similar to: Automated callback with .call file

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

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

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

Hi, I'm trying to set up a call-back system using auto-dialout files. I want the call to be terminated when a specific timeout (defined in the .call file) is detected. Both parties should then be hangup. The problem is that the timeout is never detected... How to solve this? Thank you, Pierre .call file ---------- Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111 Callerid: 111111111

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]:

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

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

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

Hi, I am using asterisk's SMS functionality for sending messages. Most of the time it works without problems (as in situation 1) but sometimes something seems to go wrong with the transmission (as in situation 2). I am using "Deutsche Telekom", Germany's main TELCO, so I suppose the problem must be on my end. Can anybody tell me what is going on or how I could narrow down

On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by

Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call failure and

I can make calls over a SIP trunk as SIP/<trunk>/number I am trying to make calls over an extension thought using the same format SIP/4452/number - its not working. person says they can connect a software as extension 4452 and it works just fine. I have my register: register => 4452 at X.X.X.X/4452 [4452] type=friend username=4452 host=X.X.X.X allow=all dtmfmode=inband When I try to

Hello asterisk-users. I have the following dial-plan: [test] exten => 482,1,Dial(OH323/106@192.168.2.73,10) exten => 482,n,Dial(OH323/102@192.168.2.73,10) exten => 482,n,Dial(OH323/103@192.168.2.73,10) exten => 482,n,Dial(OH323/104@192.168.2.73,10) exten => 482,n,Dial(OH323/105@192.168.2.73,10) exten => 482,n,Dial(OH323/106@192.168.2.73,10) When I call exten =>

Hi all, I've been trying to get SMS operational on my Asterisk box, which has a TDM400P card with a pair of FXO interfaces configured (ZAP/1 & ZAP/2). I've not had luck with either of my lines, after issuing the command "smsq --motx-channel=ZAP/1/1709400X 00000 register". I see the following output in my Asterisk console: -- Attempting call on ZAP/1/17094009 for

Hi! I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt. I put here my callfile and that I get when asterisk begins to do the call If anybody has idea, pls. Tell me TIA ;;----CallFile----- Channel: Zap/g1/8093908270 Callerid: 8093908270 MaxRetries: 2 RetryTime: 300 WaitTime: 45

Hi all, I am trying to setup the SMS of Asterisk. I have a Siemens SMS enable fixed phone which connects to my Asterisk through PSTN. Currently, I can use my fixed phone to edit and send messages to the Asterisk. However, I cannot make my Asterisk to send messages to the fixed phone. The SMS command displays TX and RX records, hang for a while and then stops with non-zero exits. I read

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'm using VOIPJET to make international calls with an IAX2 connection between my local asterisk server and their server(s). At times I seem to have a problem if 5 or more international calls are made at once - I'm on a 1024kbps download and 256kbps upload DSL line (only the asterisk server uses this DSL line). Today I switched the codec from ulaw to ilbc in an attempt to lower

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its