similar to: Are .call files working with extensions.ael ?

Hi, Is the following behaviour a bug or a feature ? Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces : [Mar 16 15:39:36] WARNING[25547]: 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 16 15:39:36] WARNING[25547]: pbx_spool.c:457

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

Steve, I am trying to get tx_fax to work. I am using a TDM2401E card. I have a 3 page fax and I only receive the first page on every attempt. I think I have enabled debug output below. Can you tell me what the problem might be? I am using snapshot from oct 26. asterisk 1.2.13 and libtiff 3.6.1-12 from redat/centos 4.4. THanks, Jerry --------- Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69

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

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:

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

I'm using Slackware 11. I unistalled the package that provides libtiff 3.8..... and installed the most current 3.7.... for lib tiff. I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them. created a simlink: ln -s asterisk-1.4.0-beta3 asterisk I've compiled spandsp from as follows cd /usr/src wget

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

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

Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168>

Greetings! Sample program: int main(int argc, char **argv) { int myLocal=0xAA; return 0; } Command: clang t.c -o a.o -c With above simple program we are observing that clang is stat-ing and trying to open various files from lib/tls location. Eventually all calls to "lib/tls" leads to ENOENT (No such file or directory)! <sample_strace>

Has anyone tried to (recently) install asterisk in a location not relative to /, as a non-root user? Ie editting the PREFIX directive in Makefile. Why? Several quite obvious reasons: a). Allows an asterisk user to be created, and operators to log into the box as asterisk user, without having root access. b). Much easier backups, because everything is beneath the same directory structure. c).

actually it sounds like a permission issue. You said you are doing it as root, but what is asterisk running as. I've found it is very sensitive, even to ownership. Make sure the owner:group is set to what Asterisk is running as before copying. Then, I've never had problems copying vs. moving - although I could imagine it might create problems in a race condition case. p From:

Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init

Hello everyone, I have just finished up work on a new release of AstLinux, the embedded/live cd/minimal built from scratch Linux distro centered around Asterisk. Most of the work for 0.2.8 has been on the ISO image (live/install cd), the init system and further system customization with new rc.conf variables. - The ISO image now allows you to install AstLinux to a local disk/CF card,

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hi, Whenever Asterisk restarts, some values such as DEVSTATE ones, should be reset. How would you proceed to have those settings done anytime Asterisk restarts ? I was thinking of using #exec statement in appropriate .conf file. Ant better idea ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux