FreePBX is a Web-based front-end to the Asterisk telephony platform.

learn more… | top users | synonyms

1
vote
0answers
30 views

How to know what number was dialled on incoming call

The scenario In the UK. I have one (BT openreach) phone line coming into a Cisco SPA232D. That phone line is ringable on a number, let's say 01235AAAAAA. I have a virtual number, 01235BBBBBB, which ...
1
vote
1answer
96 views

How to match numbers based on end of number?

I am setting up an Asterisk VoIP server (using FreePBX) and I need to identify all numbers that end with a particular pattern. This is required for routing purposes, that is, to find out whether they ...
0
votes
0answers
34 views

Asterisk Auto dial out issues

I'm trying to make auto dial out in asterisk. I have created call file in /tmp directory and moved the file to var/spool/asterisk/outgoing/. It's not working. Please help me to resolve this issue. ...
0
votes
1answer
122 views

FreePBX UK Dial Patterns Route Through Specific Trunk

I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly. I've got Freepbx installed and working properly - no issue there. I want to create a dial ...
1
vote
3answers
349 views

Asterisk flooded by incoming SIP connection from unknown peer

Hello i just installed an new asterisk configuration with freepbx and signed for a SIP account. I setup the SIP trunk with my provider data, launched the console with the asterisk -vvvr command to ...
0
votes
0answers
53 views

Asterisk using PORT NUMBER in FROM FIELD of SIP calls

I have an annoying situation with Asterisk 13. I setup a SIP trunk that works OK to receive calls. However, I can't make calls even though my SIP provider is working (I can place calls with a regular ...
0
votes
1answer
317 views

Can't dial through SIP trunk: FreePBX/Asterisk

I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration: type=friend host=201.217.31.10 callerid=mynumber ...
0
votes
0answers
82 views

In FreePBX v12 how to increase extension ringing time when transferring (Attended transfer)

The FreePBX version I'm using is 12.0.9. And I have tried changing: "Call Forward Ring Time" on the Extension "Call Forward Ringtimer Default" in Advanced Settings "Default Follow Me Ring Time" in ...
1
vote
0answers
153 views

sip register success on one host but fail on another host in same local network

I acquired antisip account and setup asterisk (lets say PBX-A) to register. It worked fine. However, I attempted to do so with another asterisk (PBX-B). No luck this time. PBX-A and PBX-B are on the ...
0
votes
0answers
189 views

Debian DHCP for voip option 66 using tftp

I have configured my tftp and I am able to send configurations from the windows server when it runs dhcp option 66 tftp to my server but when I try to do it from the debian isc-dhcp-server I get ip ...
0
votes
1answer
553 views

centos 6.5 ssh rsa key fails

Alright I have read many many articles and followed a few different tutorials and I can not get my RSA key to work on my centos servers. The servers are running centos 6.5 with FreePBX on them and I ...
0
votes
0answers
72 views

Asterisk SIP calls between remote L2L VPN sites

I am running into an issue trying to set up another remote L2L VPN site with VoIP running SIP and Asterisk. Each remote Site (site 1 and site 2) Can talk to the Asterisk server and any phones on the ...
0
votes
0answers
16 views

Nortsatar-plus MICS on pri converting to server based PBX

I currently have a Cisco router providing our PRI, Fax, and Data. PRI to a norstar plus MICS PBX (NT7B53FA). I want to basically switch our norstar KSU for a server (2008 r2 standard or I can change ...
0
votes
0answers
360 views

Cisco IP Phone TFTP Error/Timeout

I'm trying to get Cisco 7942G IP phone working on FreeBPX setup. It has issues with getting things over tftp. I tried hard reset thing and I'm getting this in tftpd log: [root@localhost ~]# tail -f ...
0
votes
2answers
1k views

Allow specific SIP connection FreePBX

Hey I am using FreePBX with firmware: 5.211.65-14 and service pack 1.0.0.0 On this asterisk server I have everything up and running, but inbound phone calls might be rejected: ...
1
vote
0answers
138 views

FreePBX server (cent OS base) locks up with no errors or kernel panics

Bit of a baffling situation I've been dealing with for a few days. I have multiple CentOS headless servers (6.4) with the following stats: Core CentOS - 6.4 (Final) Kernel - ...
1
vote
2answers
2k views

Asterisk IVR setup: disallow direct dial for some extensions?

We've got PBX to configure. It's based on Elastix 2.4.0, freePBX 2.8.1 and Asterisk 1.8.20.0. Currently we have a setup for inbound calls like that: Inbound call > Announcement (30 sec) > IVR ...
2
votes
2answers
3k views

How do I 'activate' voicemail on an extension on asterisk-Freepbx

I have asterisk-Freepbx (Version 12) hosted on a debian 7 server. I have created an extension (Cisco IP phone SPA 504G). I am able to dial in and out. The extension is configured to go to voicemail ...
1
vote
2answers
856 views

elastix cdr stop working

CDR was working before 19 march. Unfortunately i dont remember what kind of changes i made to configuration, but this exactly not changes to CDR config. elastix 2.4.0 asterisk 11.7.0 mysql 5.0.95 ...
0
votes
0answers
96 views

Asterisk virtual machine stops responding to network connections when I connect with Phoner Lite

I've installed Asterisk Now with PreePBX from bootable CD to a VirtualBox machine. Then I updated through FreePBX all the modules available for update and noticed a strange behaviour: After I ...
2
votes
1answer
2k views

How to Remove SIP or Plus Prefixes from Incoming Calls on FreePBX?

How can I remove the "sip:+" (sip plus) prefix from incoming numbers? The number is being cached on my phone, and is an illogical format, so I cannot call that number back.
0
votes
1answer
442 views

Asterisk, How can i play an audio file [closed]

Here is the dial plan [testInComingCalls] exten => s,1,Answer exten => 30953025,1,Dial(SIP/20000,20) I would like to play an audio file as soon as somebody answered the call (Welcome Message)..
0
votes
0answers
123 views

Can't listen to port 4445; sqlite3

So I have a dedicated machine running Ubuntu 12.04 server with FreePBX 2.9. Earlier I was attempting to familiarize myself with SQL via sqlite3, but I received this dreaded error: SQLite header and ...
0
votes
1answer
294 views

Asterisk Register username with special character like “@”

I am using a SIP provider that has provided me with a username like: +1122233344@aaa.bb.com (Note this is only the username part) And has a numerical password. My Register string looks something ...
0
votes
1answer
736 views

Can't receive phone call on my FreePbx extension

I'm running out of hair to pull here. I have my freepbx installed and configured ok at the beginning. I was testing it with Google Voice account. I can call out and receive call. I'm sure I must did ...
1
vote
1answer
115 views

Installing FreePBX or Asiterisk-GUI on working server

We have server with Ububuntu and asterisk 1.8 on it without any web-interfaces. Our managers ask me to make them web-interface for looking over config, users, calls and so on. I've installed ...
0
votes
1answer
4k views

What is the “Allow Anonymous Inbound SIP Calls” option under “Asterisk SIP Settings” in FreePBX for?

I am looking for the canonical definition of the “Allow Anonymous Inbound SIP Calls” option under “Asterisk SIP Settings” in FreePBX. Also, how does it relate to "Allow SIP Guests"?
0
votes
1answer
421 views

AsteriskNOW server routes outgoing calls as if they are incoming

I am very new to Asterisk so this is undoubtedly a misconfiguration on my end. Right now my test setup behaves as follows: When I make an incoming call to an FXO port on my analog card, Asterisk ...
2
votes
1answer
166 views

Polycom IP 650 Custom display

I've been googling all over the place and haven't found a good way to do this. I'm hoping that someone out there has done something like this. I have Polycom 650's and 331 deployed to my users via a ...
1
vote
1answer
1k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
14
votes
2answers
1k views

Can anybody explain why to use Asterisk PBX rather than standard PBX from Matrix or Siemens [closed]

Can anybody explain why open-source Asterisk is favourable in comparison with a traditional PBX system from a vendor such as NEC/Syntel/Matrix or Siemens.
0
votes
1answer
226 views

FreePBX & Zoostorm

I'm having some issues trying to install the FreePBX Distro on a new Zoostorm box. It keeps asking me for drivers. Here's the error message: No Driver Found Unable to find any service of the type ...
-2
votes
1answer
689 views

Inbound Caller gets Fast Busy and Extension gets ringing

Inbound caller (1636) calls a sipstation direct dial to my phone on my desk. Cellphone sounds like fast busy but then desk phone rings. Cant connect the call. then when desk phone hangs up It ...
0
votes
1answer
4k views

What are the possible reasons for a SIP register failure?

I have setup an Elastix box (asterisk/freepbx based) and added a trunk for an external VoIP provider. When I get to the Asterisk command line interface and type sip show registry I always get the same ...
2
votes
0answers
327 views

Asterisk ${CALLERID(name)

I am working with asterisk (Asterisk 1.8.11.0), freepbx (2.x) and was building a dialplan (extension_custom.conf). I wanted to get caller Name from database which i have stored for example. CLI> ...
0
votes
2answers
607 views

How do I configure additional phone lines asterisk/trixbox?

I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card 1AEX410PELF and have 4 more analog lines coming into the Trixbox server. It still runs the 4 ...
2
votes
1answer
989 views

Can a FreePBX backup be restored to a different version?

I run a small PBX based on the FreePBX distro of Asterisk. The installation has been steadily upgraded but for various reasons, we want to start again on a new server with a clean install from the ...
2
votes
3answers
504 views

How to add more incoming/outgoing lines capability to Asterisk?

We have 4 people call us from around the country at once and it causes our phones to be busy. We want to be able to have at least 8 people call us at once without anyone getting the busy signal. ...
0
votes
1answer
274 views

Get Asterisk/Freepbx to ignore digits that were unnecessarily dialed (area code for local calls)

This is the second part of this question, worded better. Remove local area code on outbound calls, and remove "dial 1" for long distance in Trixbox/Asterisk I have removed the NEED to ...
1
vote
1answer
805 views

How to play and record wav in Asterisk

Please, how could I do these things on asterisk: 1) Externally (with some API) make a call from asterisk and play wav and hangup the call 2) How should I configure the extension/route/whatever on the ...
0
votes
1answer
66 views

How to install app_echo.so in FreePBX?

What is the proper way of installing app_echo module when FreePBX is installed ? I cannot find it in Module admin and manual editing of configuration files when freepbx presented is not recommended, ...
1
vote
2answers
1k views

Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk

We have Trixbox CE 2.8 and Asterisk 1.6. I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls. I also want to remove the +1 for long distance so ...
1
vote
2answers
965 views

FreePBX How to play an announcement for misdialled calls

Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately. Some of our users who ...
-1
votes
1answer
92 views

Is a PBX responsible for routing calls, or is that handled by the provider?

I am setting up a PBX that external users will connect to and I was wondering how important PBX placement was. Are calls routed through the PBX itself, or do they go directly through the provider ...
0
votes
1answer
233 views

Automatically log into SIP phone when user logs into Windows

We run an asterisk-based PBX, with FreePBX front end, configured in 'device and user mode' to support hot desking users. That is, a user arrives and sits at any desk, keys *11 into the phone (Cisco ...
3
votes
1answer
780 views

Automated reports with FreePBX 2.8.1.4

FreePBX has a 'Reports' tab that allows you to create a few reports and some of them you can export as a .csv or .pdf, but it doesn't really have any finer features than that. I would ultimately like ...
1
vote
1answer
1k views

GUI for Asterisk 1.8.8 on Debian 6 [closed]

Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
1
vote
2answers
201 views

Advantages and disadvantages to using a VPS (or any remote server) to host a VOIP/PBX system

My company currently runs a Trixbox Pro server locally. We're getting sick of the licensing fees and terrible community support and are considering migrating to something like FreePBX. I see that ...
2
votes
1answer
2k views

Using Asterisk and FreePBX how can I map extensions to outbound routes

I have a Trixbox server (Asterisk and FreePBX) that has multiple tenants on it. I need these tenants calls to go out via different outbound routes in order to split the bills at the SIP trunk provider ...
0
votes
1answer
1k views

Sip configuration for Elastix, for Terrasip (Provider) with Multiple DIDs

I have just purchased several DIDs with Terrasip I have configured my Elastix (freepbx) SIP trunk according to their suggestion: Please follow this template configuration. (valid for outbound/inbound ...