We have made it super simple to connect dSIPRouter to Twilio Elastic SIP Trunking (ESIPT). The main component that makes this happen is the new CarrierGroup API. This API allows you to programmatically add carriers to dSIPRouter. This API follows the same format as our other dSIPRouter API. In this article, I will explain how to connect to Twilio ESIPT from the dSIPRouter UI and explain how to use the underlying API.

Installing dSIPRouter

The best way to install dSIPRouter is to use the one line installation commands which can be found here or use Terraform to deploy.

Get Twilio Credentials

You can obtain Twilio credentials from the Twilio Console

1. Create a Twilio account or login to the Twilio Console if you have an existing account.
2. Save the Account SID and Auth Token from the Twilio Console somewhere safe. You will need them to later

Create Carrier Group for Twilio

Now that dSIPRouter is installed let’s login and create a Carrier Group. This will allow you to setup a SIP connection between dSIPRouter and Twilio.

1. Click Carrier Groups
2. Click Add
3. You will need to enter a name for the Carrier Group, select Twilio from the Plugin, enter in the Twilio Account SID and Account Token that we saved in the last section.
4. Click Add

This will trigger an API to Twilio Elastic SIP Trunking that will create an Elastic SIP Trunk which can be seen in the Twilio Console.

Setup Default Outbound Route

Now make your new Twilio Carrier Group the default outbound route

1. Click “Outbound Routes”
2. Select the first rule and change the route to the MackTwilioCarrierGroup
3. Click Update
4. Click “Reload Kamailio” in the upper right of the UI

Testing: Setup an Endpoint Group

We can now test a call by creating an End Point Group using IP Auth or Username/Password Auth. I will use IP Auth

1. Click Endpoint Groups
2. Click Add
3. Click Endpoints Tab
4. Add your IP address and click Save
5. Click Add
6. Click “Reload Kamailio” in the upper right of the UI

Testing: Place a Call

We are going to use Zoiper to place a test call

1. Start Zoiper
2. Add Account
3. Just add a Domain and Username. Note, the username is the Caller ID that will be sent when you place a test call. Note, Twilio will only send calls to validated numbers on the account when your account is in Trial mode.
4. Place a call. Make sure the number is e.164 format, which looks like this +19544242424

We are introducing a new API that will enable the provisioning of extensions within a backend media server. The API will provide an abstraction layer for provisioning different backend media servers. The first release will focus on FusionPBX.

dSIPRouter has a concept of an Endpoint Group. An endpoint group can be mapped to a FusionPBX Cluster. When this happens the domains from FusionPBX is automatically sync’d with dSIPRouter. We will leverage the existing connection to enable the ability to provision and update Domains and Extensions within FusionPBX. In the future, we will support FreePBX and other media servers as well. Hence, giving users one API for provisioning users and dSIPRouter handles the complexity of translating the request into the backend media server.

The feature will be available in 0.642. You can provide comments and feedback on the API by submitting comments in the GitHub issue which can be found here

The API endpoints for this release are:

/mediaserver/domain/
/mediaserver/extension/

The Payload for this release:

Payload for Domain

{
   domain_id:
   name: string,
   enabled: boolean,
   description: string,
   config_id: endpointgroup_id | conf_id
}

Payload for Extension

{
    type: single| multiple,
    num_of_extensions: integer,
    attributes: {

      domain_id: string | null,
      account_code: string,
      extension: string,
      password: string|null ,
      outbound_caller_number: string|null,
      outbound_caller_name: string:null,
      vm_enabled: boolean,
      vm_password: string,
      vm_notify_email: string,
      enabled: boolean,
      config_id: endpointgroup_id | conf_id
    }
}

Example Usage

The following example will provision a new Endpoint Group in dSIPRouter and a Domain and Extension in FusionPBX. Note, that the API will also support updating and deleting as well.

1) The user will create an endpoint group to represent a FusionPBX standalone instance or a FusionPBX cluster

https://{{DSIP_ADDR}}:5000/api/v1/endpointgroups

They will receive an endpoint group id, which will be used for provisioning the backend media server

2) The user can then create a new domain

https://{{DSIP_ADDR}}:5000/api/mediaserver/domain/

{
   domain_id: null
   name: AprilandMackCo,
   enabled: true,
   description: "April and Mack Co,
   config_id: 64
}

The user will receive the domain_id, let’s assume its 98

3) The user can create one user or a set of users. We are going to create 10 users

https://{{DSIP_ADDR}}:5000/api/mediaserver/extension/

{
    type: "multiple",
    num_of_extensions: 10,
    attributes: {

      domain_id: 98,
      account_code: "124-24245",
      extension: "1000",
      password: "starterpassword" ,
      outbound_caller_number: "8889072085",
      outbound_caller_name: "dOpenSource",
      vm_enabled: true,
      vm_password: "94145",
      vm_notify_email: "",
      enabled: true,
      config_id: 64
    }
}

This API will create 10 extensions starting from 1000 with the same attributes. You can also create a single extension.

dSIPRouter 0.63 was recently released with out of the box support for Twilio Elastic SIP Trunking. The design goal for the first release of this feature was to make it simple for a user to download dSIPRouter and setup Twilio as the carrier.

Assumptions:

• dSIPRouter is installed. If not, you can install it within 12 minutes. There is a one-line command that will kick-off the installer and it will take about 12 minutes to install. The details can be found here{:target=”_blank”}

• You have a Twilio account. It only takes a few minutes to create an account. You can create an account by going here

Setting up Twilio Elastic SIP Trunking

1. Login to Twilio

2. Enter Elastic SIP Trunks in the top right search bar and click enter.  Notice that Twilio gives you a double digit Trial Balance.  This is better then most SIP providers that I worked with in the past.

   

3. Click the + sign to create a new SIP Trunk.  Enter a name for your SIP Trunk and click “Create”

   

4. You will see a screen that looks like this

   

5. Since we are focused on sending calls outbound we are going to click on the “Termination” menu option.  We are going to use IP Authentication.  So, we just need to define a unique SIP URI for signaling to Twilio. Twilio doesn’t accept SIP requests that come from an IP address.  Also, we need to define an Access Control List that contains the IP address of your dSIPRouter instance.  If you have multiple instances of dSIPRouter then define all of the IP addresses.

   

This is all we need to do on the Twilio side.  In the next section we will focus on configuring dSIPRouter

Setting up dSIPRouter to work with Twilio Elastic SIP Trunking

Setting up the Carrier Group

1. Login to dSIPRouter
2. Click Carrier Groups
3. Click Add
4. Provide a name for the Carrier Group and click Add

   

5. Click on the Carrier Group that you just added

6. Click on Endpoint and click Add.

7. Enter the Twilio SIP URI that you defined in the Twilio portal

   

8. Click “Reload Kamailio”

Make the Twilio Carrier Group the Default Outbound Carrier

1. Click on Global Outbound Routes

2. Click on the Default Outbound Route and select Twilio Outbound

   

Making an Outbound Call

In this section we are going to use IP authentication to send a call using a softphone.

1. Click Endpoint Groups
2. Click Add enter “Test Endpoint” as the Friendly Name
3. Click the Endpoint Tab and enter the IP address you will send calls from
4. Click Add

   

5. Click “Reload Kamailio”

6. Start your softphone and configure it to send a call to dSIPRouter

The purpose of this tutorial is to explain how to configure the Yealink W60P Cordless DECT IP Phone with FreePBX, Asterisk of FusionPBX.  We will assume that you have the extension already configured in one of the aforementioned systems.  Therefore, we will focus on the steps needed to configure the phone.

I must note that the experience with configuring the Yealink W60P was much better then configuring the Panasonic KX-TGP600

The high-level steps needed to complete this are listed below.  We will go into detail for each section.

• Locate Web Management UI IP Address
• Log into theWeb Management UI
• Configure the phone settings via the Web Management UI

Locate Web Management UI IP Address

1. Pickup one of the Handsets that is registered to the base station
2. Click OK
3. Click OK on the (i) Icon
4. Click OK on the BASE option
5. The IP address of the Web Management UI (aka Base station) will be shown

Log into theWeb Management GUI

1. Open a web browser
2. Enter http://<IP address of the Web Management UI>
3. Enter admin/admin as the username and password

Configure the phone settings via the Web Management GUI

1. Click Account
2. Select the Account you want to configure.  We will use Account1
3. Enter in the following required and optional fields:

Required Fields

Line Action

Label

Display Name

Register Name

Username Name

Password

Optional Fields
These fields are only needed if you have a proxy server in front of your media server such as Kamailio or OpenSIP’s

Enable Outbound Proxy Server

Outbound Proxy Server 1

The purpose of this article is to explain how to track down what happened to a call in Asterisk.

Suppose you want a call trace from a specific call that has already happened, so it’s too late to see it in the console live. First locate the call in the CDR, and get the uniquieid from the system column for the call in question:

Login to the Asterisk/FreePBX Server and grep the Asterisk full logs for that value:

[root@34693894 ~]# grep 1518526777.67 /var/log/asterisk/full*
/var/log/asterisk/full-20180214:[2018-02-13 08:59:37] VERBOSE[24184][C-00000001] pbx.c: Executing [s@macro-user-callerid:1] Set("SIP/5002-00000001", "TOUCH_MONITOR=1518526777.67") in new stack
/var/log/asterisk/full-20180214:[2018-02-13 08:59:37] VERBOSE[24184][C-00000001] pbx.c: Executing [700@from-internal:37] QueueLog("SIP/5002-00000001", "700,1518526777.67,NONE,DID,") in new stack

This will return a few lines, which will include the Asterisk CALL-ID (not to be confused with CallerID), the second number in the square brackets. It will also return a file name, as full logs are rotated daily and purged weekly. Perform a second grep on the CALL-ID and filename like:

[root@34693894 ~]# grep C-00000001 /var/log/asterisk/full-20180214
[2018-02-13 08:59:37] VERBOSE[29432][C-00000001] netsock2.c: Using SIP RTP TOS bits 184
[2018-02-13 08:59:37] VERBOSE[29432][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[2018-02-13 08:59:37] VERBOSE[24184][C-00000001] pbx.c: Executing [700@from-internal:1] Macro("SIP/5002-00000001", "user-callerid,") in new stack
[2018-02-13 08:59:37] VERBOSE[24184][C-00000001] pbx.c: Executing [s@macro-user-callerid:1] Set("SIP/5002-00000001", "TOUCH_MONITOR=1518526777.67") in new stack
[2018-02-13 08:59:37] VERBOSE[24184][C-00000001] pbx.c: Executing [s@macro-user-callerid:2] Set("SIP/5002-00000001", "AMPUSER=5002") in new stack
[2018-02-13 08:59:37] VERBOSE[24184][C-00000001] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("SIP/5002-00000001", "0?report") in new stack

References:

https://wiki.freepbx.org/display/SUP/Providing+Great+Debug