Regular Expressions For SIP Trunk



The regular expression is a sequence of characters that form a search pattern. See the definition in Wikipedia. It is a huge topic and takes a lot of time to explain. If you didn’t use it, you will get stuck in a nested “If” statements situation. Using regular expressions means professionalism. I am assuming you know about it and i will just apply it to SIP.


Regular Expressions For a SIP Trunk

Usually when a SIP router receives a SIP request which is addressed to a PSTN gateway (i.e. Its Request-URI contains a telephone number), it checks which group the requested user (called party) belongs to. The router also must check if the caller has permission to make this call. So the check can be done on the caller and calle. Any check done by the router is translated into this simple thinking logic:

If (Routing Condition){

  Do Specific Work


This can be in the router’s routing script or internally integrated in the router’s application.


If (The Call Is Addressed To A Local Number){

Do Specific Work


In OpenSIPS SIP router, the previous condition can be treated in the routing scripts in two ways:

  • Using the Script Variables (e.g. the reference to the request’s uri ($ru)) with the regular expression directly:

If ($ru=~sip:[2-9][0-9]{6}@){


The previous regular expression is for any 7 numbers starting by one digit from the range [2-9].

  • Using a function like “pcre_match” in “Regex” module which matches the given string against a given regular expression. The return value is TRUE if it is matched, FALSE otherwise:

If (pcre_match(“$ru”,sip:[2-9][0-9]{6}@){


The variable “$ru” is read/write variable so be aware where in the script you are checking its value. You can use “$ou” which is a reference to the request’s original URI. The module “Regex” is based on the library “PCRE” which is an implementation of regular expression pattern matching where the regular expression parameter will be compiled in PCRE object. So the development libraries of “PCRE” must be installed (“libpcre-dev” or “pcre-devel”).

Note: Use “&&” if you want to concatenate multiple conditions in one “if” statement.

If ((Routing Condition-1) && (Routing Condition-2)){

  Do Specific Work


Examples of Regular Expressions For SIP Trunk

You should test your regular expression before using it to know if you have constructed it correctly. There are many online free services that you can use to do your tests. Use search engines to search for syntax symbols that you can use to construct the regular expression. The following are examples. Note that each one defines a group:

  • Any user on any domain/IP: sip:(.*)@(.*)
  • Any user on certain domain: sip:(.*)
  • Any user on certain IP (  sip:(.*)@198\.18\.250\.10 In regular expression, the “.” is interpreted as “any character” symbol whereas “\.” is just a period (dot).
  • Any user on IP Range ( – sip:(.*)@198\.18\.250.* The symbol “*” is quantifier which means the preceding character is found 0 or more times. Here i left the last part of IP to be anything but you can restrict it.
  • To group a set of SIP URIs that are within a certain domain or certain subdomain, use .*mydomain\.com.* For example these URIs will be matched:;transport=tcp, and SIP/2.0
  • To group a set of SIP URIs that have certain string in the username part of URI and the ports 5060 and 5061 are accepted, use the regular expression .*group1@198\.18\.250.\10:506<01>.* For example these two URIs will be matched sip:serv1group1@;transport=tcp and sips:serv2group1@;tranport=tcp
  • 8-digit number on any domain: sip:[0-9]{8}@(.*) The [0-9] is a range for one number between 0 and 9 and {8} means repeat the preceding number 8 times.
  • 8-digit number starting optionally by 8 on any domain: sip:8?[0-9]{7}@(.*)  The symbol “?” means the repetition is 0 or 1 to the preceding. For example,,, and so on.
  • 4-digit number (could be an extension number) starting by 6 on a certain domain: sip:6[0-9]{3}@(.*)
  • 4-digit number which is not starting by 55 on any domain: sip:(?!55)[0-9]{4}@(.*)

 More Information

OpenSIPS Provisioning for a simple SIP Call

It is very easy to configure and manage OpenSIPS through OpenSIPS-CP.

Add the domains that OpenSIPS is responsible for

Lets add any domain (like just for test). click on System menu –> Domains


Add the users

Now we can add users to that domain. click on “Users” –> “User Management” snapshot9

User1: (username:test1, password:1234, domain:, User2: (username: test2, password:1234, domain:


Register the users and make the call (user2 –> user1)

Now we have two users: user1 and user2 and we want test2 to call test1.  Users must be registered first from SIP clients installed by the users.  Here you can find a list of open source and free SIP clients: List of Free and Open-Source SIP Software.  I recommend Jitsi,  Linphone, and CSipSimple.

Now i am using Jitsi for test1 on my laptop and i have entered the following account information in Jitsi interface:

( SIP id: test1@, password: 1234, Proxy:, Proxy Port: 5060)

And now Click på “Sign in” and the user test1 is now “online”.


I will make test2 register itself from my mobile where i have downloaded CSipSimple SIP client on it. My mobile phone and my laptop are in the same LAN. The laptop has the OpenSIPS and local SIP client (Jitsi). The laptop is protected by firewall.

Firewall rules

Everything is OK but test2 (on mobile phone) can not register itself. WHY because the firewall prevents this access. So i have to let the firewall allows SIP traffic to come to the OpenSIPS. We need to open the ports that the OpenSIPS is listening on. In my case the UDP ports 5060-5061.  To do this use dynamic firewall with Firewalld  or static firewall (system-config-firewall) with iptables.  Using static firewall means disabling and stopping of dynamic firewall.

For Firewalld you can either use the graphical configuration tool firewall-config or the command line client firewall-cmd. More information is here Firewalld

More information about editing iptables rules is here: Iptables

Note: Be aware of these “enable”, “disable”, “start”,  “stop” when using “systemctl” command.

“opensipsctl” Tool to see the Registered Users

To take information about the regitered users.

# cd /usr/local/opensips_1_11/sbin/

# ./opensipsctl fifo ul_dump
And this is the output:

AOR:: test2
Contact:: sip:test2@;ob Q=
Expires:: 894
Callid:: Vd8CYS9wJO7ffGoNAembj-bnC5Rte-k3
Cseq:: 55430
User-agent:: CSipSimple_MT15i-10/r2330
State:: CS_SYNC
Flags:: 0
Socket:: udp:
Methods:: 8063
AOR:: test1
Contact:: sip:test1@;transport=udp;registering_acc=10_0_0_4 Q=
Expires:: 521
Callid:: 3586ed724fb54b2e7505f24627e93ff6@0:0:0:0:0:0:0:0
Cseq:: 36
User-agent:: Jitsi2.2.4603.9615Linux
State:: CS_SYNC
Flags:: 0
Socket:: udp:
Methods:: 4294967295

You can also do this (# ./opensipsctl ul show). Now test1 is calling test2 :

Jitsi36- Get the Dialog Information from the OpenSIPS-CP Web Interface

During the call establishment, click on “System” menu –> “Dialog”. See the figures below of dialog early state and confirmed state.



What will be different if we want to deploy such a system ?

The answer is the existence of NAT.  I will explain this problem and how to solve it later.