similar to: autodial question

Hi, To force user to behave correctly, i want to make a process of disconnecting every member but one (special alarm phone) every day at a special time. I'm thinking of a cron job that will create a "call file" that will dial an appropriate extension to do the job, is this the correct way ? I'm facing two problems however, the call file system is here to connect a channel to

Hi all, i made a test scenario with two windoze machines: On the first one callgen323 is running in listening mode On the second one, callgen323 strarting 25 calls to the asterisk pbx, and the asterisk calls the first windoze machine. But after the second one make a few calls (mostly after 11 - 14) asterisk crashes with the only message : Segmentation fault. Are this to many calls for oh323

Hi, I want to disable function KEY of a BGT100, for examples: hold, transfer, conference, flash, ecc..... is it possible to do it? I'm searching a SIP phone that support SentText, some one can help me to found it? Thx a lot :) Igor -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello friends I've problem with my Centos 4.3. On start some information: 1. serwer HP DL 380 x Intel(R) Xeon(TM) CPU 3.40GHz 2. RAM 4GB 3. HDD storage FC 4. kernel 2.6.9-34.0.2.ELsmp #1 SMP This serwer is working with another one (that some configuration) in RHEL cluster (locking by DLM), as a file serwer with samba-3.0.22. Samba have about 100 simultaneus sessions. Problem is with load

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Hi guys. Does somebody knows how to get the concurrent calls from the dial plan? Or. How can i control not to run more than n simultaneus agi applications? Thanks in advance. rv -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140710/70b38e21/attachment.html>

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

I'm looking to build an application which requires a phone to autodial when the handset is lifted off of the hook. Since the system will be available to the general public, I want a solid phone that will withstand abuse, but not be too expensive. I'm leaning towards using a Sipura adapter with an analog outdoor handset. However, I don't nor ever have had a sipura adapter, so I'm

I'd like to setup an autodial that will do the following: 1. First try call my multiple SIP phones at my house - if no answer within n seconds, try my mobile phone. Alternatively call both SIP phones at home + mobile at the same time. 2. Once someone answer in step 1 (on any phone) initiate calling to a specific phone number/extension Currently I have it working except that in step 1 I can

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

Hello, I am using asterisk to autodial a phone number once an hour to verify an answering IVR script is working by listening for a beep played by the IVR, and that is working well. The one thing I have not been able to find is a way to trap if the call didn't connect at all, like if the entire IVR is down or the phone line is dead, since it doesnt enter the menu until the call is

Hello, I'm writing a small PHP application that generates calls automatically and tries to store call details on a Mysql Db, using manager API . When making an autodial call, I noticed that I couldn't read $DIALSTATUS values; since I can't evaluate dial status (BUSY, CONGESTION, NOANSWER), I can't understand when a receiver was busy or not. Nobody seems to have solved this

I have tried to get my TDM400P card to automatically dial a number or run an application when I pick up the phone without much luck. After reviewing the email archives, config files and source to chan_zap.c it appeared that all I had to do was set "immediate=yes" in the zapata.conf file and have a default number in the TDM400P's context (ex. s,1,Directory(default)). So far I have

Hello, Current moment, I've successfully put the incoming calls into Queues and dial to an idle agents. When the agents answer the calls, the agents can hear the pre-recorded message to incidate what's the service that the call is calling. But there one problem that I'm not able to make it having the Caller ID display on the X-Lite. Even I try to make a call direct transfer from

Hi All - I'm setting up a corporate emergency broadcast system that uses an autodialer to contact all company employees. Everything works fine except if the auto-dialed calls go to the end users' voicemail. If that happens, asterisk starts playback of the emergency message while the voicemail system on the other end is playing its outgoing message. The result is that the beginning (or

Hi! I have just install SUSE 9.1 on my computer that is on a network. First time I use samba I could view others computer on the network. The I try to share something so I configured samba server, and after that anithing works ok. When I try to go through the network via smb:/ I recieve a message saying that any workgroup is found. CAn anyone help me?

Hi, I am currently usind R to do a heckit maxlikehood model and I was wondering if there is anyway to do it but specifying the robustness of the std error. I would like it robust. I am currently working with: heckit(selection= ,outcome= , method "ml") Is there anithing else to type into this function to manage that? if not, Is there any other previous or later thing to do

Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time.

Hey, I'm still waiting for the new file server hardware to arrive so I plan to enable user logins in the meantime without profiles and home directories. Reason for this is quite some pressure from people. The setup on the 2 DCs should be done a day or two. Could there be problems if I configure profiles and home directories some weeks after users logged in for the first time? What if e.g.

Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G