similar to: CDR_DISPOSITION_FAILED - Call has been answered correctly

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Would it be sufficiant if I were to copy the makefile and res_config_ldap.c to the res/ directory of my running Asterisk and do make; make install? or do I have to do LIBS=-lldap export LIBS ./configure before that? My asterisk version is 1.2.6. Thanks in advance, Abhishek * * * * On 8/27/07, Gavin Henry <gavin.henry at gmail.com> wrote: > > I see it is res_config_ldap. You'd be

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

Hi, I'm one of those people who still need to maintain * 1.2 systems and cannot easily upgrade. :-( My 1.2 systems were very stable until I upgraded from 1.2.37 to 1.2.40. I have made some changes within my dialplan but nothing unusual. Today I've had a crash: https://issues.asterisk.org/file_download.php?file_id=25571&type=bug Yesterday I had another:

hi,all for a long time, i cant understand the difference between ${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)} i know ${CDR(start)} mean when a call is start. and ${CDR(answer)} means when a call was pick up. but what's ${CDR(calldate)} mean? Could you help me ? Thansk a lot! -- Thanks for your supporting, have a nice day. Sucan

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

I've recently finished an add-on module for CEL logging to MySQL, and it needs to be tested. The feature is being tracked at https://issues.asterisk.org/view.php?id=19058 And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110&type=bug Thank You, -Jonathan Penny -------------- next part -------------- An HTML attachment was scrubbed... URL:

this purpose. I have been reading the voip-info pages and have set up AJAM and can get results from doing http requests to the asterisk server, however, this is in the form of an action, such as login, rather than subscribing to an event. I have been looking here for information http://www.voip-info.org/wiki/view/asterisk+manager+events Does anyone know if subscriptions is possible with AJAM?

Hi all I'm deploying a VoiceMailserver with Asterisk behind a legacy pbx, providing Voicemail to email services for Lecagy PBX extensions. On busy or unanswered calls, Legacy pbx will dial a specific DID (one per extension) to asterisk, and the call is handled by Voicemail application. I've several SIP extensions on this Asterisk box, and calls between Asterisk extensions and legacy PBX

Hello I use xen and I am trying to use the parallel port of Centos 5.2 host from a Windows XP (HVM) guest. I have tried to transfer control of the port by blacklisting lp, parport and parport_pc + adding: ioports = [ "0378-037a" ] to the xen configuration file. However, in the presence of this line, xm create replies with: Error: function takes exactly 4 arguments (3

I wrote a connection pooling patch because asterisk is not usable with MSSQL without it. If you're using, or would like to use, MSSQL I recommend you to check it out. http://bugs.digium.com/file_download.php?file_id=8809&type=bug Just so you know, this is a diff against 1.2.1 and it's been only tested with 1.2.1 (because that's what we are using in production). To enable pooling,

Hello mailing list, I have been porting one of my Asterisk boxes to 1.4 and I have encountered a nasty DTMF problem. What happens is someone might come in to my IVR and enter "12345" and what will actually come through could be along the lines of "12234445". Sometimes it works, sometimes it doesn't. I had this problem with 1.2 back in November but was able to solve it

Hi folks, I repeat "as is" the title of a post someone did a few months ago, since I am facing the same problem and did not see one single answer to his post. Maybe I'll be a little bit more lucky. When I'm trying to issue an Atxfer AMI command, in the asterisk 1.8 branch, what happens is that some DTMF's are sent, like this : [Mar 22 15:46:27] DTMF[5910]: channel.c:3900

Dears, I need to save some information on userfield when calls end in Asterisk13, but I have two error cases: 1. With endbeforehexten=no in cdr.conf, I have a registry in cdr, but userfield is not set. 2. With endbeforehexten=yes, I have two lines in cdr, one with duration, src e dst correct, and a second line with userfield setting and dst h. I am using cdr_odbc.conf, with Asterisk11.14.0 it

I''ve got some problems right now using the new model for relationships in fixtures (by label, not by id) and STI. What I''ve got, is a couple of models: Attachment --> FileDownload Version Version has_one file_download with file_download_id on the versions table In my fixtures, I have two FileDownloads in my attachments.yml like so: hemingway_alpha_zip: size: 100

Hi. i am running asterisk 11.5.1 in my system (debian squeeze) and i do get the CDRs through the csv file, that asterisk creates. i would like to have the CDRs in a nice web based tool and after some search i have found http://acdr.com.au/ i do have it installed with all the dependencies (apache, php, mysql), but i do not know how to interface the CDRs. has anyone used this tool or any other

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

Hello List, I have been trying to setup T38 gatewaying with the following setup SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo Canceller: HWEC I'm aware there's no support for T38 gateway but I have been trying to get the patches https://issues.asterisk.org/view.php?id=13405 to work. It seems

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi, Is there a way to enable the usage of UNIQUEID CDR field using a MySQL database backend for storing CDRs without having to recompile asterisk-addons as stated here http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? After version 1.4 it is said in release that it can be done (not sure if it applies to mysql backend) How would it be the syntax in cdr.conf? I tried this without success in