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path: root/channels/misdn/chan_misdn_config.h
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2012-08-07Add named callgroups/pickupgroupsMatthew Jordan
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17Enhancements to connected line and redirecting work.Mark Michelson
From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-14Add outgoing_colp misdn.conf port parameter.Richard Mudgett
Select what to do with outgoing COLP information on this port. 0 - Send out COLP information unaltered. (default) 1 - Force COLP to restricted on all outgoing COLP information. 2 - Do not send COLP information. outgoing_colp=0 Also fixed sending the EctInform message so it always has the required redirectionNumber parameter when the status is active. JIRA ABE-1853 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-21Added CCBS/CCNR Party A support and enhanced COLP support.Richard Mudgett
This change adds the following features to chan_misdn: * CCBS/CCNR Party A support for PTMP and PTP modes. * Enhances COLP support for call diversion and explicit call transfer. These enhanced features require a modified version of mISDN. The latest modified mISDN v1.1.x based version is available at: http://svn.digium.com/svn/thirdparty/mISDN/trunk http://svn.digium.com/svn/thirdparty/mISDNuser/trunk Taged versions of the modified mISDN code are available under: http://svn.digium.com/svn/thirdparty/mISDN/tags http://svn.digium.com/svn/thirdparty/mISDNuser/tags Review: http://reviewboard.digium.com/r/218/ Merged from team/rmudgett/misdn_facility branch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-03This commit introduces COLP/CONP and Redirecting party information into ↵Mark Michelson
Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-23Removed trailing whitespace in chan_misdn files.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-15Fix up some doxygen issues.Jason Parker
(closes issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-11Doxygen updates, formatting. Olle Johansson
misdn stuff needs a lot of doxygenification (Hello, Qwell :-) ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89169 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89174 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-19Merged revisions 83023-83024 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83023 | crichter | 2007-09-19 11:31:55 +0200 (Mi, 19 Sep 2007) | 1 line added 'astdtmf' option to allow configuring the asterisk dtmf detector instead of the mISDN_dsp ones. also added the patch from irroot #10190, so that dtmf tones detected by the asterisk detector are passed outofband to asterisk, to make any use of dtmf tones at all. ........ r83024 | crichter | 2007-09-19 11:32:42 +0200 (Mi, 19 Sep 2007) | 1 line removed comment which violates the coding guidelines. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-16Don't reload a configuration file if nothing has changed.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-05added general Jitterbuffer Implementation. #9960Christian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@73298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-05Merged revisions 67334 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67334 | crichter | 2007-06-05 18:14:07 +0200 (Di, 05 Jun 2007) | 9 lines Merged revisions 67307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | 1 line briding is a bool, fixed copy and paste issue. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-05Merged revisions 67210 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r67210 | crichter | 2007-06-05 12:25:32 +0200 (Di, 05 Jun 2007) | 9 lines Merged revisions 67209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line added possibility to deactivate bridging per port ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18Merged revisions 59774 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines Merged revisions 59623-59624,59639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line we can now make 30 channels on a PRI (before we forgot chan 31..) ........ r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line don't be verbose if no need ........ r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18Merged revisions 59064 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59064 | crichter | 2007-03-20 14:16:06 +0100 (Di, 20 Mär 2007) | 21 lines Merged revisions 58849-58850,59062-59063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers ........ r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line fixed the crypt_keys stuff ........ r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line avoid sending a disconnect when we already received one. ........ r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line modified a loglevel ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-03Merged revisions 59804 via svnmerge from Nadi Sarrar
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, 03 Apr 2007) | 15 lines Merged revisions 59788,59803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines Use the new sysfs way of mISDN 1.2 to check if a port is NT or not. ........ r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines ptp is the 5th bit, not the 4th. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26Merged revisions 59202 via svnmerge from Nadi Sarrar
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it. * add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in' (the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-27Merged revisions 46351-46353 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ 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 ........ ................ r46352 | crichter | 2006-10-27 11:58:44 +0200 (Fr, 27 Okt 2006) | 1 line fixed not compile issue, which was just introduced ................ 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 ........ 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-11Merged revisions 44561 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ 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 ........ 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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08* first bits of decoding facility information elementsNadi Sarrar
* fail on misdn_cfg_init() if elements in the config enum don't match with the config structs in misdn_config.c * implemented first bits for encoding ISDN facility information elements via ASN.1 descriptions * using unnamed semaphore for syncing in misdn_thread * advanced fax detection: configurable detect timeout and context to jump into git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-03* removed pp_l2_check (fixed L2 bug in mISDNuser)Christian Richter
* added blocking flag to stack object. A port can be blocked/unblocked from the cli * added EVENT_PORT_ALARM to send alarm infos to the chan_misdn.c layer (later we can add a manager event for that) * added block_on_alarm option, to block the port whenever a ALARM occurs * added need_busy flag to indicate if we've sended a CONTROL_BUSY already * changed a bunch of cb_log(-1,..) to cb_log(0,..) due to funny behaviour in recent asterisk ast_log messages.. * fixed a few ETSI state violations, especially when finishing calls in different seldom states * changed debug levels a lot to make the log more readable in low debuglevels * some first fixes for the HOLD/RETRIEVE stuff (doesn't work totally still) * removed the PRECONNECTED state stuff * added cause 27 when we get a CLEANUP directly after a outgoing SETUP, this creates a CHANISUNAVAIL instead of a NOANSWER * removed the addr pointer from "misdn show stacks" that's not needed anymore and makes the output more unreadable * added cause saving on RELEASE/RELEASE_COMPLETE * set cause to 16 on prepare_bc * removed stack getting from ph_control functions, we don't really need it there * added beroec api git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-13added even more statefulness for sending out ↵Christian Richter
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-11* Introducing a new way for the l1watcher thread using the ast_sched way. ↵Christian Richter
Now l1watcher timeouts can be configured separately for every portgroup. * added a signal handler to allow waking up the misdn task thread (that may sleep in a poll call) via misdn_tasks_wakeup(). * overlap_dial functionality implemented. * fixes a bug which leads to a segfault after reordering config elements in the enum or struct git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-06* removed tone_indicate, we genrate only the dialtone by ourself (and the ↵Christian Richter
hanguptone of course) * 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
2006-07-03added misdn show config description[s] to show all the possible misdn.conf ↵Christian Richter
settings with a description in the CLI git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-29added better L2 handling for ptp, if it's down we don't try to call on that ↵Christian Richter
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-01added bearer capability reject support. we send release instead of ↵Christian Richter
disconnect in case we have no real channel yet. added support for Restarting channels added support for sending complete decoding. changed some log levels. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-23added a l1watcher timeout, therefore removed the old behaviour of guessing ↵Christian Richter
the l1state. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22added callcounters for incoming and outgoing callsChristian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-05Added option far_alerting. This option makes it possible to generate a ↵Christian Richter
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.. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-20removed dynamic switching from transparent to hdlc mode. Instead we've got a ↵Christian Richter
config option hdlc=yes now which enables the hdlc controller for a data call git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-03-09added option to change the connected party number dialplan (ton)Christian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15adde incoming_early_audio option, to avoid sending tone indications to the ↵Christian Richter
remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-15added pmp_l1_check option, to avoid l1 checking for group calls on PMP portsChristian Richter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-02-02* removed unnecessary struct elements and functionsChristian Richter
* 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
2005-12-09committed head of chan_misdn with a lot of new features. Most important of ↵Christian Richter
all: chan_misdn supports now the mISDN mqueue tree (smp,preemptible,gcc-4 aware\!). Additionally there are some code optimizations, new facility management (Calldeflect works for now). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-11-29remove extraneous svn:executable propertiesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2005-10-31finish chan_misdn commitKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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