #8046 reported by jmls)
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19 years ago
b4ef9599de
Merged revisions 44186 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44186 | pcadach | 2006-10-03 00:52:56 +0600 (Втр, 03 Окт 2006) | 1 line
Missed part of userconf functionality for chan_h323
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19 years ago
e5203bb283
Add option to logger to rename log files with timestamp (issue #8020 reported by jmls)
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19 years ago
cc1945ce1b
Add option 'keepstats' which will keep queue statistics during a reload. (issue #7908 reported by jmls)
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19 years ago
8a5cf10121
Merged revisions 44111 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r44111 | russell | 2006-10-01 11:20:12 -0400 (Sun, 01 Oct 2006) | 11 lines
Merged revisions 44110 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r44110 | russell | 2006-10-01 11:19:23 -0400 (Sun, 01 Oct 2006) | 3 lines
Fix the name of the "eventmemberstatus" option in the sample queues.conf
(issue #8065 , adamg)
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19 years ago
9cf1f14ed5
Handle HOLD/RETRIEVE notifications
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19 years ago
2b7debf368
Merged revisions 43739 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43739 | murf | 2006-09-26 20:32:47 -0600 (Tue, 26 Sep 2006) | 1 line
This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
fe1a3b8877
Add optional queue_log_name config option for logger.conf, to change the
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name of the queue_log file.
Issue #7363 , patch by Steve Davies, slightly modified by me.
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19 years ago
c479c25182
Support for negotiation and receiption of Cisco's RTP DTMF
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19 years ago
1af96a0b21
Specify RFC2833 payload on dtmfmode option rather than dtmfcodec option (deprecated)
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19 years ago
859bc68383
Merged revisions 43464 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43464 | tilghman | 2006-09-21 18:24:41 -0500 (Thu, 21 Sep 2006) | 2 lines
Twould help if we actually documented how the new features in res_odbc actually work. (Oops)
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19 years ago
bd76cda68d
Merge in SS7 changes.... need to still cleanup zapata.conf
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19 years ago
8bd82ebc0d
Add documentation on rtp packetization.
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Technically, ooh323 doesn't support it yet, but there is a patch that should be committed very soon.
Issue #7989 , patch by DEA, slightly modified.
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19 years ago
3c224654c2
Document member name logging functionality.
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19 years ago
2e5118cc49
Add the h323 config file. Arrr!!! for international talk like a pirate's day.
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19 years ago
ec4bf7a849
seperate jingle and gtalk so it will be easier to track
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changes in both of the moving specs. Currently chan_gtalk is
compatible with the latest gtalk/libjingle version, and chan_jingle
needs a lot of work.
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19 years ago
4ed1578104
Clarified the meaning of the callwaiting variable in the zapata.conf file.
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19 years ago
956b837a41
Merged revisions 43159 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43159 | file | 2006-09-18 11:05:39 -0400 (Mon, 18 Sep 2006) | 2 lines
Add number unobtainable tone for New Zealand (issue #7969 reported by nic_bellamy)
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19 years ago
ee27f9efee
Remove the suggestion of realtime hints, since that functionality will not be available until post-1.4
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19 years ago
c2d959c2f9
Improve documentation of users.conf items.
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19 years ago
6d5809297b
Skinny hold support.
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Original patch by wedhorn, with modifications by me.
Issue #7588
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19 years ago
c2c4f86c72
merge markster's usersconf branch with some slight changes
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19 years ago
091e1aed8d
Merged revisions 42716 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42716 | tilghman | 2006-09-11 11:39:06 -0500 (Mon, 11 Sep 2006) | 2 lines
Spelling/grammar fixes (Issue 7929)
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19 years ago
b0666488f3
Merged revisions 42697 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42697 | tilghman | 2006-09-11 09:40:13 -0500 (Mon, 11 Sep 2006) | 2 lines
Two grammar issues (bug 7927)
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19 years ago
e02897acd4
Board numbers and channel numbers are now 0 based, since vpb driver
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version 3.0 (released December 2005)
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19 years ago
75822388a4
Make sure we give a little warning about the echotraining option
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19 years ago
34eb4f54ba
Use lower case 'x' instead of a UTF-8 character (issue #7888 reported by flefoll)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
47d8e14871
Comment out default from extensions.ael
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
4ba458e352
Merged revisions 42014 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r42014 | qwell | 2006-09-05 09:27:46 -0700 (Tue, 05 Sep 2006) | 4 lines
Small typo in zapata.conf.sample
Reported by ppyy in 7881
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
3f888b84f3
Use GLOBAL() in dialplan examples
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
c6977b9983
Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
b281acf0f8
change default setting for autofallthrough
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
0850bb72ed
Kevins last commit made me spot a typo.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
ece7018515
add one remaining bit of functionality to the features.conf applicationmap (from Matt Nicholson in Digium Express Services)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
7b65cdc0c7
remove documentation of 'global' section in modules.conf, since it is no longer needed or supported
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
236c23269d
Merged revisions 40979 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40979 | file | 2006-08-24 13:13:04 -0400 (Thu, 24 Aug 2006) | 2 lines
I can type english. Honest. Thanks Kenny.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
e5c0665caf
Merged revisions 40971 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40971 | file | 2006-08-24 12:44:41 -0400 (Thu, 24 Aug 2006) | 2 lines
Minor documentation fix to add the 'dynamic' dialplan option from angler
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
749cd217c3
Merged revisions 40392 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40392 | kpfleming | 2006-08-18 14:10:51 -0500 (Fri, 18 Aug 2006) | 2 lines
make a feeble attempt to avoid the 'how do I enable my hardware echo canceler' questions
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
87ac16847e
- unregister SLA apps on module unload and add sample config (issue #7701 , junky)
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- rename SLAS to SLAStation, and SLAT to SLATrunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
19 years ago
a0bd41f79b
Add support for Sigma Designs cards. These basically allow you to offload dialtone generation to the board. If you're using a quicknet board where this might work, give it a try as well. (issue #6092 reported by ywalther - minor mods by moi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0f0323dbab
Clarify volgain option a bit, it needs sox to work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f37a4e3e12
Bug 6237 - add volgain parameter, such that voicemail messages may be amplified after recording
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
9d26f46fc7
remove some extraneous 'followme' in prompt names
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
2f69bec40e
Add placeholder for sla.conf sample in configs/. Please update with
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correct information not to overload the switchboard with calls to sales.
Thank you! ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4d7c67fc72
Merge my applicationmap_fixup branch to address the issues described in this
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post to the asterisk-dev mailing list:
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022174.html
This implements full control over both which channel(s) can activate a dynamic
feature, as well as which channel to run the application on. I also updated
the documentation on the applicationmap in features.conf.sample in hopes that
the configuration is more clear.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0d902b3033
Update documentation on realtime; add a workaround for lack of realtime hints by using func_odbc
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6d0742fc16
merge Russell's 'hold_handling' branch, finally implementing music-on-hold handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
837910062b
First pass at in-place file manipulation via manager
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e69e056012
config sample for the previous, regarding ADSI
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4376af0080
actually make the non-standard G726-32 behavior available for SIP clients
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
54ce0f0a22
added even more statefulness for sending out disconnect/release/release_complete messages. added support for incoming presentation/screening. fixed a bug that we generate TONE_EVENTS on hanguptone_indicatem, which caused asterisk to write blocking thread messages. added nodialtone option to prevent dialtone for always_immediate
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3b5a2aafa4
- Change filename to current file name
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
ba092c1244
And now the trunk version! Add an option for IAX2 users that allows you to set how many outstanding AUTHREQs chan_iax2 will wait for replies on.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0e0059c0f3
Remove configuration option "restrictcid" that is nowhere to
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be seen in the code. Did it exist, was it planned to exist
or was it documentationware only? Ask Dr Asterisk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
de03118578
Asterisk portion of the T309 patch. ( #7271 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bd0b801a0d
* removed tone_indicate, we genrate only the dialtone by ourself (and the hanguptone of course)
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* removed the state handling from release_chan, and simplified the ast_hangup/ast_queue_hangup stuff
* added pp_l2_check option, for pp lines where the pbx does not initially gets the L2 up
* simplified and fixed a bug in the pid generation code
* fixed a bug in empty_chan, which might cause segfaults and memorry corruptions
* added prepare_bc function, which is sort of the opposite of empty_bc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
ec9d4711d7
- Add notes about voicemail depending on res_adsi
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- Remove obsolete modules from modules.conf.sample
(make install will warn if those exist on the machine)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b971f65978
- Make use of system name in realtime SIP peers optional
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- Fix small issue with SIP history
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f3594bd1a0
Removing configuration options that does not do anything yet. No need to
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add "promises" to the sip.conf.sample...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f5c0cd2ddc
added better L2 handling for ptp, if it's down we don't try to call on that port in groupdial anymore, also we try to get it up then. Additionally added the configoptions ntdebugflags and ntdebugfile to debug the mISDNuser NT Stack (should have done that ages before..). isdn_lib.c compiles again.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dec3d7d4c0
Merged revisions 36253-36254 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r36253 | kpfleming | 2006-06-29 02:19:27 -0500 (Thu, 29 Jun 2006) | 2 lines
add documentation for peer-specific 'outboundproxy' setting
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r36254 | kpfleming | 2006-06-29 02:19:54 -0500 (Thu, 29 Jun 2006) | 2 lines
clarify documentation for 'persistentmembers' setting
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36262 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4177596e8d
reformatting sip.conf.sample a bit, adding dumphistory that was not documented
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
cc43f0bdc7
Speling error. Avoid swenglish :-) (thanks, jtodd!)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6399ac438d
Add explanation and warning about the "s" extension. (Hi Mike :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f8311adcda
METERMAIDS:
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-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e2b0c5b558
Add example of permit/deny to sip.conf.sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
89f6ffe1e5
Bug 6589 - option to display channel variables in queue events
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4c066de7cf
Merged revisions 35334 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r35334 | file | 2006-06-21 16:25:34 -0300 (Wed, 21 Jun 2006) | 2 lines
Add Venezuelan indications (issue #7402 reported by palillo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
a5d6979fac
Finally merge chan_skinny fixes into trunk.
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ready...set...GO!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
035a8b4278
Merged revisions 34627 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r34627 | russell | 2006-06-18 16:15:15 -0400 (Sun, 18 Jun 2006) | 5 lines
don't store multiple secrets delimited with semicolons for peers because this
is only valid for users. Instead, only keep the last specified secret for a
peer entry. Also, document how multiple secrets are handled in the sample
config. (Reported by PCadach on #asterisk-bugs)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0e5e744fb2
Add bulgarian indications (issue #7314 reported by KNK)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5456f425c6
Allow AST_FRAME_MODEM frames to be dumped, and document T.38 passthrough support
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1e530787f3
solves some bugs with memory allocation, and adds
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a message stack.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bc49d5bfb3
moh files will now be distributed in native format, not mp3, so...
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update sounds/Makefile to download/unpack the proper files
remove all evidence of mpg123 from the main Makefile
change the sample musiconhold.conf file to use native mode instead of mp3
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3d973a0686
Introducing app_followme into /trunk!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dd6de5ee4e
it's time... only enable global priority jumping if the config file says to do so
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4e74b8fb75
Issue #7231 - Missing indications from libtonezone (tzafrir)
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- "tuttter" changed to "stutter" ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
2dc6947144
Issue #2863 - Improved RTCP support (John Martin, Fredrik Olsson)
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Thanks to everyone involved in working with this!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
18b135f215
oops my config file was out of date not the sample
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
58cd5f2440
oops fixing example config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4c76028de9
- add the ability to configure forced jitterbuffers on h323, jingle,
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and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257 , north, with additional updates and modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4be235a974
added bearer capability reject support. we send release instead of disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6bce269454
Merged revisions 31321 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31321 | kpfleming | 2006-06-01 07:41:47 -0500 (Thu, 01 Jun 2006) | 2 lines
remove a sample entry that never should have been added (code to support it was not merged)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8cfb992c1e
adds statusmessage customization from Julian Lyndon-Smith
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and fixes bug with pruneregister
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e25c4621b4
Nobody saw this coming, I bet.
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jitterbuffer!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bb7dd96cfe
Add support for using a jitterbuffer for RTP on bridged calls. This includes
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854 , Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
18606233da
fix various typos and other bits (from Ian Kinner)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
18248f092f
Merged revisions 30239 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r30239 | file | 2006-05-25 12:27:44 -0300 (Thu, 25 May 2006) | 2 lines
Get rid of an incorrect SIP dial string in the sample extensions.conf - I even tried variations... no go (issue #7222 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8122c35675
fixed to early connect bug which came in yesterday.., also added the transmit of progress indicators through channel vars
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3e99be68d1
add a new option for 'obscuring' SIP user/peer names from fishers
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use an enum for authentication results and clean up code
fix a bug where SUBSCRIBE for an unknown user/peer would not generate a response
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
45107ed763
allows for configurable answer timeout on attended transfer
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patch 0006763 with minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7aa1a77e75
asterisk-xmpp merge in
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5235890be4
This is part 2/2 of the patches for #7090 . Adds one-step call parking to /trunk via builtin functions and 'k' 'K' application options added to app_dial. This also resolves #6340 .
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
19d46333bf
added callcounters for incoming and outgoing calls
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
9e81cc3e0c
Escaping commas within fields isn't always desireable.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1fcc86d905
Add support for logging CDR recrods to a radius server (issue #6639 , phsultan)
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- with contributions from miconda, jcollie, and sb
- branch maintained by oej
Thanks everyone!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
42cf0b0a8f
add another media path reinvite 'flavor', where we will only redirect our media to devices that we know are not behind a NAT (based on the evidence collected when we receive media from them)
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also, documented the 'canreinvite=update' option in the sample config file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6d603ec09c
Allow contexts in regexten so that extensions can be added to multiple contexts when peer registers (issue #6869 reported by and created by Marquis)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@28168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
15358932ec
Add distinctive ring detection with Caller ID for Australia, New Zealand, and other countries. (issue #3596 reported by deon patch by dbowerman with minor mods by moi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5237a0e06d
- Use systemname for realm in sip, if we have no configuration for realm
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- Optionally send systemname in manager (cool when you have a manager proxy)
- Use systemname in CLI prompt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5fb4e7019f
and chan_iax2 gets smaller... remove the old jitterbuffer
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e0f0f4b4a4
german syntax for numbers from christian richter
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26527 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
66ed134473
Allow media to go directly between IAX endpoints while signalling still
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goes through the existing path.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
ca6cf552f9
Add documentation on "allowtransfer"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25614 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3e2079e46c
Fix output delimiters and add prefix parameter to func_odbc #7025 (Corydon76)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
d83bd4d136
Integrate the MixMonitor functionality (introduced in 1.2) as an option for recording queue member conversations with callers. #7084
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0794168428
add support for having the user reminded that their temporary greeting
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is still set (issue #6120 , khaefner)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
85e0c889e4
Allow for the execution of an AGI to the caller's channel right before they get bridged with the queue member that is going to take their call. Add the option to set a MEMBERINTERFACE variable on the caller's channel that will contain the interface of the queue member that is going to/did take the call. #6843
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@25056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
efccf89eae
Added option far_alerting. This option makes it possible to generate a Ringing on other channels if they feel that they should have inband ringing, but there is non in reality. I need this due to the fact that asterisk has not the possibility to transmit progress indicators thus chan_sip and others do not know wether they should generate a Rining tone themselves if they receive AST_CONTROL_RINGING..
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7b3f3db65d
Fix autofill behavior in app_queue and document it's functionality in queues.conf.sample and UPGRADE.txt
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7bbb6bd3aa
- fix typo in rtp.c, devicestate.h
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- add information about subscriptions and realtime dial plans in sip.conf.sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
c38fbd246e
note that group assignments must be from 0 to 63 (issue #7048 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0b6bd0073b
put the default misdn.trace to /var/log/asterisk/misdn.log for better integration of existing log structure
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5f58cc8770
Merge Steve Murphy's (murf) complete re-implementation of AEL, which is now no longer considered experimental :-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e3f569532f
Deprecate prefixed options in voicemail
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5873462c2e
- Add doxygen documentation for sipsock_read locking
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- Improve documentation of pedantic
(related to issue #7016 )
From the air above Russia...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
68730ba487
update configuration, generalize date format and
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start mapping digit strings
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
64fbe4cbc5
add example syntax for new-style number and date spelling
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@21420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
e8a94a71e2
Allow the attachment format to be specified differently for different mailboxes (issue #6961 reported by the ever fabulous Corydon76)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8a5436c72f
add indications for Thailand (issue #6971 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
717445c1d8
add the ability to turn off the feature that allows agents to end calls
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by pressing '*'. This is still on by default. (issue #6897 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b04c61eeb3
Note that the res_speech module will need to be loaded first, and add a conveient line to uncomment to do so for the time being.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
afcefc4a68
Convert chan_iax2 to use linked lists for multithreading, and add dynamic threads. These are used when all pool threads are in use, and will stick around until load dies down. The theory is that during high load you'll have more threads available, and during low load you'll only have the normal pool threads sticking around.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7089dc1341
Issue #6899 - remove OSP support code from chan_sip.c and app_dial.c
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- implement all functions through internal APIs in res_osp.c and app_osplookup.c
(homesick)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
9d8260c68e
Formatting fixes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8e22245b09
Formatting fixes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
023e27f695
Formatting fixes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
95de51526a
Added information on call-limit and realtime
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bf4b484e62
Clarify the need for numeric parking positions (imported from 1.2)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bfba044b5f
Flesh out the remainder of the manager + http changes and create a sample application to partially
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demonstrate the capability of manager over http.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
cc0b49d927
Provide warning about current behavior of autofill = yes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
eb94c40702
Typo
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
139b07c76c
whitespace "fixes", and general cleanup
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It's nice to have consistency in sample configs too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
c01fc0ee03
the comment character is ';' not '#' ...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
150e0b72cc
Added more "valid" phone types to skinny sample config.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
08df3610a6
update example file
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
41f8e3728e
disable the http server by default at the request of people on IRC
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8410e0d681
support subscription-based MWI, and use proper Call-ID on NOTIFY messages (issue #6390 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
278b8e8fc7
improve IP TOS support for SIP and IAX2 (issue #6355 , code from jcollie plus modifications)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
83d9331261
Issue #5427
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- Enable videosupport per device
- Implement maxcallbitrate setting for video calls
Patch by John Martin, thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@15148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
18de2b7787
Issue #6705 (oej)
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- 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: https://origsvn.digium.com/svn/asterisk/trunk@15107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
9164eac21a
Add micro-http server and abstract manager interface, make snmp not die
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on reload.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
ba8f7b8819
Allow channels to be moved if channel change is requested in SETUP_ACK, also add a WAY cool new field to the nsf option
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7cfb9b3515
Added separate outsignalling specification, and fixed FEATDMF to allow for
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international inbound calls.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1abe304427
add a CLI command that allows converting files to other formats using
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the Asterisk file format and codec translator modules (issue #6062 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f882197157
Merged revisions 13964 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
c114587344
add indications for Malaysia (issue #6758 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
52eb1ad9d1
removed dynamic switching from transparent to hdlc mode. Instead we've got a config option hdlc=yes now which enables the hdlc controller for a data call
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
a0800bd179
these traceing option do not exist anymore
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
d7b5a18f4c
Fix reference to README files
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0efbe1aa5d
Add reference to examples for files and custom, too make it more obious
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that you're required to read on... (hello xrobau)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1a206c1bf8
Clarify documentation for "progressinband" - imported from 1.2
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
9df72acbe9
deprecate the mailboxdetail option and always use its behavior, instead (issue #6665 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4e6af293f9
add an option to cdr.conf that enables ending CDRs before executing
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the "h" extension as opposed to afterwards (issue #6193 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8e7dd52695
added option to change the connected party number dialplan (ton)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dad9d7709b
allows the table field to be configurable for
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cdr_tds.conf. patch provided by bug 6629 with
minor change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
21735de56d
added a bit more detailed description for the echotraining parameter, also changed the default from 1 to 2000. The default for the upper_threshold is now 0
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
99206286fb
Merged revisions 11946 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7377ebbd2e
cdr_csv logging parameters in cdr.conf
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usegmtime, log date/time in GMT
loguniqueid log uniqueid
loguserfield log user field
patch provided by bug 5015
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6b8701cffa
Whitespace changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bd9c89a710
better default values for jitterbuffer in code and config
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
6a86c7c5c9
Add SNMP support (bug #6439 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
16109b9d2c
Make IAX2 multithreaded
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@11192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7092b4475c
Merged revisions 10511,10535,10736 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
fecae4f64e
Changing syntax once again slightly and standardizing
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config to other asterisk samples , bug note 6530
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
afaf8e4c04
adde incoming_early_audio option, to avoid sending tone indications to the remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f6bd1b8559
added pmp_l1_check option, to avoid l1 checking for group calls on PMP ports
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
f0495e8944
add option to avoid calling members whose channels are 'in use' (issue #6315 , plus documentation)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
49a04b91a2
changed naming scheme for variables so it matches
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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: https://origsvn.digium.com/svn/asterisk/trunk@10086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b9918fb16b
set properties for new files (i need to get this documented)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dacbca4699
Commiting 5959 with minor formatting and typo
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fixes. Thanks to all those involved.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
973c12effd
Bug 6477 - minor syntax error, plus a few other syntax fixes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
af07dc8883
Add smdi support for asterisk (see doc/smdi.txt for config info) ( #5945 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b40bd71a9a
restore 'rfc2833' naming for DTMF mode in chan_sip
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4d07b89fdd
- Change "rfc2833" to "rtp" in sip.conf. Keeping backwards compatibility.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
24ceb84434
- Adding example on using european time zones in voicemail.conf
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- Moving the timezone section to below the documentation section
Example written and contributed by Henrik Olsen, Astricon Training student.
Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b42dd639ee
default values of jitterbuffer and jitterbuffer_upper_threshold should be > 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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
7133d1b006
* removed unnecessary struct elements and functions
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* 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: https://origsvn.digium.com/svn/asterisk/trunk@9114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8204f581ab
- Adding a doc/00README.1st with an INDEX over README files
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- Moving files from / to /doc or /configs
- Renaming some documentation files
Thank you for the initiative, manxpower!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3f6cc5c544
- Clarify default setting of canreinvite (thanks royk)
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- Add some extra headers and reference to other doc/ files for realtime
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
125fd8446c
Issue 5892: Set a minimum T1 timer for calls. Reporter: twisted
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b64404e039
From now on, apply maxexpiry and minexpiry to all subscriptions. Thanks to fourcheeze in the IRC channel
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for pointing this out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0ba27e0a6b
Make it clear that caller ID in sip.conf is used only on incoming calls (inspired by bug #6183 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
83ab52cdc4
changed some settings to app_args and some code
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cleaning patch 6267
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
4dc76fbcc1
Fix comments in sip.conf ( #6134 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
bab4d7a6b9
Update config. Apprently default doesn't always work now.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
125db028c3
- Add DOC file about caller ID presentation values
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- Add callingpres to sip.conf
- Add reference to README.callingpres from zapata.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3d456ce4f0
Add support for "musicclass" instead of "musiconhold" to make chan_zap compatible
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with other channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
2a9a03e6f0
Bug 5090 - sample configuration for udptl packets
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b5ed1a1a59
Add mission options to agents.conf sample file ( #6234 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
06f2040e6f
added feature for pausing and unpausing the
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monitor app from manager and in the call through
features.conf bug 5395 for the patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1874f21ff8
Implement the autologoffunavail option in chan_agent ( #6038 with some minor mods)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
130ce286ba
added two new features to meetme, autofill and
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autopause. thanks twisted from 5577
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8d4ee3882a
forgot to take out createlink sample along with
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other part.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1dc0312d01
Added option for limiting a user from logging in
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as multiple agents at same time. bug 6046 thanks
ppyy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
64f01df137
added page macro examples from 6077 with minor typo
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fixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1db503ade6
Merged revisions 7999 via svnmerge from
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/branches/1.2
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
8f7647b49f
Make sure to update the config file for chan_zap to include japanese caller id
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
353e5f1dc9
added small feature from bug 5682 with one typo fix.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
5462ec082c
- Remove "incominglimit" as a configuration option in sip.conf
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- Add documentation on call-limit, explaining that there's two counters
for a type="friend".
- Document the removval of "incominglimit" in UPGRADE.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
a1a5e746ee
add optional timestamps to manager events (issue #5535 , simplified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
3b4f660a85
Bug 5345; Add configuration option for minimum registration time. (folsson)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
ca92ea82d6
Bug 5854 - Allow quotation of the second field in extconfig, for the purpose of including commas (for LDAP configuration)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
dd161bec3b
Add SQL_ESC to allow single ticks to be escaped
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
a945b5598e
Added support for Feature Group C CAMA trunks (both DP and MF), and support
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for Feature Group D Tamdem trunks inbound (outbound was already there)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b60daeb58f
Merged revisions 7599 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7599 | russell | 2005-12-22 15:36:47 -0500 (Thu, 22 Dec 2005) | 3 lines
revert changes to videosupport to allow per-peer setting, since it isn't quite
complete and there is not an obvious fix at this point
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
28af98ae95
Bug 5055 - Simple SQL commands from the dialplan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
da1d91b688
move heading to the top of the file
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
1187d236bc
Added docs for new features (duplex mode & tailmessages)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
d37857c208
updated the documentation and the sample config to meet the present
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
9c8d7286dd
Document other use of courtesytone, make association of "One Touch Record" with "Touch monitor"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
c7cda51ffc
Document two keywords that were previously missing
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
b6b32ebd91
Merged revisions 7285,7299,7310,7329 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7285 | tilghman | 2005-12-02 15:12:05 -0600 (Fri, 02 Dec 2005) | 2 lines
Turn on executable bits for startup scripts, and fix bash var interpolation for Mandrake
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r7299 | oej | 2005-12-02 19:24:40 -0600 (Fri, 02 Dec 2005) | 2 lines
Documenting the default registerattempts setting as 0, continue hammering the server for ever and ever ;-)
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r7310 | tilghman | 2005-12-03 13:55:05 -0600 (Sat, 03 Dec 2005) | 3 lines
Bug 5925: check for "Unknown", as that's what app_voicemail puts into the field for Unknown callerid
Also, remove useless res checks (initialized to 0; never set)
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r7329 | kpfleming | 2005-12-04 12:03:07 -0600 (Sun, 04 Dec 2005) | 2 lines
use a more efficient way to get the revision number, that will also report if the working copy contains uncommitted modifications
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
0242a69b3f
Merged revisions 7265-7266,7268-7275 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
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r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875 )
remove support for 'patches' subdirectory, it's no longer useful
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r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788 )
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r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882 )
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r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810 )
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r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879 )
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r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870 )
reformat password changing code to conform to coding guidelines (issue #5870 )
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r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899 )
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r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888 )
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r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907 . Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
20 years ago
Jason Parker
Russell Bryant
Joshua Colp
Olle Johansson
Kevin P. Fleming
Matt O'Gorman
TransNexus OSP Development
Christian Richter
Tilghman Lesher
Steve Murphy
Luigi Rizzo
BJ Weschke
Matthew Fredrickson
Paul Cadach
Mark Spencer
Voicetronix Support
North Antara
Josh Roberson
Jim Dixon