Category Archives: Linux


Ask someone with headphones and a lanyard in the halls of a datacenter what transport does DNS use, there’s a good chance the answer you’d get back is UDP Port 53.

But not always!

In scenarios where the DNS response is large (beyond 512 bytes) a DNS query will shift over to TCP for delivery.

How does the client know when to shift the request to TCP – After all, the DNS server knows how big the response is, but the client doesn’t.

The answer is the Truncated flag, in the response.

The DNS server sends back a response, but with the Truncated bit set, as per RFC 1035:

TC TrunCation – specifies that this message was truncated due to length greater than that permitted on the transmission channel.

RFC 1035

Here’s an example of the truncated bit being set in the DNS response.

The DNS client, upon receiving a response with the truncated bit set, should run the query again, this time using TCP for the transport.

One prime example of this is DNS NAPTR records used for DNS in roaming scenarios, where the response can quite often be quite large.

If it didn’t move these responses to TCP, you’d run the risk of MTU mismatches dropping DNS. In that half of my life has been spent debugging DNS issues, and the other half of my life debugging MTU issues, if I had MTU and DNS issues together, I’d be looking for a career change…

FreeSWITCH mod_python3 – Python Dialplans

Sometimes FreeSWITCH XML dialplan is a bit cumbersome to do more complex stuff, particularly to do with interacting with APIs, etc. But we have the option of using scripts written in Python3 to achieve our outcomes and pass variables to/from the dialplan and perform actions as if we were in the dialplan.

This is different to the Event Socket interface for Python I’ve covered in the past.

For starters we’ll need to install the module and enable it, here’s the StackOverflow thread that got me looking at it where I share the setup steps.

Here is a very simple example I’ve put together to show how we interact with Python3 in FreeSWITCH:

We’ll create a script in /usr/share/freeswitch/scripts/ and call it “”

from freeswitch import *
import sys 
def handler(session,args):

    #Get Variables from FreeSWITCH
    user_name = str(session.getVariable("user_name"))
    session.execute("log", "Call from Username: " + str(user_name))

    #Check if Username is equal to Nick
    if user_name == "Nick":
        session.execute("log", "Username is Nick!")
        #Convert the Username to Uppercase
        session.execute("set", "user_name=" + str(user_name).upper())
        #And return to the dialplan
        #If no matches then log the error
        session.execute("log", "CRIT Username is not Nick - Hanging up the call")
        #And reject the call
        session.execute("hangup", "CALL_REJECTED")

Once we’ve created and saved the file, we’ll need to ensure it is owned by and executable by service user:

chown freeswitch:freeswitch
chmod 777

In our Dialplan we’ll need to add the below to our logic to get called at the time we want it:

<action application="system" data="export PYTHONPATH=$PYTHONPATH:/usr/share/freeswitch/scripts/"/> <action application="python" data="CallerName"/>

After adding this to the dialplan, we’ll need to run a “reloadxml” to reload the dialplan, and now when these actions are hit, the Python script we created will be called, and if the user_name variable is set to “nick” it will be changed to “NICK”, and if it it isn’t, the call will be hung up with a “CALL_REJECTED” response.

Obviously this is a very basic scenario, but I’m using it for things like ACLs from an API, and dynamic call routing, using the familiar and easy to work with Python interpreter.

Scratch’n’Sniff – An easy tool for remote Packet Captures

A lesson learned a long time ago in Net Eng, is that packet captures (seldom) lie, and the answers are almost always in the packets.

The issue is just getting those packets.

The Problem

But if you’re anything like me, you’re working on remote systems from your workstation, and trying to see what’s going on.

For me this goes like this:

  1. SSH into machine in question
  2. Start TCPdump
  3. Hope that I have run it for long enough to capture the event of interest
  4. Stop TCPdump
  5. Change permissions on PCAP file created so I can copy it
  6. SFTP into the machine in question
  7. Transfer the PCAP to my local machine
  8. View the PCAP in Wireshark
  9. Discover I had not run the PCAP for long enough and repeat

Being a Mikrotik user I fell in love with the remote packet sniffer functionality built into them, where the switch/router will copy packets matching a filter and just stream them to the IP of my workstation.

If only there was something I could use to get this same functionality on remote machines – without named pipes, X11 forwarding or any of the other “hacky” solutions…

The Solution

Introducing Scratch’n’Sniff, a simple tcpdump front end that encapsulates all the filtered traffic of interest in TZSP the same as Mikrotiks do, and stream it (in real time) to your local machine for real time viewing in Wireshark.

Using it is very simple:

Capture all traffic on port 5060 on interface enp0s25 and send it to
python3 --dstip --packetfilter 'port 5060' --interface enp0s25

Capture all sctp and icmp traffic on interface lo and send it to
python3 --dstip --packetfilter 'sctp or icmp' --interface lo

If you’re keen to try it out you can grab it from GitHub – Scratch’n’Sniff and start streaming packets remotely.


Fixing Wireshark / TCPdump pcap: network type 276 unknown or unsupported Error

Ubuntu 20.04 repos have a fairly outdated release of Wireshark, and the other day when trying to open a packet capture I got the below error:

After doing an apt-get update && apt-get upgrade wireshark, the version of Wireshark, and the issue remained.

I have compiled Wireshark from source before, but it’s a real headache, so instead I just added the Wireshark PPA with:

sudo add-apt-repository ppa:wireshark-dev/stable
sudo apt-get update -y
sudo apt-get upgrade wireshark

And presto, latest (stable) build of Wireshark, and error gone when opening PCAPs!

Wireguard in Mikrotik RouterOS

Recently I’ve been using Wireguard to fix the things I once used IPsec for.

It was merged into the Mainline Linux kernel late last year, and then in RouterOS 7.0beta7 (2020-Jun-3) the system kernel on RouterOS was upgraded to version 5.6.3 which contains Wireguard support.

Unfortunately this feature is going to stay in the Unstable / Development releases for the time being until a kernel update is done for the stable release to 5.5.3 or higher, but for now I thought I’d try it out.

After loading a beta version of the firmware, under Interfaces I have the option to add a Wireguard interface, for clients to connect to my Mikrotik using Wireguard.

It’s nice and simple to see the public/private key pair (a new key pair is generated for each Wireguard instance which is nifty) that we an use to authenticate / be authenticated.

If we want to configure remote peers, we do this by jumping over to the Wireguard -> Peers tab, allowing us to setup Peers from here.

Obviously routing and firewalls remain to be setup, but I love the simplicity of Wireguard, and in the RouterOS implimentation this is kept.

Open5GS without NAT

While most users of Open5GS EPC will use NAT on the UPF / P-GW-U but you don’t have to.

While you can do NAT on the machine that hosts the PGW-U / UPF, you may find you want to do the NAT somewhere else in the network, like on a router, or something specifically for CG-NAT, or you may want to provide public addresses to your UEs, either way the default config assumes you want NAT, and in this post, we’ll cover setting up Open5GS EPC / 5GC without NAT on the P-GW-U / UPF.

Before we get started on that, let’s keep in mind what’s going to happen if we don’t have NAT in place,

Traffic originating from users on our network (UEs / Subscribers) will have the from IP Address set to that of the UE IP Pool set on the SMF / P-GW-C, or statically in our HSS.

This will be the IP address that’s sent as the IP Source for all traffic from the UE if we don’t have NAT enabled in our Core, so all external networks will see that as the IP Address for our UEs / Subscribers.

The above example shows the flow of a packet from UE with IP Address sending something to

This is all well and good for traffic originating from our 4G/5G network, but what about traffic destined to our 4G/5G core?

Well, the traffic path is backwards. This means that our router, and external networks, need to know how to reach the subnet containing our UEs. This means we’ve got to add static routes to point to the IP Address of the UPF / P-GW-U, so it can encapsulate the traffic and get the GTP encapsulated traffic to the UE / Subscriber.

For our example packet destined for, as that is a globally routable IP (Not an internal IP) the router will need to perform NAT Translation, but for internal traffic within the network (On the router) the static route on the router should be able to route traffic to the UE Subnets to the UPF / P-GW-U’s IP Address, so it can encapsulate the traffic and get the GTP encapsulated traffic to the UE / Subscriber.

Setting up static routes on your router is going to be different on what you use, in my case I’m using a Mikrotik in my lab, so here’s a screenshot from that showing the static route point at my UPF/P-GW-U. I’ve got BGP setup to share routes around, so all the neighboring routers will also have this information about how to reach the subscriber.

Next up we’ve got to setup IPtables on the server itself running our UPF/P-GW-U, to route traffic addressed to the UE and encapsulate it.

sudo ip route add dev ogstun
sudo echo 1 > /proc/sys/net/ipv4/ip_forward
sudo iptables -A FORWARD -i ogstun -o osgtun -s -d -j ACCEPT

And that’s it, now traffic coming from UEs on our UPF/P-GW will leave the NIC with their source address set to the UE Address, and so long as your router is happily configured with those static routes, you’ll be set.

If you want access to the Internet, it then just becomes a matter of configuring traffic from that subnet on the router to be NATed out your external interface on the router, rather than performing the NAT on the machine.

In an upcoming post we’ll look at doing this with OSPF and BGP, so you don’t need to statically assign routes in your routers.

Remember Bash History Forever

History in Bash is a huge time saver.

That beautifully crafted sed command you put together to replace the contents of something a few months ago? Just search through your Bash history and there it is.

Previously I’d been grepping the output of history to find what I was looking for, and now I’ve fallen in love with the search feature, but by default, many Linux distros limit the number of lines in the Bash history to 2,000. If you’re a regular Linux user, this isn’t cutting the mustard.

By default the bashrc file that ships with Ubuntu is limited to 2,000 lines or 1MB,

We can change all this very easily, by editing the ~/.bashrc file (Bash shell script), upping the limit of entries we keep. While you’re at it adding HISTTIMEFORMAT allows you to timestamp the commands you’re running, and the PROMPT_COMMAND below also writes immediately, so you won’t get lost data or missing stuff that you’ve just run in another terminal.

Example contents of ~/.bashrc:

export HISTSIZE=100000
export HISTFILESIZE=200000
export HISTTIMEFORMAT='%d/%m/%y %T '

And you can apply the changes with:

source ~/.bashrc

Being mean to Mikrotiks – Pushing SMB File Share

I’d tried in the past to use the USB port on the Mikrotik, an external HDD and the SMB server in RouterOS, to act as a simple NAS for sharing files on the home network. And the performance was terrible.

This is because the device is a Router. Not a NAS (duh). And everything I later read online confirmed that yes, this is a router, not a NAS so stop trying.

But I recently got a new Mikrotik CRS109, so now I have a new Mikrotik, how bad is the SMB file share performance?

To test this I’ve got a USB drive with some files on it, an old Mikrotik RB915G and the new Mikrotik CRS109-8G-1S-2HnD-IN, and compared the time it takes to download a file between the two.

Mikrotik Routerboard RB951G

While pulling a 2Gb file of random data from a FAT formatted flash drive, I achieved a consistent 1.9MB/s (15.2 Mb/s)

nick@oldfaithful:~$ smbget smb://
 Password for [nick] connecting to //share1/ 
 Using workgroup WORKGROUP, user nick
 Downloaded 2.07GB in 1123 seconds

Mikrotik CRS109

In terms of transfer speed, a consistent 2.8MB/s (22.4 Mb/s)

nick@oldfaithful:~$ smbget smb://
 Password for [nick] connecting to //share1/ 
 Using workgroup WORKGROUP, user nick
 Downloaded 2.07GB in 736 seconds

Profiler shows 25% CPU usage on “other”, which drops down as soon as the file transfers stop,

So better, but still not a NAS (duh).

The Verdict

Still not a NAS. So stop trying to use it as a NAS.

As my download speed is faster than 22Mbps I’d just be better to use cloud storage.

Control + R in Bash = Life Changed

I’m probably late to the party on this one.

I was working with someone the other day on a problem over a video call and sharing my screen on a Linux box.

They watched as I did what I do often and Grep’ed through my command history to find a command I’d run before but couldn’t remember the specifics of off the top of my head.

Something like this:

How I’ve always found commands from my history

Then the person I was working with told me to try pressing Control + R and typing the start of what I was looking for.

My head exploded.

Searching through Bash command history

From here you can search through your command history and scroll between matching entries,

I cannot believe it’s taken me this long to learn!

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Docker Cheatsheet

I kept forgetting the basic Docker commands, so I made a cheat sheet for myeslf and thought I’d share:

List running Containers:
docker ps
List all Containers (Including stopped)
docker ps -a
Start a Container
docker run sdfjkdskj/sdfdsafa
List Images
docker image list
Build an Image
docker build -t myapp:v1 .
Connect to Shell of running Container
docker exec -it intelligent_chebyshev /bin/bash

mkcert – Simple Localhost Certs

Oftentimes I’m developing something locally and I need an SSL Certificate.

I’m too cheap to buy a valid SSL cert for a subdomain like “” and often the domain changes based on what I’m doing,

LetsEncrypt is great, but requires your server to be public facing and be a web server, which for dev stuff isn’t really practical,

Enter mkcert – a tool that allows you to generate valid SSL certificates on your machine for any domain, the catch is that it’s only on your machine.

I’m working on a WebSocket platform at the moment, which requires an SSL certificate.

So I set an entry in my hosts file to point “webrtc” to the IP of one of the machines,

I then generated the cert on my local machine,

mkcert -install webrtc 

Which outputs the certificate and private key, which I copied it onto the server I’m working on, twiddled some knobs in Apache2 and presto, valid cert!

The downside is of course anyone else going to this site would see the cert as invalid, but as it’s just me, it doesn’t matter!

You can get Mkcert from GitHub.

Kamailio Bytes – Siremis Installation

Siremis is a web interface for Kamailio, created by the team at Asipto, who contribute huge amounts to the Kamailio project.

Siremis won’t create your Kamailio configuration file for you, but allows you to easily drive the dynamic functions like dialplan, subscribers, dispatcher, permissions, etc, from a web interface.

Siremis essentially interfaces with the Kamailio database, kamcmd and kamctl to look after your running Kamailio instance.


I’ll be installing on Ubuntu 18.04, but for most major distributions the process will be the same. We’re using PHP7 and Apache2, which are pretty much universal available on other distros.

First we need to install all the packages we require:

apt-get update

apt-get upgrade

apt-get install kamailio* mysql-server apache2 php php-mysql php-gd php-curl php-xml libapache2-mod-php php-pear php-xmlrpc make

Enable apache2 rewrite & restart Apache

a2enmod rewrite
service apache2 reload

Next we’ll download Siremis from the Git repo, and put it into a folder, which I’ve named the same as my Kamailio version.

cd /var/www/html/
git clone kamailio-5.1.2

Now we’ll move into the directory we’ve created (called kamailio-5.1.2) and build the apache2 config needed:

cd kamailio-5.1.2/
make apache24-conf

This then gives us a config except we can put into our Apache virtual host config file:

We can now copy and paste this into the end of an existing or new Apache virtual host file.

If this is a fresh install you can just pipe the output of this into the config file directly:

make apache24-conf >> /etc/apache2/sites-available/000-default.conf
service apache2 restart

Now if you browse to http://yourserverip/siremis you should be redirected to http://yourserverip/siremis/install and have a few errors, that’s OK, it means our Apache config is working.

Next we’ll set the permissions, create the folders and .htaccess. The Siremis team have also created make files to take care of this too, so we can just run them to set everything up:

make prepare24
make chown

With that done we can try browsing to our server again ( http://yourserverip/siremis ) and you should hit the installation wizard:

Now we’ll need to setup our database, so we can read and write from it.

We’ll create new MySQL users for Kamailio and Seremis:


mysql> GRANT ALL PRIVILEGES ON siremis.* TO siremis@localhost IDENTIFIED BY 'siremisrw';

mysql> CREATE USER 'kamailio'@'localhost' IDENTIFIED BY 'my5yhtY7zPJzV8vu';

mysql> GRANT ALL PRIVILEGES ON * . * TO 'kamailio'@'localhost';


Next up we’ll need to configure kamctlrc so it knows the database details, we covered this the Security in Practice tutorial.

We’ll edit /etc/kamalio/kamctlrc and add our database information:

Once that’s done we can create the database and tables using kamdbctl the database tool:

kamdbctl create

I’ve selected to install the optional tables for completeness.

Once this is done we can go back to the web page and complete the installation wizard:

We’ll need to fill the password for the Siremis DB we created and for the Kamailio DB, and ensure all the boxes are ticked.

Next, Next, Next your way through until you hit the login page, login with admin/admin and you’re away!


If you have issues during the installation you can re-run the installation web wizard by removing the install.lock file in /var/www/html/kamailio-5.1.2/siremis

You can also try dropping the Siremis database and getting the installer to create it again for you:

mysql> drop database siremis;

Setup HOMER SIP captagent and HEP processor on Ubuntu 18.04

There are a number of ways to feed Homer data, in this case we’re going to use Kamailio, which has a HEP module, so when we feed Kamailio SIP data it’ll use the HEP module to encapsulate it and send it to the database for parsing on the WebUI.

We won’t actually do any SIP routing with Kamailio, we’ll just use it to parse copies of SIP messages sent to it, encapsulate them into HEP and send them to the DB.

We’ll be doing this on the same box that we’re running the HomerUI on, if we weren’t we’d need to adjust the database parameters in Kamailio so it pushes the data to the correct MySQL database.

apt-get install kamailio* kamailio-mysql-modules captagent

Next we’ll need to configure Kamailio to capture data from captagent, for this we’ll use the provided config.

cp homer-api/examples/sipcapture/sipcapture.kamailio /etc/kamailio/kamailio.cfg

/etc/init.d/kamailio restart

Next we’ll need to configure captagent to capture data and feed it to Kamailio. There’s two things we’ll need to change from the default, the first is the interface we capture on (By default it’s eth0, but Ubuntu uses eth33 as the first network interface ID) and the second is the HEP destination we send our data to (By default it’s on 9061 but our Kamailio instance is listening on 9060).

We’ll start by editing captagent’s socket_pcap.xml file to change the interface we capture on:

vi /etc/captagent/socket_pcap.xml 
HOMER Captagent Interface Setup
HOMER Captagent Interface Setup

Next we’ll edit the port that we send HEP data on

vi /etc/captagent/transport_hep.xml
Set HEP Port for Transport
Set HEP Port for Transport

And finally we’ll restart captagent


Now if we send SIP traffic to this box it’ll be fed into HOMER.

In most use cases you’d use a port mirror so you may need to define the network interface that’s the destination of the port mirror in socket_pcap.xml

Kamailio Bytes – SCTP

I’ve talked about how cool SCTP is in the past, so I thought I’d describe how easy it is to start using SCTP as the Transport protocol in Kamailio.

I’m working on a Debian based system, and I’ll need to install libsctp-dev to use the SCTP module.

apt-get install libsctp-dev

Next we’ll edit the Kamailio config to load module sctp in the loadmodules section:

loadmodule ""

Now we’ll start listening on SCTP, so where your current listen= entries are we’ll add one:


I’ve loaded Dispatcher for this example, and we’ll add a new entry to Dispatcher so we can ping ourselves.

We’ll use kamctl to add a new dispatcher entry of our loopback IP ( but using SCTP as the transport.

kamctl dispatcher add 1 'sip:;transport=sctp' 0 0 '' 'Myself SCTP'

Now I’ll restart Kamailio and check kamcmd:

kamcmd dispatcher.list

All going well you’ll see the entry as up in Dispatcher:

And firing up tcpdump should show you that sweet SCTP traffic:

tcpdump -i lo -n sctp

Sadly by default TCPdump doesn’t show our SIP packets as they’re in SCTP, you can still view this in Wireshark though:

Here’s a copy of the packet capture I took:

I’ve put a copy of my basic config on GitHub.

Now get out there and put SCTP into the real world!

Kamailio Bytes – Setting up rtpengine in Kamailio to relay RTP / Media

In an ideal world all media would go direct from one endpoint to another.

But it’s not an ideal world and relaying RTP / media streams is as much a necessary evil as transcoding and NAT in the real world.

The Setup

We’ll assume you’ve already got a rtpengine instance on your local machine running, if you don’t check out my previous post on installation & setup.

We’ll need to load the rtpengine module and set it’s parameters, luckily that’s two lines in our Kamailio file:

loadmodule ""
modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223")

Now we’ll restart Kamailio and use kamcmd to check the status of our rtpengine instance:

kamcmd all

All going well you’ll see something like this showing your instance:

Putting it into Practice

If you’ve ever had experience with the other RTP proxies out there you’ll know you’ve had to offer, rewrite SDP and accept the streams in Kamailio.

Luckily rtpengine makes this a bit easier, we need to call rtpengine_manage(); when the initial INVITE is sent and when a response is received with SDP (Like a 200 OK).

So for calling on the INVITE I’ve done it in the route[relay] route which I’m using:

And for the reply I’ve simply put a conditional in the onreply_route[MANAGE_REPLY] for if it has SDP:

onreply_route[MANAGE_REPLY] {
        xdbg("incoming reply\n");
        if(status=~"[12][0-9][0-9]") {


And that’s it, now our calls will get RTP relayed through our Kamailio box.

Advanced Usage

There’s a bunch of more cool features you can use rtpengine for than just relay, for example:

  • IPv4 <-> IPv6 translation for Media
  • ICE Bridging
  • SRTP / Encrypted RTP to clear RTP bridging
  • Transcoding
  • Repacketization
  • Media Playback
  • Call Recording

I’ll cover some of these in future posts.

Here’s a copy of my running config on GitHub.

For more in-depth info on the workings of RTP check out my post RTP – More than you wanted to Know

Magma – Facebook’s Open Source LTE / 4G EPC/OSS Platform

In February Facebook announced they’d open sourced their Magma project,

Magma provides a software-centric distributed mobile packet core and tools for automating network management.

Open-sourcing Magma to extend mobile networks

Magma’s modular software based architecture means you can scale up extra resources as needed, with no need to have physical hardware to run your EPC.

(Cisco’s Ultra Packet Core does have a virtualisation option, but it’s not cheap)

I got pretty excited by this, so I’ve ordered myself an eNodeB (Just a Picocell), a pile of USIMs, programmer and started installing an environment.

In the past I’ve used srsEPC and NextEPC and software-defined radio hardware (BladeRF) to run LTE stuff, so I’m looking forward to seeing if I can implement parts of them into Magma, and also eventually use Kamailio’s IMS modules to implement an IMS core and run VoLTE.

So let’s install Magma, explore it and lurk on the Discord, all while we kill time waiting for hardware to arrive!

SNgrep – Command line SIP Debugging

If you, like me, spend a lot of time looking at SIP logs, sngrep is an awesome tool for debugging on remote machines. It’s kind of like if VoIP Monitor was ported back to the days of mainframes & minimal remote terminal GUIs.


It’s in the Repos for Debian and Ubuntu:

apt-get install sngrep

GUI Usage

sngrep can be used to parse packet captures and create packet captures by capturing off an interface, and view them at the same time.

We’ll start by just calling sngrep on a box with some SIP traffic, and waiting to see the dialogs appear.

Here we can see some dialogs, two REGISTERs and 4 INVITEs.

By using the up and down arrow keys we can select a dialog, hitting Enter (Return) will allow us to view that dialog in more detail:

Again we can use the up and down arrow keys to view each of the responses / messages in the dialog.

Hitting Enter again will show you that message in full screen, and hitting Escape will bring you back to the first screen.

From the home screen you can filter with F7, to find the dialog you’re interested in.

Command Line Parameters

One of the best features about sngrep is that you can capture and view at the same time.

As a long time user of TCPdump, I’d been faced with two options, capture the packets, download them, view them and look for what I’m after, or view it live with a pile of chained grep statements and hope to see what I want.

By adding -O filename.pcap to sngrep you can capture to a packet capture and view at the same time.

You can use expression matching to match only specific dialogs.

Kamailio Bytes – Permissions Module

Kamailio’s permissions module is simple to use, and we’ve already touched upon it in the security section in our Kamailio 101 series, but I thought I’d go over some of it’s features in more detail.

At it’s core, Kamailio’s Permissions module is a series of Access Control Lists (ACLs) that can be applied to different sections of your config.

We can manage permissions to do with call routing, for example, is that source allowed to route to that destination.

We can manage registration permissions, for example, is this subnet allowed to register this username.

We can manage URI permissions & address permissions to check if a specific SIP URI or source address is allowed to do something.

We’ll touch on a simple IP Address based ACL setup in this post, but you can find more information in the module documentation itself.

The Setup

We’ll be using a database backend for this (MySQL), setup the usual way.

We’ll need to load the permissions module and setup it’s basic parameters, for more info on setting up the database side of things have a look here.

loadmodule ""
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)

Next we’ll need to add some IPs, we could use Serimis for this, or a straight MySQL INSERT, but we’ll use kamctl to add them. (kamcmd can reload addresses but doesn’t currently have the functionality to add them)

kamctl address add 250 32 5060 TestServer
kamctl address add 200 24 5060 OfficeSubnet

The above example we added a two new address entries,

The first one added a new entry in group 250 of “”, with a /32 subnet mask (Single IP), on port 5060 with the label “TestServer”,

The second one we added to group 200 was a subnet of with a /24 subnet mask (255 IPs), on port 5060 with the label “OfficeSubnet”

On startup, or when we manually reload the addressTable, Kamailio grabs all the records and stores them in RAM. This makes lookup super fast, but the tradeoff is you have to load the entries, so changes aren’t immediate.

Let’s use Kamcmd to reload the entries and check their status.

kamcmd permissions.addressReload

kamcmd permissions.addressDump

kamcmd permissions.subnetDump

You should see the single IP in the output of the permissions.addressDump and see the subnet on the subnetDump:


It’s usage is pretty simple, combined with a simple nested if statement.

if (allow_source_address("200")) {
	xlog("Coming from address group 200");
if (allow_source_address("250")) {
	xlog("Coming from address group 250");

The above example just outputs to xlog with the address group, but we can expand upon this to give us our ACL service.

if (allow_source_address("200")) {
	xlog("Coming from address group 200");
}else if (allow_source_address("250")) {
	xlog("Coming from address group 250");
        sl_reply("401", "Address not authorised");

If we put this at the top of our Kamailio config we’ll reply with a 401 response to any traffic not in address group 200 or 250.

Kamailio Bytes – Dialplan Module

Kamalio’s dialplan is a bit of a misleading title, as it can do so much more than just act as a dialplan.

At it’s core, it runs transformations. You feed it a value, if the value matches the regex Kamailio has it can either apply a transformation to that value or return a different value.

Adding to Config

For now we’ll just load the dialplan module and point it at our DBURL variable:

loadmodule ""
modparam("dialplan", "db_url", DBURL);                 #Dialplan database from DBURL variable

Restart Kamailio and we can get started.


Let’s say we want to take StringA and translate it in the dialplan module to StringB, so we’d add an entry to the database in the dialplan table, to take StringA and replace it with StringB.

We’ll go through the contents of the database in more detail later in the post

Now we’ll fire up Kamailio, open kamcmd and reload the dialplan, and dump out the entries in Dialplan ID 1:

dialplan.dump 1

You should see the output of what we just put into the database reflected in kamcmd:

Now we can test our dialplan translations, using Kamcmd again.

dialplan.translate 1 StringA

All going well Kamailio will match StringA and return StringB:

So we can see when we feed in String A, to dialplan ID 1, we get String B returned.

Database Structure

There’s a few fields in the database we populated, let’s talk about what each one does.


dpid = Dialplan ID. This means we can have multiple dialplans, each with a unique dialplan ID. When testing we’ll always need to specific the dialplan ID we’re using to make sure we’re testing with the right rules.


Priorities in the dialplan allow us to have different weighted priorities. For example we might want a match all wildcard entry, but more specific entries with lower values. We don’t want to match our wildcard failover entry if there’s a more specific match, so we use priorities to run through the list, first we try and match the group with the lowest number, then the next lowest and so on, until a match is found.


match_op = Match Operation. There are 3 options:

  • 0 – string comparison;
  • 1 – regular expression matching (pcre);
  • 2 – fnmatch (shell-like pattern) matching

In our first example we had match_op set to 0, so we exactly matched “StringA”. The real power comes from Regex Matching, which we’ll cover soon.


match_exp = Match expression. When match_op is set to 0 this matches exactly the string in match_exp, when match_op is set to 1 this will contain a regular expression to match.


match_len = Match Length. Allows you to match a specific length of string.


subst_exp = Substitute Expression. If match_op is set to 0 this will contain be empty If match_op is 1 this will contain the same as match_exp.


repl_exp = replacement expression. If match_op is set to 0 this will contain the string to replace the matched string.

If match_op is set to 1 this can contain the regex group matching (\1, \2, etc) and any suffixes / prefixes (for example 61\1 will prefix 61 and add the contents of matched group 1).


Attributes. Often used as a descriptive name for the matched rule.

Getting Regex Rules Setup

The real power of the dialplan comes from Regular Expression matching. Let’s look at some use cases and how to solve them with Dialplans.

Note for MySQL users: MySQL treats \ as the escape character, but we need it for things like matching a digit in Regex (that’s \d ) – So keep in mind when inserting this into MySQL you may need to escale the escape, so to enter \d into the match_exp field in MySQL you’d enter \\d – This has caught me in the past!

The hyperlinks below take you to the examples in so you can preview the rules and make sure it’s matching what it should prior to putting it into the database.

Speed Dial

Let’s start with a simple example of a speed dial. When a user dials 101 we want to translate it to a PSTN number of 0212341234.

Without Regex this looks very similar to our first example, we’ve just changed the dialplan id (dpid) and the match_op and repl_exp.

Once we’ve added it to the database we’ll reload the dialplan module and dump dialplan 2 to check it all looks correct:

Now let’s test what happens if we do a dialplan translate on dialplan 2 with 101.

Tip: If you’re testing a dialplan and what you’re matching is a number, add s: before it so it matches as a number, not a string.

dialplan.translate 2 s:101

Here we can see we’ve matched 101 and the output is the PSTN number we wanted to translate too.

Interoffice Dial

Let’s take a slightly more complex example. We’ve got an office with two branches, office A’s phone numbers start with 0299991000, and they have 4 digit extensions, so extension 1002 maps to 0299991002, 0299991003 maps to extension 1003, etc.

From Office B we want to be able to just dial the 4 digit extensions of a user in Office A.

This means if we receive 1003 we need to prefix
029999 + 10003.

We’ll use Regular Expressions to achieve this.

We can use a simple Regular Expression to match any number starting with 1 with 3 digits after it.

But the problem here is we want to collect the output into a Regex Group, and then prefix 029999 and the output of that group.

So let’s match it using a group.


So let’s put this into the database and prefix everything in matching group 1 with 029999.

We’ll use dialplan ID 3 to separate it from the others, and we’ll set match_op to 1 to use Regex.

As you can see in repl_exp we’ve got our prefix and then \1.

\1 just means the contents of regex matching group 1.

After running dialplan reload let’s try this one out:

dialplan.translate 3 s:1003

We tested with 1003, but we could use 1000 through to 1999 and all would match.

But if we’ve only got a 100 number range (0299991000 to
0299991099) we’ll only want to match the first 100 numbers, so let’s tweak our regex to only allow the first two digits to be wildcards.


Now let’s update the database:

Then another reload and translate, and we can test again.

dialplan.translate 3 s:1003 (Translates to 0299991003)
dialplan.translate 3 s:1101 (no translation)

Interoffice Dial Failure Route (Priorities)

So let’s say we’ve got lots of branches configured like this, and we don’t want to just get “No Translation” if a match isn’t found, but rather send it to a specific destination, say reception on extension 9000.

So we’ll keep using dpid 3 and we’ll set all our interoffice dial rules to have priority 1, and we’ll create a new entry to match anything 4 digits long and route it to the switch.

This entry will have a higher priority value than the other so will only mach if nothing else with a lower priority number matches.

We’ll use this simple regex to match anything 4 digits long into group 1.


Now let’s run through some test again.

dialplan.translate 3 s:1003 (Translates to 0299991003)
dialplan.translate 3 s:1101 (Translates to 9000 (Attributes: Interoffice Dial - Backup to Reception)

Translate 0NSN to E.164 format numbers

Let’s say we’ve got a local 10 digit number. In 0NSN format it looks like 0399999999 but we want it in E.164 so it looks like 613999999999.

Let’s use Kamailio to translate this from 0NSN to E.164.

The first thing we’ll need to do is create a regular expression to match

We’ll match anything starting with 03, with 9 digits after the 0 matched in Group 2.


Now we’ve got Group 2 containing the data we need, we just need to prefix 613 in front of it.

Let’s go ahead an put this into the database, with dialplan ID set to 4, match_op set to 1 (for regex)

Then we’ll do a dialplan reload and a dialplan dump for dialplan ID 4 to check everything is there:

Now let’s put it to the test.

dialplan.translate 4 s:0399999999

Bingo, we’ve matched the regex, and returned 613 and the output of Regex Match group 2. (999999999)

Let’s expand upon this a bit, a valid 0NSN number could also be a mobile (0400000000) or a local number in a different area code (0299999999, 0799999999 or 0899999999).

We could create a dialplan entry for each, our we could expand upon our regex to match all these scenarios.

So let’s update our regex to match anything starting with 0 followed by either a 2, 3, 4, 7 or 8, and then 8 digits after that. 


Now let’s update the database so that once we’re matched we’ll just prefix 61 and the output of regex group 2.

Again we’ll do a dialplan reload and a dialplan dump to check everything.

Now let’s run through our examples to check they correctly translate:

And there you go, we’re matched and the 0NSN formatted number was translated to E.164.

Adding to Kamailio Routing

So far we’ve just used kamcmd’s dialplan.translate function to test our dialplan rules, now let’s actually put them into play.

For this we’ll use the function

dp_translate(id, [src[/dest]])

dp_translate is dialplan translate. We’ll feed it the dialplan id (id) and a source variable and destination variable. The source variable is the equivalent of what we put into our kamcmd dialplan.translate, and the destination is the output.

In this example we’ll rewrite the Request URI which is in variable $rU, we’ll take the output of $rU, feed it through dialplan translate and save the output as $rU (overwrite it).

Let’s start with the Speed Dial example we setup earlier, and put that into play.

                xlog("rU before dialplan translation is $rU");
                dp_translate("2", "$rU/$rU");
                xlog("rU after dialplan translation is $rU");

The above example will output our $rU variable before and after the translation, and we’re using Dialplan ID 2, which we used for our speed dial example.

So let’s send an INVITE from our Softphone to our Kamailio instance with to 101, which will be translated to 0212341234.

Before we do we can check it with Kamcmd to see what output we expect:

dialplan.translate 2 s:101

Let’s take a look at the out put of Syslog when we call 101.

But our INVITE doesn’t actually go anywhere, so we’ll add it to our dispatcher example from the other day so you can see it in action, we’ll relay the INVITE to an active Media Gateway, but the $rU will change.

                xlog("rU before dialplan translation is $rU");
                dp_translate("2", "$rU/$rU");
                xlog("rU after dialplan translation is $rU");
                ds_select_dst(1, 12);

Let’s take a look at how the packet captures now look:

UA > Kamailio: INVITE sip:101@kamailio SIP/2.0
Kamailio > UA: SIP/2.0 100 trying -- your call is important to us
Kamailio > MG1: INVITE sip:0212341234@MG1 SIP/2.0

So as you can see we translated 101 to 0212341234 based on the info in dialplan id 2 in the database.

That’s all well and good if we dial 101, but what if we dial 102, there’s no entry in the database for 102, as we see if we try it in Kamcmd:

dialplan.translate 2 s102

And if we make a call to 102 and check syslog:

rU before dialplan translation is 102
rU after dialplan translation is 102

Let’s setup some logic so we’ll respond with a 404 “Not found in Dialplan” response if the dialplan lookup doesn’t return a result:

if(dp_translate("2", "$rU/$rU")){
  xlog("Successfully translated rU to $rU using dialplan ID 2");
  xlog("Failed to translate rU using dialplan ID 2");
  sl_reply("404", "Not found in dialplan");

By putting dp_translate inside an if we’re saying “if dp_translate is successful then do {} and the else will be called if dp_translate wasn’t successful.

Let’s take a look at a call to 101 again.

UA > Kamailio: INVITE sip:101@kamailio SIP/2.0
Kamailio > UA: SIP/2.0 100 trying -- your call is important to us
Kamailio > MG1: INVITE sip:0212341234@MG1 SIP/2.0

Still works, and a call to 102 (which we don’t have an entry for in the dialplan).

UA > Kamailio: INVITE sip:102@kamailio SIP/2.0
Kamailio > UA: SIP/2.0 404 Not found in dialplan

Hopefully by now you’ve got a feel for the dialplan module, how to set it up, debug it, and use it.

As always I’ve put my working code on GitHub.

ASN.1 Encoding in a Nutshell

What is ASN.1 and why is it so hard to find a good explanation or example?

ISO, IEC & ITU-T all got together and wrote a standard for describing data transmitted by telecommunications protocols, it’s used by many well known protocols X.509 (SSL), LDAP, SNMP, LTE which all rely on ASN.1 to encode data, transmit it, and then decode it, reliably and efficiently.


Let’s take this XML encoded data:

<?xml version="1.0" encoding="UTF-8"?>
  <text>Hello my friend!</text>

As you can see it’s human readable and pretty clear.

But what if we split this in two, had the definitions in one file and the values in another:


Here we’ll describe each of our fields

NumberĀ – Intiger- Destination of Message
Text – String – Message to be sent


Now we’ll list the values.

Hello my friend!

By taking the definitions out our data is now 28 bytes, instead of 122, so we’re a fraction of the size on the wire (becomes important if you’re sending this data all the time or with a limited link budget), and we’ve also defined the type of each value as well, so we know we shouldn’t have an integer as the heading for example, we can see it’s a string.

The sender and the receiver both have a copy of the definitions, so everyone is clear on where we stand in terms of what each field is, and the types of values we encode. As a bonus we’re down to less than 1/4 of our original size. Great!

That’s ASN.1 in a nutshell, but let’s dig a little deeper and use a real example.


Now let’s actually encode & decode some data.

I’ll be using asn1tools a Python library written by Erik Moqvist, you can add it through pip:

pip install asn1tools

We’ll create a new text file and put our ASN.1 definitions into it, so let’s create a new file called foo.asn which will contain our definitions in ASN.1 format:

Message ::= SEQUENCE {
number INTEGER,
text UTF8String

Copy and paste that into foo.asn and now we’ve got a definition, with a Module called Message containing a field called number (which is an integer) and a filed called text (which is a string).

Now let’s fire up our python shell in the same directory as our new file:

>>import asn1tools
>>foo = asn1tools.compile_files('foo.asn')
>>encoded = foo.encode('Message',  {'number': 61412341234, 'text': u'Hello my friend!'})
    bytearray(b'0\x19\x02\x05\x0eLu\xf5\xf2\x0c\x10Hello my friend!')

(You’ll need to run this in the Python shell, else it’ll just output encoded as plain text and not as a Byte Array as above)

So now we’ve encoded our values (number = 2 and text = ‘Hi!’) into bytes, ready to be sent down the wire and decoded at the other end. Not exactly human friendly but efficient and well defined.

So let’s decode them, again, Python shell:

>>> import asn1tools
>>> foo = asn1tools.compile_files('foo.asn')
>>> decoded = foo.decode('Message', '0\x19\x02\x05\x0eLu\xf5\xf2\x0c\x10Hello my friend!')
>>> decoded
    {'text': u'Hello my friend!', 'number': 61412341234}

So there you have it, an introduction to ASN1, how to encode & decode data.

We can grow our definitions (in the .asn file) and so long as both ends have the same definitions and you’re encoding the right stuff, you’ll be set.

Further Reading

There’s a whole lot more to ASN1 – Like how you encode the data, how to properly setup your definitions etc, but hopefully you understand what it actually does now.

Some further reading: