similar to: call files

Hello, I'm trying to call different SIP-accounts to connect them to a conference. This is my call-file : Channel: SIP/test3&SIP/test1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: from-conf Extension: 1000 I get the following in the CLI : [Mar 22 14:40:26] -- Attempting call on SIP/test3&SIP/test1 for 1000 at from-conf:1 (Retry 1) [Mar 22 14:40:26] WARNING[29908]:

Hi everybody, I've got a very strange problem: As far as I remember I didn't change anything on my Asterisk side. I have 2 SIP providers to which I can place outbound calls. Today I noticed that outbound calls to provider "inode" fail (and inbound from this provider too). On the CLI I get every 20 seconds following messages: Nov 9 20:01:07 NOTICE[952]: chan_sip.c:5422

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI> originate IAX2/my-iax-provider/number2call application wait 10 [Apr 18 19:31:12] DEBUG[32331]: chan_iax2.c:4000 create_addr: prepending 40 to prefs -- Call accepted by 62.149.202.150 (format ilbc) -- Format for call is ilbc -- Hungup

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 =>

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

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

Hi, Using a 1.4 system in which dialplan is written using extensions.conf, I can use a custom .call file. On another system in which dialplan is written using extensions.ael, I can't use any custom .call file : system keeps replying : "apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file

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

Hi Just installed Asterisk on a Debian Woody/testing. I want to create a AGI script that is run after an outbound call is answered. I did this a while back (many versions ago). The problem is Asterisk does not seem to know the AGI application. I create a file test.call and place it in the outbound spool directory: the test.call file looks like this: #Simple test call script. #call my

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

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, 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

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

Folks, Am new to RoR and am building an example to get myself familiar. I am running into a simple issue while creating a user registration page. I have a User and Address models defined as below (partial/relevant code included below). User has_one address and Address belongs_to user. I have a foreign key defined in address table that refers to user(id) In a form I take in username, password,

I have set up extension.conf and sip.con with default parameter of UNIVOICE server, but Asterisk show this message when I call a number: Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899 create_addr: No such host: univoice,Ttr Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is

Hello! I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT) but only calling party can do forward. How to configure '*' to take this possibility to called party? ps both calling/called use sip --

As I got to compile 1.0.6 and got it to run but having the same problem as before I thought of creating a new mail thread about this instead of continuing with one where topic is about something else. (Sorry) So, I can't do register anymore. It worked just a couple of days before and I haven't done anything special as far as I remember. *CLI outputs: Feb 28 22:16:55 NOTICE[53475]:

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