similar to: 3-way calling woes... Nasty static and inconsistent flash detection?

Hi All, I am a newbie and having some trouble getting Asterisk to work. I have checked out zapata zaptel libpri and asterisk (both v1-0_stable and regular--in separate directories). All built according to the documentation I have found and installed correctly. I have modified/created zaptel.conf zapata.conf as attached. I get the following messages/error during asterisk startup. ==

I am having a hell of a time getting adobe flash to work on a recent 5.2 install. >From rpm install: Running firefox as root -- I have verified there is a symlink in /usr/lib/mozilla/plugins that points to the actual plugin located in /usr/lib/flash-plugins nada. SoI then copied the plugin directly into /usr/lib/mozilla/plugins and eliminated the symlink from the equation. nada. I then

I am having dial out issues and was hoping someone could shed some light. The problem is Intermittent: extensions.conf [globals] ; Trunk Info for outbound calls via PSTN - See the zapata.conf file in /etc/asterisk TRUNK=ZAP/G1 ;Trunk Interface ;MSD digits to strip (usually 1 or 0), 1 = remove a leading 9 TRUNKMSD=1 ; -------------------------------------------------- ; [trunklocal] - Defines

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 >

Greetings all, I'm not real good with dial plan programming and need some help. I've looked at the 2nd edition of the Asterisk book about GotoIf() and have a basic idea what I need to do but not sure about the correct way or the best way, to set it up. I need to branch based on whether the dialed number is long distance (international or not) or not. I have branch offices on SIP and IAX

Hello All, I'm trying to dial out with no luck. I'm using Asterisk@Home defaults. I have one X100P card and SJPhone. *CLI> dial 96985628 No such extension '96985628' in context 'default' Here is my exten [trunklocal] ; ; Local seven-digit dialing accessed through trunk interface ; exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>

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

On Tue, Sep 25, 2018 at 10:20:07AM +0200, Fabien Dupont wrote: > "uuid": "7ee62bdc-1c9d-4193-bb8f-fdbbbdfded0f" > } > > The information we need is the uuid. I don't have a way to test it, so this is only compile tested, however something like this? Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones Read my programming

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

I had to return my TDM11B because it put the PSTN line 'off hook' the moment I plugged it in and wouldn't hang it up. The new card seems to work because I can actually make an outgoing call from the FXO port to my cell phone, so I'm pretty happy about that. But Asterisk doesn't recognize incoming calls from the PSTN. If I dial my home phone from my cell phone asterisk

I'm trying to add an emergency dial to my context. However, when I try to dial it, I get caught in an endless loop. For debugging, I have pared out nearly all the control flow and just have ChanIsAvail() and Dial() called. Using two different extensions to call teh same number, I get two different actions by *. Here is the vvverbose output: -- Starting simple switch on

Hello, all. In section 4.3.10 of the Asterisk Handbook, there is an example of an LCR/Failover Trunking scenario. I've tried it, and it works, as long as I fail over from something else to ZAP, but I can't get it to "hunt" to the other context if the zapata channel (or group) is used first. Can anyone help? Here is my extensions.conf, and the error message I get.

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

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 =>

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

When I attempt to start asterisk with my modem setup listed it will not start attached are the error messages i get and also the modem.conf that i am currently using. Any assistance would be greatly appreciated. running CVS ver 12/7/03, modified only to allow the RxFax and TxFax to compile and run with it (from http://www.opencall.org) just e-mail me privately if you need more info Thanks in

Any ideas? Thanks. [root@localhost root]# asterisk -r Unable to connect to remote asterisk [root@localhost root]# asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Asterisk 0.7.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Parsing

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

On my current instance, the meta_data.json is the following: { "availability_zone": "nova", "devices": [], "hostname": "ims-host-1", "keys": [ { "data": "ssh-rsa