similar to: Can't run AGI for outbound call

Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm

Hi I've just started to play around with AGI scripts and have run into problems. When I run Asterisk in console mode everything works just fine. If I run Asterisk in 'regular' mode (not console) it crashes if I hang up on the script. I have used Python scripts to test this and also the "agi-test.agi" script. (the Asterisk code was compiled from CVS code just a few days

I am using the * auto-dial out feature but don't want to have to specify a channel (Zap/G2/) to connect to the extension. Current file I use: Channel: Zap/G2/12127778866 #<< ==== I have to specify a specific channel MaxRetries: 1 RetryTime: 60 WaitTime: 30 # # Assuming that your outgoing call logic is kept in the # context called [line1out] # Context: line1out Extension: 7632

I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called - bad! What I would like to

hello list, i have created a callfiles with my asterisk 1.4.43 like: Channel: SIP/watara/06xxxxxxxx MaxRetries: 10 RetryTime: 5 WaitTime: 20 Context: mycontext Extension: s Priority: 1 extensions.conf mycontext exten => s,1,Ringing() exten => s,n,Playback(hello-world) exten => s,n,Dial(SIP/105) exten => s,n,Hangup() it works with one number how can i do in order to create a

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file

Hi, all, I met a problem for several days, any suggestion is really appreciated!!! I'd like to do autodial using Asterisk. For example, I have a file under /var/spool/asterisk/outgoing, which include: channel: zap/g1/12345 MaxRetries: 0 RetryTime: 60 WaitTime: 20 Context: default Extension: 2222 Priority: 1 And in my "extensions.conf" file, I have [default] exten =>

If you're using perl on RedHat 9 make sure you put this command somewhere in your boot scheme: export LANG=C or at least execute it before running perl scripts. Redhat has EVERYTHING set to LANG=UTF-8 and it screws up all sorts of perl stuff, and several other pre-written programs in other languages too MATT--- -----Original Message----- From: Ray Burkholder

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid

When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop "properly" into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where

Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

Hello everyone. I'm working on an application that needs to automatically send faxes. To send the faxes I create .call files but the .call files mostly fail because my lines are always congested within business hours! Is there any trick I can use to give the end user a better chance at actually receiving the faxes? I already tried using the local channel for dialing (so I can put in

Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450 at ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In

Hi there, Is there a way to program asterisk to dial an extension Monday to Friday at a specific time and then read a specific string? eg: "Kids, go to the bus stop now, you're about to miss the bus!" Many thanks, Pierre

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various

Has anyone made, or have any simple PHP, or HTML interfaces where by a user could enter their number and the number they want to call, and have asterisk bridge the calls?

Ok, I've been running applications on 1.4 for quite some time using meetme to hold a person, while the person on the other end of the call accepts, etc. I was playing status messages to the calling party using a context like this: [status-one-en] exten => 100,1,Playback(my_status_message) exten => 100,1,Hangup() and then creating a call file like this: Channel: Local/100 at

I'm making outbound calls by placing call files in the asterisk outgoing directory. At times, the call would be hung by SS7 without even attempting (due to error in the outgoing number). I get the following on console: -- Attempting call on ss7/9297210213 for s at croom:1 (Retry 1) -- Sent IAM CIC=22 ANI=9134904821 DNI=9297210213 RNI= -- SS7 hangup 'SS7/callserver/22'