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${ noResults }
818 Commits (726928cb6091abe5a90403993b2b584e347c0927)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
a3da18c244 |
Issue 7789 - some telcos want the TON set based on the number, but without the NANP prefix removed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Jason Parker
|
97ab07a9e8 |
Allow a Skinny device to monitor a dialplan hint (w00t!).
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 |
Russell Bryant
|
9138e53bc9 |
Merged revisions 56277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
006817c0e7 |
Merged revisions 55553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Joshua Colp
|
6ad66e51ae |
Allow both an external application and SMDI to do voicemail notification at the same time. (issue #8625 reported by lters)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55410 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
f11d0b3d54 |
Merged revisions 55006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4, and trunk. I decided that once a conference is created from meetme.conf, it is acceptable behavior that the pin can not be changed until the conference goes away. I also added a note in meetme.conf to describe this behavior. We still have another issue in 1.4 and trunk where some conferences with no users don't go away. That is the real bug that needs to be addressed here. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55007 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
b8ab0abb83 |
Allow the user to specify where to enable the respective features for when a parked call is picked up. (ie: transfers and parking)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
ae6898cbe5 |
Add option to features.conf that enables parking via DTMF on picked up parked calls. (issue #9082 reported by francesco_r)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Olle Johansson
|
1f52d1cc81 |
Issue #7443 - amdtech - Optionally SIP registrations in another
realtime family. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
88928f67ed |
Make documentation match the source code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54379 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
1bf40c4da3 |
Merged revisions 54002 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12 Feb 2007) | 2 lines Fix a typo where "vmpassword" should be "vmsecret" ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
32495f91f0 |
Add support for outbound proxy for peers and [general]
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 |
|
Russell Bryant
|
5715b49c30 |
Merged revisions 53810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Kevin P. Fleming
|
44c6630e4d |
rename busy-limit to busy-level, since it is not a limit
actually parse the busy-limit option from sip.conf, instead of ignoring it git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53577 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
cfe66e6b26 |
Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
0b84b386b9 |
Implementing "busy-limit".
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
064e6cff1a |
Merged revisions 53062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
174606b4bd |
Merged revisions 52160 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52161 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51560 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Jason Parker
|
641f38105a |
Merged revisions 51350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Jason Parker
|
9e220dfd97 |
Merged revisions 51348 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
04426fab2c |
Add support for G729 passthrough with Sigma Designs boards. (issue #8829 reported by ywalther)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
b7ebcec300 |
Fix a couple of typos in the sample osp.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Matt O'Gorman
|
a4640ee9d8 |
Patch allows for changing voicemail password in users.conf from voicemail main, written by AnthonyL bug #8436
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
fea98f6a44 |
Clarify what the trunkmaxsize value is in (bytes).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
c4b4615dcd |
Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by marcodmb, branch by anthonyl)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50676 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Jason Parker
|
cece8001dd |
Merged revisions 50647 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
TransNexus OSP Development
|
8c4c8b6648 |
1. Update osp module configuration file.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49491 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Christian Richter
|
1fe0e3d192 |
Merged revisions 49313 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ r48319 | crichter | 2006-12-06 15:35:25 +0100 (Mi, 06 Dez 2006) | 1 line changed a few debugs to higher debug levels ........ r48321 | crichter | 2006-12-06 16:48:45 +0100 (Mi, 06 Dez 2006) | 1 line added the export and import of the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the extension is already completely dialed or if there might come additional digits by information elements. also added some docs for that. ........ r48467 | crichter | 2006-12-14 14:03:49 +0100 (Do, 14 Dez 2006) | 1 line removed FIXUP state. added check for channel allocation conflict when we create a setup while the other site creates a setup on the same channel, besides the check we resolve this conflict. ........ r48552 | crichter | 2006-12-18 11:19:39 +0100 (Mo, 18 Dez 2006) | 1 line when our PTP Partner sends us a SETUP with a preselected channel we just accept it, even when we're NT. added some checks for segfaults. ........ r48576 | crichter | 2006-12-19 14:08:51 +0100 (Di, 19 Dez 2006) | 1 line when we reject a channel, because it's in use already, we shouldn't process the setup anymore. made the channel allocation a bit easier and more understandable, removed a few unused lines ........ r49135 | crichter | 2007-01-02 11:07:22 +0100 (Di, 02 Jan 2007) | 1 line added check for channel ranges in the set/empty channel functions. set pmp_l1_check default to no. added misdn restart pid cli command. added cleaning of channel when we send a RELEASE_COMPLETE. ........ r49303 | crichter | 2007-01-03 09:24:00 +0100 (Mi, 03 Jan 2007) | 9 lines * Added check for bridging in misdn_call to avoid setting echocancellation when 2 mISDN channels are involved and when bridging is set. That lead to a kernel panic before under different situations, because we switched about 2 times between hardware bridging and echocancelation * readded MISDN_URATE variable which got lost before, this should make app_v110 work again * fixed typo ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
0c3298a573 |
Update sample config
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
0375227e5c |
Added some docs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Tilghman Lesher
|
94d71436ec |
1. Rename 'maxmessage' to 'maxsecs' to differentiate from 'maxmsg'.
2. Rename 'minmessage' to 'minsecs' for parity. 3. Make 'maxsecs' a per-user option, in addition to global. (Issue # 8624) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Tilghman Lesher
|
1e1fd3c3e0 |
Integrate functionality tested on svncommunity users back into trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
29ed493b40 |
Be politically correct
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
da7a35a1cc |
Add support for buggy Cisco MWI firmware > 8.0.3 (issue 8575 - flewid)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
850dd4ea61 |
Use spaces as a separator for the redirect option to improve readability
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
2c5071a006 |
- Convert the list of URI handlers to use the linked list macros. While doing
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
19 years ago |
Steve Murphy
|
9327720c37 |
As per bug 7978, this version introduces the jittertargetextra option in config files
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
Luigi Rizzo
|
437f4288cd |
- Generalize the function ssl_setup() so that the certificate info
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Russell Bryant
|
c7efdf6759 |
Merged revisions 48323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option in the sample configuration file. (issue #8526, arkadia) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48325 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
d1b621c6a5 |
Adding docs on t.38
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Jason Parker
|
3e8669595e |
Merged revisions 48230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48231 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
c23bc46089 |
- Disable RTP timeouts during T.38 transmission
- 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Jason Parker
|
97614cb6b4 |
Merged revisions 48186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines Fix a small typo - issue 8848, reported by pabelanger ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48187 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
4ce5b7c080 |
- Remove T.38 early media, since T.38 requires two way communication (imported from 1.4)
- Small fixes to limitonpeer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Joshua Colp
|
c946e3b3fb |
Merged revisions 48143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
7e46275b51 |
Clarify some settings for status reports in subscriptions, queues and manager
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
Olle Johansson
|
e5145bebe4 |
Explain RTP timeouts
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |