The Palner Group, Inc.

Kamailio, Asterisk, VoIP, and IT Consulting

Category: VoIP (page 1 of 4)

Fortune 25 Open Source Telephony

Open source telephony, such as Kamailio, Asterisk, and FreeSWITCH provide any company the ability to control their communications. This statement holds true whether you happen to be a small, family business or one of the most successful companies in the world.

Today LOD Communications, Inc. began work on a new 1-year contract with a Fortune 25 company to provide communications systems design, consulting, and technical support services (NDA prevents disclosing name) utilizing open source software.

Regardless of size, every company can benefit from using software such as Kamailio. For more information on how The Palner Group and LOD can assist you, please give us a call at +1 (212) 937-7844.

CSRP: High Performance Kamailio Tuning

Our friends at CSRP have written a great discussion on high performance turning for Kamailio.

When you’re running decent call volume, tuning Kamailio for high throughput and high performance is essential to success.

We’ve often recommended the CSRP solution, and this article is a great reason why we continue to so.

Tuning Kamailio for High Performance and High Throughput (CSRP)

Palner Group, Asterisk World, ITEXPO

Join Fred Posner, Director of The Palner Group, Inc., in Fort Lauderdale as he presents Expanding Asterisk with Kamailio at the 2016 Asterisk World.

Held in beautiful Fort Lauderdale, Florida, Asterisk World at ITEXPO showcases Asterisk (the open source pbx) and how it can assist your company with telecommunications.

Fred Posner, a Kamailio/Asterisk consultant, has more than ten years experience with both Kamailio and Asterisk; working to assist companies worldwide with their communication needs.

From call centers to business phone systems, both Asterisk and Kamailio provide an incredible opportunity for your company to remain competitive in today’s market without being tied into arbitrary licensing systems and fees.

For more information, please see Fred at Asterisk World or contact us today.

Why Fred Loves Kamailio

At The Palner Group, we specialize in deploying open source VoIP solutions. Kamailio SIP Server is one of the greatest tools in our possession. Our customers love the product, and more importantly, Kamilio based solutions are reliable and affordable.

Recently, Fred Posner (Director of The Palner Group), penned Why I Love Kamailio on his personal blog.

It’s a great read — of course, we may be biased.

Kamailio Behind NAT

Fred Posner wrote this article describing running Kamailio behind NAT. Printed with permission.

After returning home from AstriCon 10, I decided to start-up a new server and see how long it would take me to run a working Kamailio server behind NAT (on a private IP). Bottom line? About 30 minutes.

I was lucky to help staff the Kamailio booth this year and was honored to help so many other VoIP professionals (and enthusiasts) with their questions about Kamailio, so without further ado… Continue reading

LOD Communications partners with Team Forrest

FORT LAUDERDALE, FLORIDA — The Palner Group, Inc. d/b/a Team Forrest partners with LOD Communications to provide clients excellent service and support.

LOD Communications, Inc. ( has been providing quality UNIX, Linux and Cisco consulting services to small and medium sized businesses throughout the United States since 1990.

The Palner Group has been provided quality VoIP and IT consulting service since 2000.

For many years, The Palner Group, Inc has had a great working relationship with the LOD team. With overlapping client needs, the time seemed appropriate for me to partner with LOD to provide my clients additional support and services.Fred Posner, Director

Generally, new Team Forrest customers are billed as LOD clients. Posner added:

LOD an amazing history of billing the right price for the right service. Now, Team Forrest can focus on providing excellent service and focus less on billing and administrative tasks. Some of our older customers are still handled internally, but most of our clients have already been transitioned.

The immediate benefit to clients included:

  • Access to the LOD cloud infrastructure
    Dedicated hosting facilities provide clients with geographic redundancy and high speed data connectivity
  • Access to LOD expertise
    Linux, Cisco, Unix, Solaris, and more
  • 24/7 Support
    When you need LOD or The Palner Group, help is just a phone call away. Clients with SLA packages received guaranteed response times 24/7/365.

VoIP Services / Hosted VoIP

The best news of the partnership rested in the creation of the LOD hosted VoIP service. Powered by Kamailio (formerly openSER), the LOD hosted VoIP offering provides reliable communication, robust features, and a reasonable price.

For more information on how LOD and The Palner Group, Inc. can assist your business with their communication needs, please contact us or visit

Phishing Attack Targets Florida First Credit Union

Team Forrest owner Fred Posner recently posted a piece regarding a phishing attack he received on his cell phone:

We’ll see. With luck, most people will realize this is a scam and it will fail. That being said, the script was well written. Social engineering is very powerful when done correctly. The writing on this attack was excellent.

Read the full post regarding this phishing attack on Fred’s website:

Explaining Sip Brute Force Attacks to Non-Techs

Today we received a call from a federal employee investigating a “hack” on a client’s system. Basically, the client suffered a SIP Brute Force attack on their elastix system. Besides the shock of a call from the feds (why did they ignore those Amazon attacks?), the realization of explaining a sip attack to someone not familiar with SIP, telephony, networking, or servers posed a little challenge.

So, how do we start?

First step: We will no longer use the words SIP, Brute, Force, and Attack. =)

What we’re talking about is a scheme to make expensive calls through your phone system. Of course, this isn’t true for all scenarios, but the vast majority simply want to make expensive calls on your dime.

How does it work?

The bad guys trick your phone system into thinking they are a valid user.

How can they do that?

When phones connect to your phone system, the system replies with different messages. Based on those messages, the bad guys can figure out phone names. Think of your phone system as the receptionist. An attempt might be similar to…

Bad Guy: “Hi, is Alice there?”
Receptionist: “No, there is no Alice here. You have the wrong number.”
Bad Guy: “Hi, is Bob there?”
Receptionist: “Yes, who may I say is calling?”

Basically, there’s a different response based on if that person exists in the company. Same thing with the phones. Once the Bad Guys find out phone names, they then use their computers to crack the phone password.

Once the password is detected, they connect their phone to your system and begin making calls.

What can I do to stop this?

If the person in charge of your phone system doesn’t understand what this attack is, you need to hire a consultant to help you and/or train your administrator. If you or your administrator understand this attack, then you need to make sure you are following the best practices for SIP security (here’s a good link for asterisk best practices).

If you’re running asterisk, you might wish to install a script that checks for attacks and blocks those connections.

Even better… consider Kamailio.

Kamailio (pronounced KAMA-ILLY-OH) is an open-source SIP proxy, registrar, application that is extremely robust and powerful. The software includes anti-flood features that really help protect your system and truly helps to minimize these annoying attacks.

Remember, the Internet is like a big city. Sure there’s great museums and entertainment, but there’s also bad, bad places filled with bad, bad people. If you’re going to leave your BMW unlocked in Hell’s Kitchen, don’t be surprised when it’s been taken around the block a few times.

Team Forrest assists Dream Day Cakes

GAINESVILLE, FL — Team Forrest recently assisted Dream Day Cakes with phone and intranet services. The Gainesville, Florida bakery specializes in wedding cakes, event cakes, and other bakery items benefited from a hybrid VoIP / Analog phone system.

Continue reading

Use ENUM to Save Real MONey

Ok — it almost rhymed.

ENUM (read the wiki) refers to the mapping of telephone numbers to internet addresses. Think of it almost as reverse DNS for your phone number. Although there are many methods of integrating ENUM into your system, our current “favorite” is

From their website:

ENUM sources are very segregated and there was no global repository – until now. ENUMPlus queries all of the top ENUM lookup sources and returns the most accurate result with minimal overhead; meaning you only need to specify one source. ENUMPlus allows you to offload all of the query processing to our powerful servers so you don’t have to waste time and precious resources.

Integrating ENUMplus into Asterisk can be very quick and there’s a few choices/methods of going about it. You can choose to use their php scripts, go direct from the dialplan, or run your own lookup script. Here, we’ve chosen to write our own lookup script that basically does the following:

  1. Checks for a result (with a 2 second timeout)
  2. Sets a variable of ENUMRESULT and returns to dialplan
  3. The dialplan then evaluates that variable, and if a sip value is provided calls the number directly via SIP.

Here’s an example dialplan:

exten => _X.,1,Set(CALLTO=${EXTEN})
exten => _X.,n,Goto(out,1)
exten => out,1,AGI(,${CALLTO})
exten => out,n,GotoIf($["${ENUMRESULT}" = "FAIL"]?pstn)
exten => out,n,GotoIf($[${ISNULL(${ENUMRESULT})}]?pstn)
exten => out,n,Dial(${ENUMRESULT},55)
exten => out,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL" ]?pstn)
exten => out,n,GotoIf($["${DIALSTATUS}" = "CONGESTION" ]?pstn)
exten => out,n,GotoIf($["${DIALSTATUS}" = "BUSY" ]?busy)
exten => out,n,Hangup()
exten => out,n(pstn),Dial(SIP/${CALLTO}@yourprovider); or DAHDI, etc.
exten => out,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL" ]?busy)
exten => out,n,GotoIf($["${DIALSTATUS}" = "CONGESTION" ]?busy)
exten => out,n,GotoIf($["${DIALSTATUS}" = "BUSY" ]?busy)
exten => out,n,Hangup()
exten => out,n(busy),Busy(5)
exten => out,n,Hangup()

And here’s the script:

#!/usr/bin/perl -w
use strict;
my ($phone, $url, $apikey, $result, @sip);

while(<STDIN>) {
	last unless length($_);

if ($ARGV[0]) {
	$phone = &URLEncode($ARGV[0]);
} else {
	&setvar("ENUMRESULT", "FAIL");
	&printverbose("enumlookup: No CALLTO received.",2);

#Get via WEB
$url = "";

$result = qx(curl -m 2 -s -d 'key=$apikey' $url/$phone);

if ($result) {
	if ($result =~ /SIP/i) {
		@sip = split(/\|/, $result);
		&setvar("ENUMRESULT", $sip[0]);
		&printverbose("enumlookup: $sip[0]",2);
	} else {
		&setvar("ENUMRESULT", "FAIL");
		&printverbose("enumlookup: No sip address found.",2);
} else {
	&setvar("ENUMRESULT", "FAIL");
	&printverbose("enumlookup: Timeout or error",2);

sub URLEncode {
   my $theURL = $_[0];
   $theURL =~ s/([\W])/"%" . uc(sprintf("%2.2x",ord($1)))/eg;
   return $theURL;

sub setvar {
	my ($var, $val) = @_;
	print STDOUT "SET VARIABLE $var \"$val\" \n";
	while(<STDIN>) {
		m/200 result=1/ && last;

sub printverbose {
	my ($var, $val) = @_;
	print STDOUT "VERBOSE \"$var\" $val\n";
	while(<STDIN>) {
		m/200 result=1/ && last;

Happy Coding!

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