diff --git a/docs/guide-routr-as-asterisk-frontend.md b/docs/guide-routr-as-asterisk-frontend.md
index c2f484b26..c0c8ccdc7 100644
--- a/docs/guide-routr-as-asterisk-frontend.md
+++ b/docs/guide-routr-as-asterisk-frontend.md
@@ -5,7 +5,7 @@ title: Routr as Asterisk Frontend
This guide explores the use case of using Asterisk merely as a Media Server and a more specialized software, like **Routr**, to take care of the signaling and resource management. In other words, Asterisk will be in charge of the ivrs, voice mail, call recording, while **Routr** deals with connecting Agents, Peers, and Gateways. The following illustration depicts our scenario:
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**Content**
@@ -47,22 +47,22 @@ The first file we will examine and change is `config/peers.yml`. Make note of th
Head to the console and run the command `rctl -- get peers` to confirm that the Peer exist. The result should be as follows:
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Next, we focus our attention to `domains.yml` and `agents.yml`. With a fresh installation, we don't need to make any changes to this files. However, you could run the commands `get domains` and `get agents` to ensure that both, the Agent and the Domain, exist on the server. Your output should look similar to:
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Use the information in `agents.yml` to configure your SIP phone. The relevant information is found in `spec.credentials`. Mine looks like this:
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> Make the adjustments based on your prefer SIP phone.
You can verify that your device registered correctly with **Routr** by running the `locate` command:
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## Configuring Asterisk
@@ -120,7 +120,7 @@ exten => 1001,n,Hangup
Restart your Asterisk and check the location service. A new device will appear.
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## Calling Asterisk from John's device
diff --git a/docs/guide-securing-the-signal.md b/docs/guide-securing-the-signal.md
index 19a98f4ce..d7cb57864 100644
--- a/docs/guide-securing-the-signal.md
+++ b/docs/guide-securing-the-signal.md
@@ -7,7 +7,7 @@ Follow this guide to secure the signaling between your endpoints and **Routr**.
> For this guide we will use a fictitious domain name to demonstrate the process of securing the signaling path
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## Creating a Java Keystore(.JKS) Certificate
@@ -94,8 +94,8 @@ openssl s_client -host 192.168.1.2 -port 5061 # Remember to use Routr's IP
Go to the account that you want to secure, select `Advanced -> Sip Signaling` and change the parameter `Primary Proxy` to `${proxyHost}:${proxyPort};transport=tls`. See the example in the following image:
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If everything went well you should see a green padlock like the one in the image bellow:
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diff --git a/docs/guide-voip-network-setup.md b/docs/guide-voip-network-setup.md
index 4ab716b64..5c4b96f95 100755
--- a/docs/guide-voip-network-setup.md
+++ b/docs/guide-voip-network-setup.md
@@ -121,11 +121,11 @@ Your output should be as follows:
**Starting the Server**
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**Verifying the Configuration**
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## Configuring the Sip devices
@@ -133,14 +133,14 @@ Your output should be as follows:
Configure your softphone using the information you gather in the last step. Start by completing only the required information: _username_, _domain_, _password_. Also, In the advanced section use the server's **IP** as your _Registry Server_ and _Proxy_. Here is how mine looks like:
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> Make sure to check the box "Use this account" to register your device
If everything went well we just need to confirm that both softphones have registered correctly. Conveniently you can use the `.rctl locate` to obtain a list of "online" devices. This may seem like a lot of information. But what's relevant here is that both `1001` and `1002` are present in the location service and therefore can reach each other.
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## Making calls
diff --git a/docs/theme/partials/header.html b/docs/theme/partials/header.html
index 93a6ebeb2..c57307b4f 100644
--- a/docs/theme/partials/header.html
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@@ -51,7 +51,7 @@
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