similar to: Resolving groupcalls

Hi,I've been trying to code through CommandLine the options I want my tool accepts, but I find quite impossible to achieve robustly what I need .Look, I want the tool accepts a list of arguments in a particular order. For this goal, I know the cl::Positional flag. But, the problem is that the first argument must be one of a set of options (like a kind of subcommand of the tool). In my case,

Does anyone have a Dial script or a hint on how I can dial 10000 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me.

Hi, I've been trying to code through CommandLine the options I want my tool accepts, but I find find quite impossible to achieve robustly what I need. Look, I want the tool accepts a list of arguments in a particular order. For this goal, I know the cl::Positional flag. But, the problem is that the first argument must be one of a set of options. In my case, only the three next commands are

hi all after getting some (or - a lot) of messages from nagios, claiming asterisk to be down, I found this out... astping xxx times repeatedly, and the manager fails to start. this little script was used for testing. below, I've pasted the output from 'ps axfv' before and after the DoS, showing asterisk having allocated ~2GB RAM. roy #!/usr/bin/perl -w use strict; my $i = 0;

Hi All, I am trying to get Asterisk up and running on my new Mandrake 9.1 install. I've installed Linux in the "standard" mandrake security mode, and "su" to do my attempts at install. I managed to obtain the source from CVS, and have been able to compile Zaptel. I then ran insmod zaptel, and also make config. I think I have compiled and loaded Zaptel successfully as

Hi Daniel, Just in the moment you replied my message, I was rewriting it as I hadn't noticed it hadn't a correct format. Sorry for that and thanks for answering it anyway. > ./prog <option1> | ( <option2> --arg1 --arg2) | ( <option3> --arg1 ) > Yes, that is exactly what I need. It's a pity commandLine doesn't implement that possibility. So I will do what

Hi Daniel, I would like to go deeper with CommandLine and I was asking if you could help me again. Look, following the same example you put in the last message: ./prog <option1> | ( <option2> --arg1 --arg2) | ( <option3> --arg1 ) What I really really want is the same except I don't want the "--" prefix is present in any of the arguments. ./prog <option1>

On 10 May 2013 09:44, Pedro Delgado Perez <pedro.delgadoperez at mail.uca.es> wrote: > Hi Daniel, > > I would like to go deeper with CommandLine and I was asking if you could > help me again. > > Look, following the same example you put in the last message: > > ./prog <option1> | ( <option2> --arg1 --arg2) | ( <option3> --arg1 ) > > What I

Hi Daniel, Thanks for your answer. > I am considering fixing the bug in the LLVM CommandLine library but is > going to be a long time before I can look at it. So don't wait for me > to do that. > Ok, ok, but, if you end up fixing this, please keep me post. > If you are really desperate to have the command line options in the > way you want they you may need to implement

Hi, folks I wroted the codes below. I have the trouble with the line start from "Dir.chdir". When I run this code, the FTP of photos is successfuly done. But, the display of browser is white out. When I comment out the line start from "Dir.chdir", the trouble doesn''t happen. Is there anything wrong with my usage of "Dir.chdir"? def ftp_photo

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

I have a pretty strange problem. I have about 100 DID's that come down a PRI from SBC in the United States. On Friday afternoon, one of my DID's flipped out. When you call it, the SBC operator comes on and says that the line has been disconnected. I contacted them and they ran test and they are telling me the problem has to be on my end. My problem is that the CLI never shows the

Hi Friends, I am new to linux and new to asterisk. I need some help setting up asterisk in my linux box. Does anyone have a step by step guide ? On my PC i have installed a phonejack (from Quicknet) as well. Your help is appreciated.. I kind lost.. thanks, _________________________________________________________________ Expand your wine savvy — and get some great new recipes — at MSN Wine.

I've looked all around, and I can't find an answer to this. I apologize if this has been discussed already or is buried somewhere in voip-info.org. I have an asterisk setup on linux 2.6.11.11 kernel, a revision E/F TDM400P, and Polycom IP501 phones. As soon as I load the zaptel module into the kernel, the voice prompts and voicemail system ceases to work. The asterisk logs

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

I have a number of Polycom phones to setup with my * server. For my initial few phones I hand wrote configs. Does anyone here who uses Polycom phones have some form of management utility for automating their setup? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

On Tue, Sep 6, 2016 at 1:55 AM, Olivier <oza.4h07 at gmail.com> wrote: > Hello, > > After testing "pkill -SEGV -f /usr/sbin/asterisk" on Debian Jessie > platform, I've got several questions : > > > 1. When I issue a "cd /tmp; asterisk -cvvvvvvvvvvvg -U asterisk -G > asterisk" command, and then issue a "pkill -SEGV asterisk" command,

Hello, I'm looking for some advice on securing Asterisk. Recently my servers been under several brute-force SIP attacks. I have several remote sites, as well as many roaming users, who may have PC softclients and/or SIP based hardphones. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone

I have to agree. A large number of people are looking for this feature. I have written a web script that can show Agent logged into the system. I think integration/gateway between Asterisk and Jabber would be a amazingly wonderful product. There is always MSN. Shad Mortazavi --------------------------------------------------- Nexus Technical Manager n|m Nexus Management Inc Netural Bay

Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a "reload app_queue.so" all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -------------- next part