similar to: Achoring MEdia

Hi, How do I display/log the results of variables from extenstions.conf? I've several macros, where I'd like to use ${CONTEXT} to help in GotoIf's, but I'm not convinced what value CONTEXT is being set to. I tried using NoOp(${CONTEXT}) and then set debug on for messages, but all I see is: Dec 7 16:09:25 VERBOSE[815] logger.c: -- Executing

I am about to setup an asterisk pbx for 2 customers and would like to use a single box. Since I don't want to use 1xx for one and 2xx for the other customer, is there a way to setup overlapping extenstions? If not anybody can give me an example of how a similar situation like this was solved? thanks micko

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

Hello, We are having a problem with linker lookup on Windows. When the driver tries to locate a program by its name, e.g. a linker, it scans the paths provided by the toolchain using ScanDirForExecutable and then, if fails, uses llvm::sys::findProgramByName, which on Windows searches the CWD and system path. ScanDirForExecutable disregards file extensions, so search in the toolchain-provided

I'm trying to setup my first Asterisk setup on a CentOS 5 installation on VMWare Workstation 6. Got two Linksys SPA941s working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they "connect" but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the

Hello, I've been writing a program in C that will be called by R. I seem to have stumbled upon an odd error that seems to suggest there is a limit on the number of times "Realloc" (the R version as defined in the manual "R-extenstions" not the C version "realloc") when I try to use the following program: #include <R.h> #include <Rinternals.h> SEXP

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my

I have been working on a project that involves a lot of front-end functionality and I noticed that Position.cumulativeOffset was not as accurate as I had expected/needed. I also needed some more options and I just didn''t really like having to type Position.cumulativeOffset all the time. I wanted to simply do element.getOffset() and be done. After lots of time spent going back and forth

Hi, I would like to replace the last tree characters of the values of a certain column in a dataframe. This replacement should only take place if the last three characters correspond to the value "/:/" and they should be replaced with ""(blank) I cannot perform a simple gsub because the characters /:/ might also be present somewhere else in the string values and then they

From: Joerg Roedel <jroedel at suse.de> Do not allocate a vmcb_control_area and a vmcb_save_area on the stack, as these structures will become larger with future extenstions of SVM and thus the svm_set_nested_state() function will become a too large stack frame. Signed-off-by: Joerg Roedel <jroedel at suse.de> --- arch/x86/kvm/svm/nested.c | 47

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

What I want to do is have one phone number for multiple call bridges (meetme) so that first users are prompted for their call bridge ID then their password. exten => 7001,1,Playback(/var/lib/asterisk/sounds/if-you-know-the-extension-you want-to-dial-press-that-extension:gsm) exten => 7001,2,set $foo to whatever was entered on the phones keypad exten => 7001,3,Dial($foo,60) Thanks!

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hey everyone, Currently, I've got a PBX that is emailing me on call failures to an international SIP provider of ours. I'm doing this with exten => 1,1,System(mail -s "Call from ${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}" nick at flhsi.com < /dev/null) This works fine, However it's a little lacking. For Instance, Our INTL SIP

Hello, I wish to play a recorded announcement on reply to 4xx 5xx 6xx messages . According to the status a audio file would be played from asterisk server via ser to the caller How can I configure a such feature ? My configuration: Ser act as an outbound sip proxy . Asterisk a sip media server and registrar. sip agents ---- SER -----Asterisk Harry

My goal is to have only 1 primary phone number that can seamlessly "pool" multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to "forward" or "roll over" the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

Hi, I had someone who can see help install some games in Wine. The games work great except for shades of Doom. It starts correctly, but when the game should begin, it freezes. It requires DirectX8 to work. It actually did work once for a few minutes, then, it froze. Will someone good with Wine Please install Shades of Doom and help figure out what's going wrong? I am using Ubuntu 9.04 with