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2007-04-28Merge changes from team/russell/eventsRussell Bryant
This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27Add a min-announce-frequency option to queues.conf which allows you to ↵Russell Bryant
control the minimum amount of time between queue announcements for use when the caller's queue position changes frequently. (issue #9604, patch by Matthew Roth) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-18Mini-voicemail - an embryo for a new voicemail system based on buildingOlle Johansson
blocks instead of one large monolithic app. Supports multiple templates and is designed mostly for voicemail delivery over e-mail. There's a todo with a list of ideas in the source code if you want to contribute. Feedback is appreciated! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Issue 6082 - New DTMF event for managerTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Merged revisions 60603 via svnmerge from Russell Bryant
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
2007-04-05Merged revisions 60323 via svnmerge from Steve Murphy
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
2007-03-30Merged revisions 59452 via svnmerge from Steve Murphy
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
2007-03-19Merged revisions 59040 via svnmerge from Tilghman Lesher
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
2007-03-16Merged revisions 58957 via svnmerge from Russell Bryant
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
2007-03-14Merged revisions 58894 via svnmerge from Russell Bryant
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
2007-03-13Merged revisions 58870 via svnmerge from Russell Bryant
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
2007-03-13Merge changes from team/russell/sqlite:Russell Bryant
* 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
2007-03-12Merged revisions 58779 via svnmerge from Joshua Colp
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
2007-03-07Add the ability to dynamically specify weights for responses to DUNDi queries.Russell Bryant
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
2007-03-06Merged revisions 58119 via svnmerge from Russell Bryant
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
2007-03-05Remove no longer present CLI commands from sample extensions.conf. (issue ↵Joshua Colp
#9193 reported by junky) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-03Add the missing configuration template to the sample config file.Russell Bryant
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
2007-03-01Merged revisions 57364 via svnmerge from Russell Bryant
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
2007-02-28Merged revisions 57207 via svnmerge from Russell Bryant
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
2007-02-28Merged revisions 57203 via svnmerge from Russell Bryant
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
2007-02-28Merged revisions 57144 via svnmerge from Russell Bryant
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
2007-02-28Merged revisions 57089 via svnmerge from Russell Bryant
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
2007-02-27Issue 7789 - some telcos want the TON set based on the number, but without ↵Tilghman Lesher
the NANP prefix removed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Allow a Skinny device to monitor a dialplan hint (w00t!).Jason Parker
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
2007-02-22Merged revisions 56277 via svnmerge from Russell Bryant
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
2007-02-20Merged revisions 55553 via svnmerge from Russell Bryant
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
2007-02-19Allow both an external application and SMDI to do voicemail notification at ↵Joshua Colp
the same time. (issue #8625 reported by lters) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16Merged revisions 55006 via svnmerge from Russell Bryant
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
2007-02-16Allow the user to specify where to enable the respective features for when a ↵Joshua Colp
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
2007-02-16Add option to features.conf that enables parking via DTMF on picked up ↵Joshua Colp
parked calls. (issue #9082 reported by francesco_r) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-15Issue #7443 - amdtech - Optionally SIP registrations in another Olle Johansson
realtime family. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14Make documentation match the source code. Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12Merged revisions 54002 via svnmerge from Russell Bryant
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
2007-02-11Add support for outbound proxy for peers and [general]Olle Johansson
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
2007-02-10Merged revisions 53810 via svnmerge from Russell Bryant
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
2007-02-08rename busy-limit to busy-level, since it is not a limitKevin P. Fleming
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
2007-02-02Patch based on this patch with small changes for trunk...Olle Johansson
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
2007-02-01Implementing "busy-limit".Olle Johansson
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
2007-02-01Merged revisions 53062 via svnmerge from Olle Johansson
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
2007-01-25Merged revisions 52160 via svnmerge from Russell Bryant
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
2007-01-23Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the ↵Joshua Colp
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
2007-01-20Merged revisions 51350 via svnmerge from Jason Parker
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
2007-01-20Merged revisions 51348 via svnmerge from Jason Parker
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
2007-01-16Add parkedcalltransfers option for res_features. This basically ↵Joshua Colp
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
2007-01-16Add support for G729 passthrough with Sigma Designs boards. (issue #8829 ↵Joshua Colp
reported by ywalther) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-16Fix a couple of typos in the sample osp.conf.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-16Patch allows for changing voicemail password in users.conf from voicemail ↵Matt O'Gorman
main, written by AnthonyL bug #8436 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-13Clarify what the trunkmaxsize value is in (bytes).Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-13Drop trunkrealloc option and just have the maximum size be a configurable ↵Joshua Colp
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
2007-01-13Merge in trunkrealloc option for chan_iax2. (issue #8267 reported by ↵Joshua Colp
marcodmb, branch by anthonyl) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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