similar to: Fedora Core 3 or Fedora Core 4? yum update ornot?

Hi all, One of the areas I am trying to research before I can confidently start deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines / any similar) I'm talking about the credit card swipe boxes at point of sale desks. I believe they dial out to the specific bank provider everytime a card is swiped. My question is: - Does anyone have any experience using

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real "English" asterisk prompts. The only one I have found is here >> http://www.g7ltt.com/VoIP/vmfiles.html <http://www.g7ltt.com/VoIP/vmfiles.html> And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc)

Hello, Does anyone know how to conference a call on the SNOM 200 phone? Whenever I push the cnf/tran button it just hangs up on the active call. The manual says you have to push the cnf function key but it doesn't appear in the lcd on my phone. Thanks -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have had quite a few customer complaints about attended transfer cutting off callers. The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. I have checked the scripts I don't *think* this is a dial plan error but if anyone has this working correctly on Asterisk

I am having trouble using wildcards ("*") with yum in Centos 5. Listing or installing packages using them just immeidately fails with a "No match" error, and I'm not sure why. Here is an example. Any comments? I'd love to fix this. [root at granta ~]# ps -e | grep yum 2706 ? 00:00:22 yum-updatesd [root at granta ~]# yum list | grep kern kernel.x86_64

Mike Burger wrote: > On 2018-02-16 9:29 am, hw wrote: >> Mike Burger wrote: >>> On 2018-02-16 8:16 am, hw wrote: >>>> William Warren wrote: >>>>> I would just buy a cloudkey and not have to bother installing the software >>>>> onto your machine directly.? If you do not have a power over ethernet >>>>> switch you'll need

>>> 1) SYSLINUX on EFI must be extended to load arbitrary >>> EFI binaries >>> (perhaps through a module first >>> like how linux.c32's code became the >>> Linux loader). >> >> Do you know 6.03 status on this topic? >> i.e. can something like bootmgfw.efi be booted today ornot? > http://bugzilla.syslinux.org/show_bug.cgi?id=17

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Mike Burger wrote: > On 2018-02-16 8:16 am, hw wrote: >> William Warren wrote: >>> I would just buy a cloudkey and not have to bother installing the software >>> onto your machine directly.? If you do not have a power over ethernet >>> switch you'll need a micro USB cable and power supply adapter to run it but >>> after that it takes care of running

Hi everyone, I wrote a set of Mozilla Ubiquity commands that allow the user to search on rdocs, also featuring autocomplete, load of any rdoc hosted on the web and changing the default rdoc when searching. The project is hosted here: http://projects.talleye.com/ubiquity-rdoc and GitHub. Please, let me know any bug, suggestion, etc Luis Cipriani WebCo Internet brasigo.com.br

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Hi, I think this is one area that needs to be developed. I am curently implementing a system for my home so cannot really justify the cost of financially supporting the development of this when all I really need to do is buy a telephone extension lead for my existing fax modem!!! I am more than willing to devote some testing/documentation time (I am not really a programmer) if that helps.

Word. I'm doing a dupe of my production server this week as a CYA. Guess what: FC2. Once I yum update to the current kernel, no more yum. There's no reason to. You may have your own reasons (publicly avaliable server, for example) but why add uncertainty to an, at best, quite uncertain process (that of creating a stable Asterisk install given random hardware, network conditions, PSTN

Hi Guys, I have tested and isntalled Asterisk 1.6.2 with FreePBX from Digium repos based on this url: http://www.asterisk.org/downloads/yum BUT that doesn't seem to work with Fedora instance which I am running on Amazon Ec2. Apparently Asterisk 1.4 is natively included in Fedora repository but not Asterisk 1.6. And when I added the Digium repository, it give me a 404 not found. I check and

William Warren wrote: > I would just buy a cloudkey and not have to bother installing the software > onto your machine directly. If you do not have a power over ethernet > switch you'll need a micro USB cable and power supply adapter to run it but > after that it takes care of running your software for you you don't have to > install anything on to your machine One would

Hey, maybe lynx package should be a requirement of the whois package...? ? $ whois ubiquity.io ? [Querying http://www.io.io/cgi-bin/whois] ? [HTTP: Unable to run web browser: /usr/bin/lynx: No such file or directory] Thx, JD