837 Commits (c4c2def7166543bdde49987be9e3737d2ee6527e)

Author	SHA1	Message	Date
Tilghman Lesher	47dd5a15af	Issue 6082 - New DTMF event for manager ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	0a9750ef9f	Merged revisions 60603 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines To be able to achieve the things that we would like to achieve with the Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	cd88d132ce	Merged revisions 60323 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60323 | murf | 2007-04-05 16:35:11 -0600 (Thu, 05 Apr 2007) | 1 line Added some clarification to the example configs for CDRs, on how to select a backend. Also, made cdr-csv the default if you 'make samples', and no other changes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	684527fcfd	Merged revisions 59452 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59452 | murf | 2007-03-29 18:56:36 -0600 (Thu, 29 Mar 2007) | 1 line A small clarification to keep bugs from being filed, and confusion from rising, if clearglobalvars is set, and globals are set in the AEL file. (9419) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	7b905e1282	Merged revisions 59040 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59040 | tilghman | 2007-03-19 10:42:26 -0500 (Mon, 19 Mar 2007) | 2 lines Fix unescaped semicolon (reported via -dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	79a3c3b9e1	Merged revisions 58957 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58957 | russell | 2007-03-15 20:42:37 -0500 (Thu, 15 Mar 2007) | 1 line fix a couple SLA documentation references ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2ea01c893c	Merged revisions 58894 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines By default, don't attempt to do any CallerID handling at all with SLA because it is known to not work properly in some situations. However, add an option to enable it for those that would like to use it anyway. The short story behind this is that to properly handle CallerID with SLA, we need the ability to change the CallerID on an existing call, and we are not ready to handle that. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2e2c6e52ee	Merged revisions 58870 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58870 | russell | 2007-03-13 18:11:08 -0500 (Tue, 13 Mar 2007) | 1 line fix the reference to the SLA documentation ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58871 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5bea998a55	Merge changes from team/russell/sqlite: ... * Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a SQLite3 database. (issue #7149, alerios) * Add new module, res_config_sqlite, which adds realtime database configuration support for SQLite version 2. I decided that this was ok since we didn't have any realtime support for version 3. If someone ports this to version 3, then version 2 support can be removed or marked deprecated. (issue #7790, rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom. Also, note that there were other modules on the bug tracker that did not make the cut because they provided some duplicated functionality. Those are: * cdr_sqlite3 (issue #6754, moy) * cdr_sqlite3 (issue #8694, bsd) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	ea226e9d77	Merged revisions 58779 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	32e03f9e4a	Add the ability to dynamically specify weights for responses to DUNDi queries. ... This can be done using a global variable or a dialplan function. Using the SHELL() function will allow you to use an external script to determine what the weight in the response should be. This can be very useful in load balancing applications. (inspired by discussions with blitzrage and jsmith in #asterisk-bugs) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	ba432b7319	Merged revisions 58119 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58119 | russell | 2007-03-06 17:00:57 -0600 (Tue, 06 Mar 2007) | 3 lines Clarify the documentation of the dialout and sendvoicemail options. (issue #9000, caio1982 and serge-v) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	1dd8e4b0b5	Remove no longer present CLI commands from sample extensions.conf. (issue #9193 reported by junky) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	746f3fcdb2	Add the missing configuration template to the sample config file. ... Thanks to Lacy Moore on the asterisk-users list for pointing out that this was missing! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	3d6e6e07ef	Merged revisions 57364 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines Merge changes from svn/asterisk/team/russell/sla_updates * Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	ae8c0f3fcb	Merged revisions 57207 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) | 2 lines minor tweaks to the sla docs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	9c58ead89b	Merged revisions 57203 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | 7 lines Merge more changes from svn/asterisk/team/russell/sla_updates * Add support for private hold. By setting "hold=private" for a trunk, only the station that put the call on hold will be able to retrieve it from hold. Also, by setting "hold=private" for a station, any call that station puts on hold can only be retrieved by that station. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	69b0eb24ed	Merged revisions 57144 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | 6 lines Merge changes from svn/asterisk/team/russell/sla_updates * Add support for the "barge=no" option for trunks. If this option is set, then stations will not be able to join in on a call that is on progress on this trunk. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	4fd59356ef	Merged revisions 57089 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) | 8 lines Merge current set of changes from svn/asterisk/team/russell/sla_updates * Add support for station ring delays. Ring delays can be set globally for a station or for specific trunks on the station. * Fix a few bugs in existing code. * Restructure and Reorganize code to improve readability and maintainability. * Improve formatting of the "sla show (trunks|stations)" CLI commands. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
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