similar to: Outgoing call immediately disconnected

Helo WipeOut, I have found a solution for sending dtmf after dial. I use spooling. Take a look at the sample.call file inside asterisk dir. You need to edit this file and dump it in /var/spool/asterisk/outgoing. Asterisk will precess this file automaticlly I create the sample.call do something like this: Channel: OH323/4324324324 #dial the access way MaxRetries: 3 RetryTime: 60 WaitTime: 30

I managed to sort out my earlier query regarding flash times (changed delay in zapata.conf) Now, I am getting a timeout after the Flash command in an outgoing call-file based call: -- Attempting call on Zap/1/108 for 567112@demo:4 (Retry 1) > Channel Zap/1-1 was answered. -- Executing Festival("Zap/1-1", "Dialling now") in new stack == Parsing

asterisk-16.8.0 Hi I've set up a callback script to retry a number if it's busy, but as I watch the console output asterisk seems to rush 3 or 4 calls at once before waiting the RetryTime of 20 seconds that I've set. The script: -----8<------ CALLERID=$1 EXTENSION=$2 TEMP=`mktemp /tmp/call-XXXXXX`.call cat <<EOF > $TEMP Channel: IAX2/account at

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

Hello everyone I'm trying to make asterisk get a call out using the .call system. The setup is A@H 2.6 This is the content of the file is : <<< Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1 >>> I'm

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

I've seen other posts about this problem, but I haven't found a solution. I'm dumping eight call files into the "outgoing" directory at one time. Three of the calls are successful while the other five are lost. Here is the call file: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Note: All

I have created a call file which has been moved into the outgoing directory. However the log file displays the following message: Unable to open /var/spool/asterisk/outgoing/1.call: Permission denied, deleting I have executed chmod 777 1.call on the file prior to moving it to the outgoing directory but is there something else I need to do before the file can be used by Asterisk? Any help

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 have a PSTN termination provider "foo" which will accept standard U.S. calls in the form 1<10 digit ph#>. I have an outbound route named "foo", with dial pattern "5|.", with the only entry in trunk sequence being "IAX2/foo". I have an X-lite local extension, on which I can dial 51<10 digit ph#>, and asterisk will call out over foo and the

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

Just a shot in the dark... but have you tried Answer() before Playback()? Josh McAllister -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Engleward Sent: Monday, May 01, 2006 11:43 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] /var/spool/asterisk/outgoing/ prematurely hangingup I have

I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I have just about everything working except for the message waiting indicator. I have the following setup in context [ccm] in my extensions.conf file: ;MWI exten => _2807XXX,1,SetCallerID(${EXTEN:3}) exten => _2807XXX,2,Dial(SIP/28888@65.202.115.240) exten => _2807XXX,3,Answer exten => _2807XXX,4,Wait,1

hi! I have a question of ISDN PRI "calling number" setting for outgoing call. I need the receiver to see the CLI that I have set. Eventough I've set the CLI as below, the receiver keep on getting a fixed number( used with the PRI to receive incoming calls). My ISDN PRI E1 provider says he has not done any restricions on the custom CLI side.

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

I have a voicemail application that users can listen to messages and leave messages. I am looking for a way to play a beep tone to a user when a new message is received when they are on the phone. Here is what I have come up with: in extensions.conf: [beepvoicemail] exten => 1000,1,answer() exten => 1000,2,NoCDR() exten => 1000,3,wait(2) exten => 1000,4,Set(TIMEOUT(absolute)=5)

Like some other people on here, I am trying to integrate Asterisk for VM with CCM version 3.x. I've got gnugk and Asterisk running, I've got CCM registering with the GK, I've got the voicemail pilot and profiles setup. A call comes into a CCM phone, it rings, rolls to the correct VM on ASterisk and asterisk emails the voicemail and I can check the voicemail, but I cannot get MWI

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

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will