This will install the Digital Ocean provider since our terraform script is configured to work with Digital Ocean. Edit the terraform.tfvars file, which sets the values of variables needed by the terraform script.
export DIGITALOCEAN_TOKEN=<Digital Ocean token key> git clone https://github.com/dOpensource/dsiprouter cd dsiprouter/resources/terraform/do copy terraform.tfvars.sample terraform.tfvars terraform init
Once modified, you should have all you need to deploy an instance via Terraform.
pvt_key_path="C:/Users/alems/Working/andrew" (adjust using path format, andrew is actually the pvt key)
If you want to build from another branch you would pass in the branch variable with the value of the branch like this
This is what it will look like as it starts to build: This is what it will look like after it’s completed: You will now be able to login to dSIPRouter using the credentials generated.
terraform apply -var branch=v0.643
4. Save and run the following commands from the FreeSWITCH cli
<list name="my_list_001" total-weight="2"> <node name="978fefdd-45cd-4457-8ecd-5d82eaea3b12" weight="1"/> <!-- my_gateway_001 - its just comment with actual GW name --> <node name="1a8d448d-e06e-2dab-b6f9-4421a98e4d8e" weight="1"/> <!-- my_gateway_002 --> </list>
5. Updated your Outbound Routing setting. You have something like this now: But, you want to change the last line to something like this: The Distributor command above takes a string value that contains a list that was defined in distributor.conf.xml and returns a gateway based on the weight defined in the configuration. In this case, “my_list_001” is the list used. This module can be used for other use cases where you need to generate random/weighted values.
fs_cli > reloadxml fs_cli > distributor_ctl reload
In another example, we will run SIPp at 7 calls every 2 seconds using a SIP Proxy such as Kamailio and OpenSIPS friendly scenario file. If you need help building a scenario file please contact us and we can estimate the about of hours needs to build a scenario file to meet your requirements :
sipp -sn uac -r 7 -rp 2000 <ip of server>
You can pause the traffic by pressing the ‘p’ key. SIPp will stop placing new calls and wait until all current calls end. You can resume the traffic by pressing ‘p’. To quit SIPp, press the ‘q’ key. SIPp will stop placing new calls and wait until all current calls end. SIPp will then exit. You can also force SIPp to quit immediately by pressing the ‘Q’ key. Current calls will be terminated by sending a BYE or CANCEL message (depending if the calls have been established or not). The same behaviour is obtained by pressing ‘q’ twice.
sipp -sf sipproxyfriendly.xml -r 7 -rp 2000 <ip of server>
sipML is the WebRTC Client that we are going to use. We need to download the repository
yum install git cd /var/www/html/ git clone https://github.com/DoubangoTelecom/sipml5.git chown -R asterisk:asterisk sipml5/
vim /etc/asterisk/http_custom.conf tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/pki/tls/certs/localhost.crt tlsprivatekey=/etc/pki/tls/private/localhost.key
We also need to give asterisk permissions to read the tls certs
chown asterisk:asterisk /etc/pki/tls/certs/localhost.crt chown asterisk:asterisk /etc/pki/tls/private/localhost.key
In older version of freepbx, they do not support wss transports, so this will need to be manually configured in /etc/asterisk/sip_custom.conf replacing SOME_EXTENSION and SOMESECRET. The important line is the
transport=wss,udp,tcp,tls, which will have wss as the first entry. dtlscertfile and dtlsprivatekey will need to be pointed at the same key cert and key setup in http_custom.conf
[SOME_EXTENSION] deny=0.0.0.0/0.0.0.0 secret=1234pccw dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes mediaencryption=yes sendrpid=pai type=friend nat=force_rport,comedia port=5060 qualify=yes qualifyfreq=60 transport=wss,udp,tcp,tls avpf=yes force_avp=no icesupport=yes encryption=yes callgroup= pickupgroup= dial=SIP/1103 permit=0.0.0.0/0.0.0.0 callerid=SOME_EXTENSION callcounter=yes faxdetect=no cc_monitor_policy=generic dtlsenable=yes dltsverify=no dtlscertfile=/etc/pki/tls/certs/localhost.crt dtlsprivatekey=/etc/pki/tls/private/localhost.key
https://<server-name>/sipml5. Make sure you include the https and click on the demo button. You should now be at a registration screen. Enter in the extension you would like to register as in the display name and private identity. The public identity will follow the following format:
The password will be the secret set for your extensions and the realm will be the ip address or domain name of your server.
We also need to configure expert mode to set the wss address and stun settings.
Under expert mode, the WebSocket Server URL follows the following syntax:
Then set the ice server to the following google address:
Enable RTCWeb Breaker and hit save.
You should now be able to register to your extension. To troubleshoot, you can bring up the console in chrome by right clicking and selecting inspect. Additionally, make sure you have opened the necessary wss and rtp ports in your firewall (8089/tcp, 10000-20000/udp)
To make a call between two webrtc phones, you will need to install chromium, an open source version of the Google Chrome browser. You can alternatively use two computers with chrome installed. You can add a second extension in
/etc/asterisk/sip_custom.conf following the same syntax as the previous extension. After an asterisk restart, you should be able to register to the new extension using the same methods and place a call between the two browsers.
We recently worked with a customer that had a requirement that their application needed to connect via Site-to-Site VPN to there clients application. They had a few choices, but they decided to move there application to Amazon Web Services (AWS) and connect to there clients datacenter from there. Therefore we setup a Virtual Private Cloud (VPC) within Amazon and started down the path of setting up a Site-to-Site Virtual Private Network (VPN) connection.
There are multiple ways of implementing a VPN within Amazon as discussed here. In most cases, it’s going to come down to using a AWS Hardware VPN or a Software VPN. The AWS Hardware VPN can be configured within a couple clicks and it gives you the option to generate the configuration for multiple well known firewalls, which you can use to configure your firewall or you can provide to your firewall administrator. The Software VPN consists of running an EC2 instance that has software that implements VPN functionality.
The main factor in deciding AWS Hardware VPN versus Software VPN should be based on who’s initiating the traffic. In our case, the customers application needed to initiate the request. This means that we had to leverage the Software VPN approach because the AWS Hardware VPN can not initiate traffic. It can only accept request. So, it’s great for a company that wants to migrate systems from there datacenter to Amazon and then have there user access the systems. Hence, their users are the initiators of the traffic.
The installation and setup of a Software VPN isn’t really that difficult, but you have to have some basic understanding of how AWS networking works. There are a few Software VPN implementations, but we selected OpenSWAN. There’s a few good articles that we used.
One of the main gotcha’s in setting up OpenSwan are to ensure that the Access Control List (ACL) defined by the far end (the Router that you are establishing the VPN with) matches the Right side configuration parameter within the setup. Once you read thru the above articles you will know what I mean.
The average time to setup an AWS Hardware VPN is 5 hours. This includes configuration, testing and turn-up with the far end.
The average time to setup a Software VPN is 10-20 hours. It really depends on the complexity of the Amazon VPC and how you need traffic to be routed and represented.
We provide Amazon Web Services (AWS) consulting with a focus on DevOPs and Infrastructure Migration.
We are proud to be based in Detroit, MI with coverage from 9am-8pm ET. We have staff on the East and
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This quick article explains how to configure Aastra phones to register with FreePBX 13.x as an extension.
In order to manually provision phones you need the following basic information:
In our example, lets assume the following internal network info for the FreePBX server:
You now need the Extension and Secret. This can be found by logging into the FreePBX GUI and going to extensions section of the interface. You will see a list of extensions. Below we only show one extension, which is 10000
You can click on that extension and locate the Secret, which is outlined in red.
Now that you have this information you can login to the Aastra phone and click on “Global SIP”. You will enter the Extension and Secret into the fields that are outlined at the top and the Server IP and Server Port will replace the data outlined at the bottom.
Note – you will need to reboot your phone after configuring the phone.
Issue: Needed to get the IP address of a VM running on my XenServer.
Resolution: Execute this command
xe vm-list params=name-label,networks | grep -v "^$"
The command yielded me a list of the VM’s with the ip address.
name-label ( RW) : DevStackOSDomU networks (MRO): 2/ip: 172.24.4.10; 1/ip: 10.255.255.255; 0/ip: 10.10.10.187 name-label ( RW) : Control domain on host: warrier.dopensource.com networks (MRO): <not in database>
Note, I only had 1 VM on this Xenserver, but I needed the IP address:
There’s a lot of FreeSWITCH material out on the web. Our goal is to provide you a concise document that provides the necessary steps or scripts to install the latest stable release of FreeSWITCH.
FreeSWITCH is an OpenSource VoIP platform that enables you to build communication platforms that span from a simple PBX to a sophisticated backend communication framework that allows thousands of calls to interconnect per second.
wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -
echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list
apt-get update && apt-get install -y freeswitch-meta-all
sed -i "s/1234/10082013/" /etc/freeswitch/vars.xml fs_cli -x reloadxml
Where the extension passwords will now be 10082013. You can change the default password by just replacing 10082013 with your password.
Now we can validate that the install was successful. We can do this by registering 1 or more SIP phones. The default install of FreeSWITCH comes with a set of users. Let’s register extension 1000 and 1001 with two separate VoIP phones and make calls between the sip phones. Remember that your password for each extension is 10082013 or whatever you manually changed it too.