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Configuring Asterisk 13

Reference Number: AA-02067 Views: 10638 0 Rating/ Voters

Asterisk 13 + UniMRCP 1.3.0 + LumenVox 13.1

We have many customers running Asterisk PBX using our speech services, and these work very well together, however we often hear of users running into difficultly installing and configuring Asterisk or UniMRCP before they even have a chance to set up the LumenVox services.

In this article, and the accompanying video series, we will go through the installation and configuration of Asterisk 13, UniMRCP 1.3.0 and LumenVox 13.1 by way of a practical example to show the various steps involved.

Please check out the accompanying Asterisk 13 Installation and Configuration whitepaper and video series for this article on our LVDN site.

Before we begin, let's go over the basic structure - along with some general assumptions regarding the installation approach :

  • We will be starting with a fresh installation of CentOS 6 64-bit, however the process would be virtually identical for CentOS or RHEL 5 or 7 if that is your preference.
  • We will be following Justin Hester's wonderful blog at : that describes installing PJSIP and Asterisk 13
  • Asterisk 13 will be using a new library called PJSIP, so the PJSIP library will need to be installed prior to asterisk. PJSIP is the new channel library for Asterisk, replacing the older DAHDI and LIBPRI drivers.
  • We'll be installing UniMRCP 1.3.0
  • We'll be installing LumenVox 13.1, although the steps would be virtually identical for any version of LumenVox, since we try to make the installation process consistently easy between releases. You will need LumenVox licenses to correctly configure the LumenVox services. The examples shown here use both ASR and TTS services, and therefore require both license types, however if you require only ASR or TTS, the process should be essentially the same, but without the missing portion.
  • You will probably need to SIP phone to perform some test calls into the system. I used my old version of X-Lite, but you can use whichever SIP phone or or soft-phone you prefer.


Although this article and accompanying video series was created with the goal of using explicit versions during the walk-through, it is important to understand that this same approach would work for versions of LumenVox newer than the version 13.1 being described here.

The same is not necessarily true of Asterisk and UniMRCP, which are not written or maintained by LumenVox, so caution should be exercised when using different versions of those. Be sure to note that only certain combinations of Asterisk and UniMRCP are supported by LumenVox, as described in our Before you begin Asterisk Installation article.

The Big Picture

This is a basic diagram that describes the relationship between Asterisk, UniMRCP and LumenVox. As you can see, the UniMRCP module acts as an interface between Asterisk and the LumenVox Media Server. The LumenVox Media Server allows external software, such as UniMRCP to communicate using MRCP and request things like ASR and TTS resources when needed.

Asterisk does not know how to do this by itself, so UniMRCP is providing that functionality, allowing Asterisk programs to become speech enabled. This diagram shows all of these services running within the same server, however users can choose to install the LumenVox components on their own server if desired.

The preferred method of communicating with the Asterisk PBX is via the use of Asterisk modules provided by the UniMRCP Project. UniMRCP is an open-source implementation of the Media Resource Control Protocol (MRCP) which is fully supported by the LumenVox Media Server. UniMRCP makes available a number of Asterisk modules that allow Asterisk to act as an MRCP client to make automatic speech recognition (ASR) and text-to-speech (TTS) requests to the LumenVox Media Server.

Starting From a Fresh Instance

You don't need to start with a new instance of an operating system to install these things, but it is certainly cleaner to do so. In this example, we are using a brand new installation of CentOS6 64-bit. There is nothing special about that Operating System, and the installation process should be virtually identical for any other supported OS (such as CentOS/RHEL 5 and/or CentOS/RHEL7).

Setup the Environment

Before starting any installation, you should verify that your machine is currently up to date by running the yum update command. This should ensure you have the latest versions of things installed.

# yum update

Next, you should install packages that will be required by PJSIP, Asterisk and UniMRCP that we will be installing (again, several of these following steps are based on Justin Hester's blog at :

# yum install -y epel-release dmidecode gcc-c++ ncurses-devel libxml2-devel make wget openssl-devel newt-devel kernel-devel sqlite-devel libuuid-devel gtk2-devel jansson-devel binutils-devel libtool jansson-devel may run into some other packages that are required along the way, but these should cover most of the things you will need.

Installing PJSIP

Move to the /usr/src folder and get a copy of the PJSIP source that will be needed, then unpack it

# cd /usr/src
# wget 
# tar -jxvf pjproject-2.3.tar.bz2 

Move into the new folder that was created, containing the pjsip project source files and then run the configure script as shown below:

# cd pjproject-2.3
# ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr 

The next four commands will build, install and link the pjsip libraries:

# make dep
# make
# make install
# ldconfig

To verify that the PJSIP libraries have been dynamically linked, use the following command:

# ldconfig -p | grep pj

You should see a screen like this:

Installing Asterisk 13

To build and install Asterisk 13 from source, change directory  to /usr/src

# cd /usr/src

Use wget command to download the tarball from

# wget

Use tar to unpack the Asterisk source code into a new directory named asterisk-13.4.0 then cd into the directory

# tar -zxvf asterisk-13-current.tar.gz
# cd asterisk-13.4.0

The next set of commands will build and install Asterisk.

# ./configure --libdir=/usr/lib64

If you get any errors here, it probably means you have some tool or dependency missing that the configuration script needs. Typically the error will indicate what it needs, so you should install the missing components then re-run the ./configure script above, then once successful, continue on.

You should see a screen similar to this when it's successfully configured:

Run the make menuselect command to verify if the pjsip channel driver dependencies have been successfully installed

# make menuselect

Use the arrow keys to navigate to “Resource Modules” in the left column, about halfway down the list. Press the right arrow key and then scroll down until you see the list of modules beginning with “res_pjsip_”.

If these modules have [XXX] to the left of their name then the dependencies have not been met. You’ll need to go back to the /usr/src/pjproject directory, run the “make distclean” command and start over carefully looking for any error messages and proceed from there.

If you see [*] instead of [XXX] then the res_pjsip module’s dependencies have been met and you can proceed to the next steps. Your menuselect screen should look like this if everything was installed correctly:

After exiting the menuselect screen the next set of commands will build and install Asterisk along with a set of sample configuration files.

# make && make install
# make samples

Use the following command to setup Asterisk to start at boot time.

# make config

To immediately begin the Asterisk service without the need to reboot first.

# service asterisk start

To connect to asterisk

#asterisk -rvvvvvvvvvvvvvvv

You should see the something like this to confirm that asterisk is up and running:

Now, quit from the Asterisk console before moving on...

*CLI> quit

Simple dialplan to test Asterisk configuration

Edit the /etc/hosts file with the following settings and add a line for the IP address of your server and hostname (note that your IP address will almost certainly be different than the one shown here): EL6-VBOX

TIP: Make a backup copy of the /etc/asterisk/pjsip.conf file before we start any modifications, so that you will always have a copy of the original for later reference. Also, there is a sample copy of this file attached to the bottom of this article that you can download if you prefer.

# cp /etc/asterisk/pjsip.conf /etc/asterisk/pjsip.orig

Edit the /etc/asterisk/pjsip.conf file with the following settings:





TIP: Make a backup copy of the /etc/asterisk/extensions.conf file before we start any modifications, so that you will always have a copy of the original for later reference. Also, there is a sample copy of this file attached to the bottom of this article that you can download if you prefer.

# cp /etc/asterisk/extensions.conf /etc/asterisk/extensions.conf.orig

Edit the /etc/asterisk/extensions.conf file with the following settings (from our Testing the LumenVox Installation on Asterisk knowledge base article):


exten = 100,1,Goto(hello-world,s,1)
exten = 101,1,Goto(lumenvox-mrcp-tts-test,s,1)
exten = 102,1,Goto(lumenvox-mrcp-asr-test,s,1)
exten = 103,1,Goto(lumenvox-test,s,1)


exten = s,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()

exten = s,1,Answer
same = n,Wait(1)
same = n,MRCPSynth(Welcome to LumenVox)
same = n,Verbose(1, ${SYNTHSTATUS})

exten = s,1,Answer()
same = n,Wait(1)
same = n,MRCPRecog(builtin:grammar/boolean,&f=beep)
same = n,Verbose(1,Status is: ${RECOGSTATUS} and Result is: ${RECOG_RESULT})

exten = s,1,Answer
same = n,Wait(1)
same = n,SpeechCreate
same = n,SpeechLoadGrammar(yesno,builtin:grammar/boolean)
same = n,SpeechActivateGrammar(yesno)
same = n,SpeechBackground(beep)
same = n,Verbose(1,Result was ${SPEECH_TEXT(0)})
same = n,Verbose(1,Confidence was ${SPEECH_SCORE(0)}) 

Restart asterisk so that these changes take effect

CLI> core restart now

Configure SIP Phone and test the Hello World prompt playback

In this example, we're using an X-LITE SIP Phone, but you should use whichever SIP-phone / Soft-phone you are most comfortable with.

Be sure to set Domain to the IP address of the asterisk server, the UserID is 6001 and Password is password (both of which we configured in the PJSIP settings above)

Open an asterisk console window, with some verbosity

# asterisk -rvvvvvvvvvvvvvvv

Use the SIP phone to dial extension 100. You should hear a female voice saying "Hello World" and the asterisk console screen should look similar to this:

Installing UniMRCP dependencies 1.3.0

Navigate to the /usr/src directory

# cd /usr/src

Use wget command to download the tarball from

# wget –O unimrcp-deps-1.3.0.tar.gz

Use tar to unpack the unimrcp-deps  source code into a new directory name unimrcp-deps-1.3.0  then cd to the directory

# tar -zxvf unimrcp-deps-1.3.0.tar.gz
# cd unimrcp-deps-1.3.0

Run the script ./ and follow the instructions.

# ./

Make note of the installation directory and type y to continue installing the APR libraries and Sofia-sip.

You should confirm installation of the APR and Sofia-SIP libraries when prompted - these are both required by UniMRCP. This will complete the dependency installation requirements for UniMRCP, which can be installed next.

Installing UniMRCP

#cd /usr/src

# wget -O unimrcp-1.3.0.tar.gz
# tar -zxvf unimrcp-1.3.0.tar.gz
# cd unimrcp-1.3.0
# ./bootstrap

If you get any error messages about any missing components or tools here, you should install them, then rerun ./bootstrap before running the next configure command:

# ./configure --with-apr=/usr/local/apr --with-apr-util=/usr/local/apr

You should see a summary report similar to this:

****************************** REPORT ******************************

UniMRCP version............... : 1.3.0

APR version................... : 1.5.1
APR-util version.............. : 1.5.4
Sofia-SIP version............. : 1.12.11-225-g94cb919

Compiler...................... : gcc
Compiler flags................ : -g -O2 -pthread
Preprocessor definitions...... : -DLINUX -D_REENTRANT -D_GNU_SOURCE
Linker flags.................. :

UniMRCP client lib............ : yes
Sample UniMRCP client app..... : yes
Sample UMC C++ client app..... : yes
Misc ASR client lib and app... : yes

UniMRCP server lib............ : yes
UniMRCP server app............ : yes

Demo synthesizer plugin....... : yes
Demo recognizer plugin........ : yes
Demo verifier plugin.......... : yes
Recorder plugin............... : yes

Installation layout........... : classic
Installation directory........ : /usr/local/unimrcp


Assuming this worked correctly, you should then build the modules as shown here:

# make
# make install

Once this is completed, you can proceed to installing the asterisk-unimrcp modules...

Installing asterisk-unimrcp modules

Change into the /usr/src folder again and download/unpack the asterisk-unimrcp package we want to build:

#cd /usr/src
# wget -O asterisk-unimrcp-1.3.0.tar.gz
# tar -zxvf asterisk-unimrcp-1.3.0.tar.gz

Once unpacked, we can move into the asterisk-unimrcp folder and run the bootstrap and configure scripts as shown below

# cd asterisk-unimrcp-1.3.0
# ./bootstrap
# ./configure --prefix=/usr/lib64/asterisk/modules

If this is all successful, you should see the following report:

Which means you can continue to build the asterisk-unimrcp modules:

# make
# make install

Check to make sure that the following files exist:

# ls /etc/asterisk/mrcp.conf
# ls /etc/asterisk/res-speech-unimrcp.conf
# ls /usr/local/unimrcp/conf/client-profiles/lumenvox.xml
# ls /usr/local/unimrcp/conf/unimrcpclient.xml

All of these files should exist in the locations indicated. If you find one or more is missing, please recheck the recent instructions - in particular, verify that you ran "make install" correctly, which is what generates these sample configuration files.

Follow the instruction on our Installing and Configuring UniMRCP modules to copy and paste the contents of these files. You may want to make copies of the original files before making any changes.

Note:  Make sure the RTSP port of the MRCP server-port = 554 is correct in /etc/asterisk/mrcp.confAlso, there is a sample copy of this file attached to the bottom of this article that you can download if you prefer.

Start up asterisk if it's not already running with the -c option (console).

# asterisk -cvvvvvvvvvvvvvvvvvvvv

If asterisk is already running then use the -r option (remote) to connect to asterisk console

# asterisk -rvvvvvvvvvvvvvvvvvvvv

Check to make sure that the unimrcp modules are loaded
CLI> module show like  mrcp

You should see something like this:

If you do not see this, restart asterisk and try again:

CLI> core restart now
CLI> module show like mrcp

Installing LumenVox

Download and install LumenVox following the instructions in the Linux Installation article, which involves creating or updating the /etc/yum.repos.d/LumenVox.repo file according to your Linux Operating System. In the case of CentOS6 64-bit that we're using here, the file should look like this:

name=LumenVox Products x86_64

Edit the media_server.conf settings file and configure the sip_port = 0 setting to disable the LumenVox SIP interface altogether, since UniMRCP will only be communicating with LumenVox using MRCPv1 (RTSP).

We can run some basic sanity check to make sure LumenVox is configured and working correctly.

# lv_show-config -a 

Note that we expect the MRCPv2 (SIP) tests to fail here, because we disabled the sip_port in the LumenVox media_server.conf settings above. If you would like more information about the results of this test by reading our Running lv_show_config article.

Testing  the installation of LumenVox, Asterisk and UniMRCP

Use the same  softphone (X-LITE) that you configured earlier to make some test calls to verify that everything is installed and working.

Dial extension 100

This will test asterisk basic playback of the "Hello World" prompt. You should hear a female voice saying "Hello World". This verifies that Asterisk and PJSIP are installed and configured correctly (without using UniMRCP or LumenVox)

Dial extension 101

This will test MRCPSynth(Welcome  to LumenVox). You should hear the "Welcome to LumenVox" TTS voice audio. This confirms that Asterisk, UniMRCP (MRCPSynth) and LumenVox TTS services are all working and configured correctly.

Dial extension 102

This will test  MRCPRecog(builtin:grammar/boolean,&f=beep) - you should say either "yes" or "no" after you hear the beep. A successful result indicates that Asterisk, UniMRCP(MRCPRecog) and LumenVox ASR services are all working and configured correctly.

Dial extension 103

This will test  ASR with Generic Speech API. A successful result to this test indicates that Asterisk, UniMRCP (using the Generic API), LumenVox ASR services are all working and configured correctly.


If you reach this point, this means you have successfully installed and configured Asterisk 13, UniMRCP 1.3.0 and LumenVox 13.1

From here, you can continue on to build your own Asterisk applications. For more advanced examples and additional tips, please visit our LVDN site at, which has a number of useful articles and examples, such as the Asterisk Speech-Enabled Call Router in Dialplan example.

Sample Configuration Files

The file attachments below were used in the making of the videos - you may use these as a reference if you like, however be aware that any IP addresses in these settings may need to be updated to match your specific machine, as was mentioned above and in the videos.

extensions.conf 31.7 Kb Download File
lumenvox.xml 1.8 Kb Download File
mrcp.conf 1.4 Kb Download File
pjsip.conf 41.1 Kb Download File
res-speech-unimrcp.conf 883 b Download File
unimrcpclient.xml 4.1 Kb Download File