similar to: vm-duration announcement missing?

Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files

I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8. I do not pass the 'n' option to any call to Queue() in my dialplan. Yet since I upgraded to 1.4.9, I have occasionally seen this on my console: -- Nobody picked up in 20000 ms -- Exiting on time-out cycle That log message "Exiting on time-out cycle" is exclusive to the logic in app_queue meant to

With pjsip (asterisk 13.14.1) I see the problem that an anonymous from header gets user=phone appendend to the URI if user_eq_phone=yes is specified: On the incoming leg: From: anonymous <sip:anonymous at anonymous.invalid:5060>;tag=Q5zBj7BMnvI6Fe6O2866fox3ZHmn-smt Get transformed to From: "Anonymous" <sip:anonymous at

Has anyone found a softphone that supports pulling it's configuration from a central server via TFTP/FTP/HTTP, much like hard desk phones use? I'm looking for something for a call center that I can provision from a central location by generating config files. If the phone has "soft keys" (yes, I know they're all soft - but you know what I mean; programmable buttons whose

The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok connector available to attach to a card that needs more power than the PCI bus can provide, like the TDM400P when FXS modules are used. HP has confirmed that there is no part they sell to give you such a connector, and Digium says their business edition folks got it to work, but only by doing nasty warranty-voiding things to the

One of my models has a column that is updated very frequently from a separate process, so it is important that when a record is saved in rails, this column should be left alone. In the update method of the controller I have: @record = MyModel.find(params[:id]) @record.update_attributes(params[:my_model]) params[:my_model] doesn''t have a reference to the column I''m talking

I am trying to compile the meetme2 application with the latest CVS head and it fails. Here is the error message that I get. Can someone point me in the right direction? gcc -pipe -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS

Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/XXXX|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

Dear all I am get the following problem when trying to compile app_meetme2 using mysql...it seems to want to use pgsql.....? anyone my Makefile looks like app_meetme2.o: app_meetme2.c #$(CC) -pipe $(CFLAGS) -c -o app_meetme2.o app_meetme2.c $(CC) -pipe -I/usr/local/include/mysql -L/usr/local/lib/mysql $(CFLAGS) -c -o app_meetme2.o app_meetme2.c app_meetme2.so: app_meetme2.o

Hi, I cannot compile Meetme2 on Suse 9.1 and Asterisk rc2. For latter 2 errors I guess I need development for mysql and postgres, but what about first error ? Regards, Robert. In file included from app_meetme2.c:13: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:300: error: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

Does anyone know what can cause queue members to go into a status of "Unknown"? pbxtel-01*CLI> queue show cs has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

Hello all. I'm setting up a new call center PBX using * v1.4, and figure it's better to go with AddQueueMember over AgentCallbackLogin. The functionality of AgentCallbackLogin still works, but without a firm idea of how long it will be in the codebase, I'm wary of building a system on top of it. The basic mechanics work, but I'm having some trouble on logging. With

I have a need to have a GUID (for example, bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the "-"]) generated in the dialplan. Is there any asterisk function that would do this ? I would prefer not to have to shell out every time a call comes in. Julian

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

Hello * users, I know that passing variable in the AGI script is by exten => _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being passed and simple_c_prgm is C code Now how will I receive these variables within C code ? Is it by the same way arguments are passed in command line to C by using argc and argv or there is more to be done than that? Thanks Regards -- Arpit Mehta

Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the

I'm building a dialplan for use with a bunch of GXP2000 desk sets. During testing, we had some user issues surrounding the lack of an on-phone dialplan. Users would hit 9 and sit there waiting for a redial tone, and the GXP would time out, sending just '9' to *, which couldn't do much other than spit back a 404 or play pbx-invalid. I turned on the "early dial" option

The Asterisk.org development team has announced the release of Zaptel versions 1.2.18 and 1.4.4. These releases are maintenance releases that fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. Both releases are available as a tarball as well as a patch against the previous release. They