Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

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1answer
198 views

Cheap PSTN Card with caller Id for Asterisk Linux server [closed]

For my house I just wanted to setup a caller Id system that would auto-reply on certain callers to avoid harements. Any sugestion for this? I'm thinking on a Linux + Asterisk + PSTN Card with caller ...
0
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0answers
36 views

ASA 5505 base license and Voice vlan [duplicate]

My current setup contains a cisco asa 5505 with two cisco sf300-24p switches, the data and voice vlans have been setup on the switches and I've started to setup the needed vlans on the ASA, but it ...
2
votes
5answers
240 views

VOIP Power outage strategy

In the past 2 years, we have had 4 instances where we have been without power for a period of 4-6 hours - due to construction crews cutting power lines, car accidents involving downed power lines, ...
5
votes
2answers
103 views

Trans-atlantic VOIP

We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the ...
2
votes
1answer
189 views

Asterisk 911 dialing rule

Our company is looking to replace our outdated PBX with VoIP. One function we really like about our current PBX is that it alerts the secretary when a line dials 911 and it shows the extension where ...
0
votes
1answer
405 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 ...
0
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1answer
494 views

How can I find a log of calls message a specific extension in Asterisk?

Question says it all. Trying to view a log of calls to/from an extension using Asterisk. Thanks in advance.
2
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1answer
163 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 ...
0
votes
1answer
160 views

Variable to store Asterisk extension dialed by user

I am using Asterisk for voice calling. I am using below context If user call back- [from-pstn] exten => _X.,1,Playback(demo-thanks) suppose my number is 74900 on which user can call back. ...
1
vote
0answers
367 views

UDP Range Nat Translation port forwarding RPT Traffic to inside host PBX Cisco ASA 5505 8.2a

I have an Asterisk based PBX that I'm providing VoIP services with, My gateway is a Cisco ASA 5505 running 8.2 and I only have one public IP Available along with other services I provide with that IP. ...
2
votes
1answer
441 views

Why does asterisk (and other systems) feel the need to regenerate in-call DTMF?

I am using Asterisk to interact with analog telephony devices that can be programmed and tested with DTMF interaction. Some of these guys speak rather quickly. Too quickly, you could convincingly ...
1
vote
1answer
286 views

Troubleshooting a failing SIP connection

I have a linux Asterisk server that is behind a firewall. The firewall has three interfaces : eth0 is the LAN, where Asterisk resides, eth2 is the default gateway to the internet (via an ADSL modem ...
0
votes
1answer
478 views

Networking issue with asterisk server

I replaced an IVR machine for incoming call after going down. It is running asterisk 1.4.23 on ubunutu 10.04 I decided to put the server behind iptables because my server was under brute force attack. ...
0
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0answers
968 views

Not able to register via SIPclient on asterisk server?

I'm unable to register a SIP client on my Asterisk server. What changes do I need to make to my Asterisk server configuration to fix this? Here are my configuration files for Asterisk. sip.conf ...
0
votes
2answers
1k views

PBXact Behind ASA 5505 nat udp range 10000-20000

I'm installing a new pbxact phone server onsite and I've ran into an issue using a 3rd party sip, I've got the acl's for the traffic created and the traffic is allowed, I'm assuming i need to create a ...
2
votes
1answer
722 views

Asterisk IVR - input variable and go to an outbound URL

I have an Asterisk outbound IVR system set up, and I'd like it to talk to my LAMP stack on a different server. I've set up the PHP end to receive data from the IVR server and record it in MySQL, but I ...
1
vote
1answer
551 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
3
votes
2answers
7k views

Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's ...
1
vote
1answer
1k views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
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 ...
0
votes
1answer
249 views

can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) ...
0
votes
2answers
124 views

Determine which user initiated call in Asterisk

I had the following code in my extensions.conf file: [local] exten => _NXXNXXXXXX,1,Set(CALLERID(name)=${OUTGOING_NAME}) exten => _NXXNXXXXXX,n,Set(CALLERID(num)=${OUTGOING_NUMBER}) Now I ...
0
votes
1answer
118 views

Parse and validate Asterisk dialplan before commiting

I recently made a number of changes to my Asterisk dialplan and would like to validate these changes before I commit. I am thinking more from a "write code" -> "compile" -> "debug" approach. I am ...
1
vote
1answer
489 views

Asterisk outbound calls hanging up imediatellly on Zap trunks

The inbound calls work fine. Outbound calls fail as follows: Dial("SIP/xxxxxxxxx","ZAP/g0/14442223333|300|") Couldn't call g0/14442223333 Hungup 'Zap/2-1' Everyone is busy/congested at this time ...
1
vote
1answer
444 views

How to establish SIP connection, when SIP-proxy is required?

I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. ...
1
vote
1answer
248 views

Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...
0
votes
1answer
550 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
2
votes
1answer
574 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
5
votes
3answers
2k views

How do I secure my Asterisk server?

Our asterisk server was compromised. some calls were made to Asia countries last weekend. Thought we have improved our network configuration, we still want to determine how the intrusion was done, we ...
2
votes
1answer
1k views

cant make outbound calls - asterisk

I have a basic Atcom IP01 with the following config Registered Voip (SIP) Trunk Registered Voip Phone - ext Dial Plan Outbound Call rule I made use of this manual that the manufacturer supplies: ...
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.
1
vote
2answers
1k views

Outgoing calls from vtiger to an asterisk extension

I've successfully installed vtiger and asterisk on Ubuntu, and I managed to configure them such that I get an incoming and an outgoing popup in vtiger. The problem is that when I click on an extension ...
2
votes
1answer
602 views

Elastix and two ISDN lines

Since I am not an expert on communications, I will describe everything from the user perspective. We had a PBX with elastix embedded and two ISDN cards in it (we have two isdn lines). The mobo of the ...
0
votes
1answer
843 views

asterisk queues priority and penalty

queues.conf shared_lascall=yes strategy=rrmemory wrapuptime=15 A1,A2,A3 are members of 2 queues: queue1(Q1) and queue2(Q2) A3 has penalty 3 in Q1 where min/max penalty are 0/3 and A3 has penalty ...
0
votes
1answer
615 views

Automatically reconnect to ODBC sources?

I am using Asterisk 1.8.10.1 and a MySQL database connected via ODBC to store CDRs. When my MySQL database isn't available when Asterisk starts or has an outage while Asterisk is running, I would ...
0
votes
2answers
890 views

Timing Source on Asterisk [closed]

I have tried to look for simple language of explanation what is timing source for in asterisk? and why it has significant impact on the asterisk Anyone can give me insight will be appreciated
0
votes
1answer
169 views

Enable auto dial on Polycomm VoIP phone? [closed]

I was wondering if it was possible to make the polycomm devices automatically dial out without the use of the "Dial" button. We have user's who are mistaking this for the phone being broken when they ...
3
votes
2answers
2k views

How can I make Asterisk keep track of dynamic SIP agent statuses?

I am setting up a new server using Asterisk 1.8.11-certified4. In testing, we're seeing that agents dynamically logged into the queue will receive a second queue call as a call-waiting when ...
0
votes
2answers
135 views

handle actions diferent to Command with Asterisk::AMI

I'm learning Asterisk :: AMI, but all examples deal with the action Command. i've tryed to run the following action (no success) my $action = $astman->action({ Action => "Agents" }); i ...
-2
votes
1answer
658 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
2answers
88 views

Installing a GUI over a running Asterisk

The server is Ubuntu Server 10.04 have asterisk-1.6.2.2.tar.gz installed. I have read that most of the GUI for Asterisk administration are something more than simple GUIs, and overwrite the previous ...
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 ...
1
vote
3answers
973 views

Elastix and extensions.conf

My VOIP provider required me to add this into /etc/asterisk/extensions.conf: exten => h,1,Hangup exten => _00X.,1, Answer exten => _00X.,n, Set(CALLERID(num)=96354205144239) ; exten => ...
0
votes
2answers
955 views

Skype Connect and Elastix for incoming and outgoing calls

I ordered Skype Connect, And i want to integrate skype connect with my Elastix server to handle incoming and outgoing calls. I created new SIP Trunk through GUI with the following info : ...
0
votes
1answer
1k views

Elastix call-center with computer telephony crm integration

The title isn't explanatory enough but I can't explain it better. I want to setup a call-center using elastix which would have the following feature: Suppose bob(client) calls the center and ...
2
votes
0answers
320 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
599 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 ...
3
votes
2answers
858 views

cli commands not working asterisk on ubuntu

I installed VMware and then Ubuntu. I am running asterisk on top of it. I started the asterisk server successfully by following a tutorial. After making few changes into the sip.conf, I want to ...
2
votes
2answers
1k views

asterisk incoming/miss call notification (to skype)

My environment: Centos 5.6, Asterisk with freepbx , skype i.e.sends message with parameter skype.sh skype_user message. Now i wanted to send skype notification so that my asterisk server notification ...
5
votes
2answers
3k views

Asterisk does not recognise DTMF tones from mobile phones

We have an Asterisk 1.8.7.0 (the Elastix derivative) switchboard. Every since a month ago, seemingly out of the blue, the switchboard does not recognise DTMF tones any more from mobile phones. ...