similar to: Variables for dial plan

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before

This was posted at The Asterisk Blog Forums <http://asteriskblog.com/forum/> Click here for the original post.<http://asteriskblog.com/forum/viewtopic.php?t=20> I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore

Hi All, I am a new bee to *. I just installed Asterisk@home on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In

I have two Asterisk boxe. One is running 1.6 and the other 1.2 The users on the 1.2 system press # plus a local 7 digit number to place local calls through the trunk to the 1.6 box. For some reason this dial pattern fails right away with "unavailable". There is no activity in the CLI. Other patterns for the trunk work just fine. Dial pattern: #|. or #|NXXXXXX exten =>

Will this work to match any number from the 770,404, or 678 area codes? _[404|770|678]NXXXXXX If this won't work, is there a pattern that will do this? Yours, Michael Munger, dCAP 404-438-2128 michael at highpoweredhelp.com <mailto:michael at highpoweredhelp.com> Attachment encrypted? click here <http://www.highpoweredhelp.com/tutorials/wincrypt/> .

On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz <motty.cruz at gmail.com> wrote: > here is what I did and worked for me: > > exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) > > exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have a leading underscore on your patterns. Your users

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

Hello All, Hope you all are fine and good ... I m facing an odd problem. I am able to dial some local and Mobile Number and some are not working. I also try to remove and re create dial plans , but it's not working. I live in New Delhi and Area Code is {011} but i am bale to call the number without adding {0} by just {11}, Previously i was thinking it was PRI problem, so i called the

Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No

I think I am overlooking a call or concept in R to help me easily and quickly restructure my data.frame:   Sometimes the data I receive looks like: VariableName, Run1, Run2, Run3, Location temp,               15.0,  16.0,  17.0, There   And other times it looks like: VariableName, Run, Location temp,               17.0, There   I would like to use the header information in order to be able to

I have a Windows 2008 domain (one Win 2008 DC, one Win 2012 R2 DC.) I am trying to join a Solaris 11 machine to the domain for both Samba and other services. For "unix" logins and ssh, Solaris 11 is configured to use LDAP for user and group lookup and kerberos for authentication. The "kclient -T ms_ad" command joins the Solaris machine to the AD domain. It even

Hello, I am having troubles with the following configuration in that it produces the following errors: *Masterserver:* On the server (yea, I know) side is a Linux machine called "*masterserver*". It should have a VPN IP of 192.168.2.1 and it sits behind a pretty generic ADSL Router (with port 655 forwarded) which can be found via a dynamic host address. A small snippet of errors this

Dear R experts This probably seems very easy to you guys, but I'm a beginner and would be really glad if someone helped me with this: I am trying to automate the execution of an R script (let's call it "myscript.R") by passing a variable from a bash script to myscript.R. I know I can use the command Rscript, but I don't know how to declare in bash which variable will be

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

Hi,All, I wang to use AGI in asterisk1.4. AGI file / myperl.agi #!/usr/bin/perl use strict; ...... print STDERR "7. Testing GET VARIABLE..."; print "GET VARIABLE EXTEN \"\"\n"; my $result = <STDIN>; &checkresult($result); ...... when the agi execute; asterisk conosle show that : AGI Rx << GET VARIABLE EXTEN "" AGI Tx >>

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten => s,1,NoOP same => n,Answer same => n,Set(filename=$PAGE) same => n,Wait(1) same =>

I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried: ; strip leading plus sign same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} ) same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = ${CALLERID(num):1})

Hi Everyone, I want to call first party using a .callfile and a second party using a context and then bridge the two calls. I MUST make sure that first party picks up first and then the second party should be dialed. Trying the following using an internal extension works nicely and the playback file is play after the extension picks up. But using the same method for calling an outside phone

How to is possible this? > [root at lucatest ~]# ls -lid /var/log /var/log/ispconfig /var/log/ispconfig/httpd /var/log/ispconfig/httpd/prova.it /var/log/ispconfig/httpd/prova.it/test /var/www /var/www/clients /var/www/clients/client1 /var/www/clients/client1/web3 /var/www/clients/client1/web3/log /var/www/clients/client1/web3/log/test > 706 drwxr-xr-x. 15 root root 4096 29 mag 08:44

Here is an example of a couple of macros that help me where I have a SOHO with a home phone line and a work phone line. If I pick up line 2 my work line I would prefer the call I make to go out my office phone line same with if I pick up line 1 my home phone line I would prefer it go out my home line but want it to roll if needed. So with this little macro it is possible for that to happen.