similar to: system() app changed drastically! How do I use it now?

Try the following: exten => s,1,Answer exten => s,2,Wait(1) exten => s,3,Read(PIN,87) exten => s,4,System(/usr/local/bin/pin -c ${PIN}) ; check it exten => s,5,System(/usr/local/bin/pin -d ${PIN}) ; delete it Exten => s,6,GotoIf($[${SYSTEMSTATUS} = FAILURE]?105:7) exten => s,7,SetAccount(${PIN}) exten => s,8,Newt,pinout-config ; connect them exten =>

It would be prudent the test for success and continue rather than failure and drop. For example: exten => s,5,GotoIf($["${SYSTEMSTATUS}" != "SUCCESS"]?105:6) That way only the result that you know is good, Will continue a call.. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On

Hi all, I'm running into a easily replicated problem at the moment, with Asterisk 1.6.0.28 (built from source, no special configure parameters, other than a path) running on top of a fully up-to-date CentOS 5.5, and I'm looking for suggestions as to why this is occurring. I've spent some time looking into the issue, and really haven't been able to come up with much. We have the

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as

Hi, I created a script named example2.sh which goal is read some text from my HP Service Desk using an application in java and send this text to the text2wave application for TTS. example2.sh java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-sd.wav When I execute the script in prompt, everything is ok, but when I use the system() command in my extensions.conf it isn?t

All, Could someone please help me resolve this: [admx:test] $ ls ERR01 ah01 ah02 an01 an02 mp01 mp02 [admx:test] $ ls {an,mp,ERR}* ERR01 an01 an02 mp01 mp02 I want to rsync only the "{an,mp,ERR}*" files across using the following command but do not see the expected results. [admx:test] $ rsync -va --exclude="*" --include="{an,mp,ERR}*" ./*

Hi. I've spent the past few hours trying to get SSL working right in Dovecot 1.0.5 and now I must turn to you for help. I purchased an SSL certificate from Go Daddy. I pointed ssl_cert_file to the .crt file and ssl_key_file to the .key file, but the client (Mail.app) complains: Mail was unable to verify the identity of this server, which has a certificate issued to

Hi all, we''re just in the process of upgrading to the new asset pipeline, but I''m having an annoying issue with Sprockets. It turns out that I can''t enable both *debug* and *digest* in development mode. When I do, I get a Errno::ENAMETOOLONG because the digest becomes the *entire* *content* of the file instead of a hex. I get this with both sass and javascript.

Hi all. I've been trying to build some new systems, and no matter what I do, if I load the zaptel and tor2 drivers, the system panics within an hour, even with no traffic. These systems are using dual Athlon MP 2800 chips with one, two, or three T400P boards and 2 GB of system memory. I'm currently using Fedora Core 1, but I also went back to our old reliable Red Hat 7.3 and the systems

Does * support ISDN Primary Rate Interface (PRI) - 2B Transfer Capability for T-1/PRI? In other words the ability to take a call and join it to another call and then drop off letting the CO-switch take over. -Kevin Kevin Fjelsted, President AltiCom CTI, Inc. Track Me Down! One number Access, Press 11# during the voice mail message greeting to have me F-O-U-N-D! Phone: 612.259.0722 Fax:

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 been testing both T400P and TE405P boards and I'm running into some kind of hard limit on the number of simultaneous calls. This is on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1. Everything is fine up to 190 channels, but the 191st call fails every time with errors like: Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 Dec 14 15:44:00 WARNING[1215]: Failed

Semi-offtopic, Anyone know of voip providers who can provide tollfree number service? E.g. route 800-xxx numbers to our * ? Even better if they are familiar with * or can speak IAX ... -Dan

I've just upgraded to 1.6.2.6 on one of my test systems. I started out happy, with some improvements in transfers to Local() channels from a SIP channel, and much nicer verbose fax handling. However, something is really weird when I need to do System() calls. It was really, really weird. This was also affecting AGI, when I needed to read system variables from asterisk into an AGI Perl script.

I'm using Asterisk 1.4 and I'm wanting to set an account code for incoming calls. In the extensions.conf file I have the following: exten => s,1,SetAccount(1234) exten => s,n,Dial(SIP/1234) Then when I dial the extension the following error message pops up in the CLI: [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 pbx_extension_helper: No application 'SetAccount' for

> I'm using Asterisk 1.4 and I'm wanting to set an > account code for incoming calls. In the > extensions.conf file I have the following: > > exten => s,1,SetAccount(1234) > exten => s,n,Dial(SIP/1234) > > Then when I dial the extension the following error > message pops up in the CLI: > > [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 >

I've asked this question in the IRC Channel, and have had no happiness yet :-( I have incoming lines hooked to asterisk using X100P's. Unfortunately, when we cal forward our lines using the phone company, the line still rings about a half of a time. This is enough to get * to start 'simple switch' and after my 2 second wait, answer the line. Unfortunately, * doesn't see the

Hi there, I am configuring a small voicemail server and I am facing the following problem. Executing this command: exten => 1234,1,VoiceMail(${NUMBER}@test) When a user does not have a customized temporary greeting vm-intro message is played asking for the message to the user but when the user has already a temporary greeting both the temporary greeting and vm-intro are played. Basically

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1