What ports to open for Vicidial Vicibox Asterisk
Posted on May 13, 2011 by Paul White
For those of you who are setting up your own Asterisk based dialer using one of the many variants such as Vicidial
, or GoAutoDial
, you might have found that one of the most painful parts of setting it up is knowing what settings will make it work. Specifically what ports do you forward from your router, and what special settings do you need to set in your sip.conf file. After trial and error I have discovered the settings that work.
Dell Pentium D Midtower Server with 250 GB SATA
and 4 GB RAM
32 bit installation of Vicibox
( free off their website )
TouchTone Communications Sip Trunk with 4 lines
Data Service: AT&T U-Verse 20 Mbit downstream / 4 Mbit upstream.
DSL Modem: All in One DSL Modem / Wireless Router / 4 Port Router
Goal is to run our Predictive Dialer supporting a dozen agents at once.
What installation of Vicidial / Asterisk am I using?
I am running with the latest version of Vicibox
using the 32bit install CD. We installed it to a Dell Server ( Mid tower ) that has and Intel Pentium D 2.8 Ghz. It has 2 GB of RAM, and is running two hard drives
1 ( mirroring ). The installation went very smooth, and we ran all the Zypper updates after the initial installation to make sure we had all the latest updates. Asterisk version 18.104.22.168-vici
What Network setup am I using for Vicidial / Asterisk
First of all I am assuming you have a setup similar to me, meaning you have your server sitting at your home or office and your server is plugged into a router and is not directly plugged into your Cable Modem or DSL Modem. In our case we have AT&T Uverse, and they gave us one of those all in one boxes ( Modem, Router, Wireless ). Your Server will need to stay on a static IP address on your local network. You can usually set this from within your router's webadmin system. All it does is tie the MAC address of your server to a specific Local IP address. This way when you reboot the server or the router, the same local IP will be assigned to your server. In our case we set the router to 192.168.1.75 on our local network. Its also a good idea to purchase a static IP from your ISP. Although this is not 100% needed if you have cable modem, as most cable modem companies like comcast
tend to lease your IP to you for 6 months to 18 months. Normally the only time cable modem providers
will change your Public IP would be after an outage or network maintenance. If you have DSL, then usually your modem will get a new IP almost daily, so its recommended you pay the extra $10 - $15 / month for 8 static IPs ( 5 usuable, 3 for gateway and such ). Then for convenience I usually buy a domain and point its DNS to that IP. That way our agents can type a domain into their soft phone settings rather than an IP that might change.
What Ports do I forward for Vicidial / Asterisk?
You will need to forward the following ports for Vicidial
TCP 5060 ( For SIP )
UDP 5060 ( For SIP )
UDP 10000 - 20000 ( For RTP )
What special settings do I need in sip.conf
If you are experiencing problems where the calls are made but you can't hear the person on the other line, and if you are logged into Asterisk it gives you errors having to do with RTP failed packets or something similar, then this is likely either a firewall issue or you need to input the correct settings in your sip.conf
Here are the magic settings I used in my sip.conf
first login via SSH to your server.
Then there are three lines that you must have.externip=Your public IP address
localnet=Your local IP address / Subnet
So lets say your public IP address ( the one agents will type into there soft phone's settings ) is 22.214.171.124, and your server's IP on your local network is 192.168.1.75. And your local network uses the subnet mask of 255.255.255.0. Your settings would look like thisexternip=
If you see multiple localnet entries, I am not 100% but I believe you can delete them. I did and everything was working fine.
After you change the settings in your sip.conf you will need to reboot the server so they take effect.
Hope this helps others.UPDATE 5/14/2011
This morning I got a call from my client saying the dialer is not working.
I connected my softphone ( X-Lite 4.0 ) and tried to manually make a call, and even though the call went through, there was no voice or audio. This leads me to believe that either there is a problem with our server, a problem with the firewall on the Uverse Modem / Router, or a problem with our SIP trunk provider. I am almost 100% certain its not my laptop or my home network as everything was working yesterday without any problems, and I haven't changed anything on my side.
I am about to just DMZ the server on our Uverse box which isn't great for security
, but sometimes its the only way. Another option is to DMZ the Uverse Modem shutdown the wireless on it, and basically turn it into a simple stand alone Modem. and then add our linksys wireless router and just configure everything around that. I will update this post again once I get the bugs worked out. I am sure I am not the only one having these issues. UPDATE 5/16/2011
Even though this is not ideal for security
, we have DMZ the router for the Server. Since doing that everything has been working perfect. We are going to leave it like this for a few days to make sure everything stays this way. Then we will add a linksys router to the setup. At this point I am convinced the U-Verse All in one Modem Router box is somewhat defective. I will update this blog again in a few days.UPDATE 5/17/2011
So far so good. Since going DMZ with the U-Verse All in one Modem / Router, the server has been running perfectly. Just a note for anyone who wants to do this themselves. I recommend running the update IP script on your dialer, which changes the IP from the local IP ( 192.168.1.x ) to your public IP. Then changing your router's configuration to DMZ to your server, then reboot the server. What is even more amazing is how much faster the server seems to run. Seems the U-Verse Modem has a rather slow firewall.