similar to: check if not human

Anyone have much luck with these on ATA's? I have a few sites that use them succesfully with multi-port Audiocodes boxes, but just connected ten machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb switched network that is barely utilized, then out a T1 on a Sangoma card. Perhaps there is some tuning on the Linksys or the credit card machine itself? Going to look

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 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 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 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!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 ! 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

Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. I've been digging google, wikis but have not found what might be causing this. SIP works fine, this problem seems to be only zaptel specific. I could use the NVLineDetect application

Hello, We currently have a scenario where a large percentage of inbound calls on a telco PRI are intended for professors who aren't currently in their offices/at their desks. My question is, is anyone actively using the Asterisk "RLT" (Release Link Trunking) feature to bounce these sorts of calls back to the telco? The idea being to forward the call to their mobile phone without

Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of

Hi! I am trying to do a FastAGI server in windows. I am using the example from their page but I dont get anything. Anybody here has experienced with Fastagi in windows and perl that give a rigth direction to do this. I have experience with AGI but fastagi dont *-------------------------------------------------------* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL

I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox D110PG, T1, when a person calling from the PTSN will listen to them but then begins to distort the voice I heard that name. I probe the card in another computer and it works perfectly. Anyone has any idea or help. I'm going crazy with this problem. Install Debian on this server and the same thing happened to me. I bought

Hi! I have playing my icecast using Ice for Vorbis and ezstream for mp3 , of course , one by one. Now my question I want to have a few channels with different kind of music. How can I do this? Can I have a few mount point into ices or ezstream or can I run a few instance with different config files? I try this with ezstream but it doesnot work. Any cluees ? Edwin Quijada

Hi These are my config files ices and icecast. I am using ices 0.3 to play mp3. I have a few questions: 1-)There is a way to say ices to play a list from a database tio create a list into a table database. 2-)How can I play from a web page without open PLAY URL from winamp? TIA Edwin QUijada <p><p><p>>From: Geoff Shang <gshang@pacific.net.au> >Reply-To:

Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [s at from-pstn:1] Answer("DAHDI/7-1", "") in new stack -- Executing [s at from-pstn:2] Playback("DAHDI/7-1", "vm-intro") in new stack -- <DAHDI/7-1> Playing 'vm-intro' (language

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.

Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got