similar to: answer a voip call, play info.

Hi all - Searching for java agi in the mailing list archives turns up ancient posts. Anyone else using java for their AGI? How well is it working & what are you using? My script is pretty simple, and I could write it with perl easy enough, but I just would feel better if I can keep most programming code for our system in java. Thank you- Matt

Hi all -- I'm having awesome fun with Asterisk & voicepulse connect together. So cool. I'm trying to have the caller id read back to me. Do I need to do something to have this sent across in the sip.conf? Or is there something I need to do somewhere to enable the reading of this data? Thank you! Matt Here is my extensions.conf exten => _XX.,1,Answer() exten

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all - We are using voicepulse connect & asterisk together. We'd like to record our own outgoing messages, to be played back people that will be dialing into our voicepulse connect supplied DID. (am I getting this lingo right?) What is the best audio format to ensure the highest level of audio quality? Thank you - Matt Pease Parking Hero

Deal list > I want to generate a random number in an interval using exponential low > I know how to make that simply using uniform low > x1 <- runif(1, 5.0, 7.5) > this will generate a number between 5 et 7.5 but with uniform low. > Could you help pease.? > Regards > > -- > PhD candidate in Computer Science > Address > 3 avenue lamine, cité ezzahra, Sousse 4000

Hi All I have to run multiple stimations and to compute Likelihhod ratio. If I compute ls function with coef and summary I can extract "outputs" that I need. I am not able to find something similar to log liklihood.... Can you pease tell me running a ls function x on y how to extract if posible LR statitic or Likelihood or Log likelihood. Many thanks in advance. If you send me

Can someone pease tell me what I am doing wrong? Im trying to register an external function with the JIT, so can call functions in my music application. Here my function int Execute(llvm::Module *Mod, char * const *envp) { llvm::InitializeNativeTarget(); std::string Error; llvm::OwningPtr<llvm::ExecutionEngine> EE( llvm::ExecutionEngine::createJIT(Mod, &Error)); if (!EE) {

gafferuk <gafferuk at gmail.com> writes: > Can someone pease tell me what I am doing wrong? It is hard to say because you don't say what the problem is, but let's try. > Im trying to register an external function with the JIT, so can call > functions in my music > > application. > > Here my function > > int Execute(llvm::Module *Mod, char * const *envp)

hi all, Iam using the following CBQs to limit the FTP bandwidth pease tell what changes are rquired to make it work.... tc qdisc del dev eth0 root tc qdisc add dev eth0 root handle 1:0 cbq allot 1514 avpkt 1000 bandwidth 100Mbit tc class add dev eth0 parent 1:0 classid 1:1 cbq allot 1514 bandwidth 100Mbit rate .4Mbit weight .4 prio 2 maxburst 20 cell 8 avpkt 1000 tc class add dev eth0

I have a similar situation here, I want to include TTS in my asterisk IVR system. Could someone give suggestion(s) please, I prefer open-source thanks in advance! Chatila, A. C. P. O. Box 365, Kihesa Street, Njombe, Tanzania. *Mob:* +255 765 154 235 *Whatsapp:* +255 653 258 608 *Website:* chax.me.tz On Thu, Aug 27, 2015 at 9:07 PM, Steve Edwards <asterisk.org at sedwards.com> wrote:

Hi all, The documentation for SWAT indicates that logging in as the "root" user will allow the user to update & change Samba parameters. The question I have is, does SWAT define "root" on a Solaris 8 system as "a user with user id 0", or "a user with the user name root"? The reason I'm asking is, my system administrators are (rightfully)

On 07/05/2015 09:17 AM, linush at verizon.net wrote: > Someone please tell me what I did to screw this thing up so badly. On 07/05/15, Gordon Messmer<gordon.messmer at gmail.com> wrote: Have you looked at the log files in /mnt/sysimage/root/? ------------- Quoting broken in this mailer ------------ So I looked in /mnt/sysimage/var/log/anaconda and found this in anaconda.packaging.log:

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> Hello,<br> <br> I have an * installation that sometimes receives&nbsp; a 302 "Moved Temporarily" response to an INVITE. * sends the ACK but takes about 30 seconds to start the new INVITE

Hello I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

I have a VPS account with 256MB guaranteed RAM. I am running it as Apache1.3 -> Pen -> mongrel_cluster (with 2 mongrel servers). "top" shows 2 "ruby" instances (I guess for both of the mongrel servers), each weighing in at about 25MB. Does that sound about right? If you are using 2 mongrel servers per client app, does that mean that each RoR app needs 50MB+

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

I have an application setup as follows: Core code is in: /home/subdomain1/current accessed at subdomain1.domain.com I then have various instances of this app that are sym-linked to the core (minus a custom config file and views) setup like: /home/subdomain2/current accessed at subdomain2.domain.com I have backgrounDRb running to update a cache of RSS files. Is there a way to get all

I use samba 4.1 as dc with acl. I have user with uid 3000023. However, I don't have group with guid 3000023. However, when this user creates a folder samba in acl list creates permissions for group 3000023 and as result I have broken link. Rowland Penny (thanks to him) said that I could see the type: ID_TYPE_BOTH setting in /usr/local/samba/private/idmap.ldb. As I understood I must change

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files