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 fine, but what do I need to do to add the additional 4 phone numbers/lines? I want it to act exactly as before, there’s nothing special about the new lines. We just need more lines so [...]

I want to make a real time program that shows the inbound calls. The problem is that i don’t know where to get the call status, like hold, calling or answered. Asterisk has a Variable ${CALLERID(num)} to get the number but don’t know where to get the rest of the info. Asked by sebastian You will get it through PHP-AGI. You can connect to Asterisk Cli using this. The command “queue show http://sysadminman.net/blog/2012/asterisk-elastix-queue-and-agent-wallboard-3604 Hope the [...]

Let’s say I have an Asterisk system with a bunch of connections: there are phones (who register itself with *) and providers (who wish to establish SIP trunks to put a lot of calls over, with different Caller IDs). Here is my vision about how calls should be placed over an authenticated SIP trunk: remote end of SIP trunk should send INVITE‘s with From field set to it’s identity (username for authentication) and Contact field [...]

We have a server onto which we plan to install CentOS 6.2. The server will host Asterisk for our office, but is not a webserver. If we install the minimal installation for CentOS – that is, no GUI for the OS, then that means that Asterisk will also not be able to be interfaced with a GUI such as FreePBX or Elastix. I found this thread useful in the consideration: Should I use an Asterisk [...]

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, is it ? Thanks. Asked by John Yes, freepbx web have shortcut for Echo. It can be founded in “Feature Code”. Default is *43 and it is enabled. app_echo.so by default is on in all major asterisk’s version. Note, freepbx is web,and [...]

Have an issue with the redial numbers. Outgoing line is configured to the digit 0, so on the dial plan it is set to “0|.” (without the quotes). Now all missed calls from the UK, starts with 44 on the IP Phone. If i hit redial, it refuses the connection as expected. What do i need too add, that when it sees 44, it either replaces it back with 44 or add additional 0. I [...]

First off: I’m not sure if this should be on superuser or here. I have recently built a few Asterisk boxes with OpenVOX FXO/FXS ports little or no trouble. My current project is building an Asterisk box with SIP trunks. My current employer insisted on getting Skype Business/Skype connect for that purpose. After reviewing the Skype Connect plan, I agreed, because I thought it is going to be straightforward: Purchase G729 licences and setup SIP [...]

I’m in pain. We are moving to a SIP based VOIP system and for whatever reason, we could not get our hosted Asterisk solution to work with our Sonicwall. Our VOIP provider gave up and is recommending an open source vendor, pfSense. A little background: We have about 30 users in our network. We use a few IPSec VPN connections for remote networks. I would like, but don’t need, application layer filtering. We’re active internet [...]

I have a Debian host $ uname -a Linux voip 2.6.25-2-amd64 #1 SMP Mon Jul 14 11:05:23 UTC 2008 x86_64 GNU/Linux Old asterisk asterisk 1:1.4.21.2~dfsg-3 It has been working for a long-long time $uptime 13:50:37 up 1047 days, 21:02, 7 users, load average: 0.17, 0.19, 2.81 Today the host was froze, pings were answered , but I could not ssh, no dhcp or any other services. After I logged in from the console and looked [...]

There seems to be a hard limit on incoming calls set a certain way- a telephone engineer told me about this, but I don’t know the specifics. When on a call, the moment the call hits 15 minutes it drops. Does anyone know the specifics of this behaviour, and how to change it? Asked by chrism2671 This is a classic symptom of a NAT session timing out on a firewall. Options without changing your firewall: [...]