similar to: Credit Card processing machines

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT & T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the

Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer

Hi! I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. I need a lot of script done in perl and I wanna move to windows server. I checked Asterisk::fastagi but I see that everything is for Linux. Somebody has idea to do this in perl. I dont want to change the language. TIA *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase

HI all: Iam planning to use my asterisk box as callcenter?,any one can advice me with the best callcenter open source program based on asterisk . ? Any help will be apreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100307/116f1b75/attachment.htm

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

Hi!I have Samba 3.2.5 as PDC for 20 users with windws XP now I need that 5 users can write into C:\windows folder from each machine in my LAN. I have a Administrators group with RID 544 and i added these users to this group but it doesnt work, I did the same adding to Domain Admin and didnt . There is a something way to give to these users access to can write into this folder Thks.

Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

I've never used "reinvite" in systems I have installed to date, and I have finally run across a situation where it would be preferred. A remote office has a flaky Internet connection. With G729 encoding the calls to the central office over the 'net are tolerable. One Linksys 2102 drives two phones at this location, and when the first one calls the second one it travels to

a friend gave me an old apc smart-ups 1250 it has a rs232 interface but i have no pc with a rs232 port i attempted to use a rs232 to usb adapter but the ups goes nuts i connected the ups to batteries and verified it was working properly when i plug the usb cable into the pc the ups shuts down when i attempt to turn it on it beeps twice and shuts down this all occurs before i start the nut software

On 5/28/19 3:39 PM, Julian H. Stacey wrote: > grumpy at mailfence.com wrote: >> a friend gave me an old apc smart-ups 1250 >> it has a rs232 interface but i have no pc with a rs232 port >> i attempted to use a rs232 to usb adapter but the ups goes nuts >> i connected the ups to batteries and verified it was working properly >> when i plug the usb cable into the pc

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

I installed Board Master a software that control a CNC machine to produce PCB (Printed Circuit Board). The software install flawlessly but to check if it works I need RS232 working to control CNC machine. I tried to set connection parameters in Board Master choosing COM1 (the only available) but CNC machine don't move and software keep asking for switching machine on. Of course software and

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

In a few weeks, no SIP call is going to terminate unless they are signed properly, as mandated by law. We are in the business of Stir-Shaken, signing calls, as an FCC-approved provider. A big differentiator between our service and the rest: we are the only ones who don't need to receive the calls in our servers to sign them. We do this over a MySQL call, easily connectable to Asterisk via

Hi, How can I, from a single "driver" file, source other files in such a way that I can access their functions with parameters defined in my "driver" file? I wish to do this to avoid creating a single, self-contained but clunky piece of code. I have searched and found functions such as file(), pipe(), open(), found manuals on creating my own packages, etc. but I become

I've several USB <-> RS-232 dongles around. As well as a few embedded devices. They all "Just Work (tm)" on Redhat, CentOS, Fedora, Debian, Raspian and Kali. Knock on wood - never had a problem using any of them. As the drivers are part of the kernel, I'd expect any distro using a recent kernel to do well. On Wed, Jul 8, 2020 at 9:24 AM Leroy Tennison <leroy at