5513806d0954cd857b80d2563744d91445f9582f
howto/Telephony-Asterisk.md
| ... | ... | @@ -45,7 +45,7 @@ services: |
| 45 | 45 | restart: unless-stopped |
| 46 | 46 | ``` |
| 47 | 47 | |
| 48 | -# Example configuration |
|
| 48 | +## Example configuration |
|
| 49 | 49 | |
| 50 | 50 | Asterisk relies on many configuration files. This guide will focus only on two main config files: |
| 51 | 51 | - `pjsip.conf`: Network, Endpoints, and Authentication |
| ... | ... | @@ -59,7 +59,7 @@ When copying the configurations below, replace the following variables (remove t |
| 59 | 59 | * `<FULL_NUMBER>`: The full `+042` number for your extension (e.g., `+042400000001`) |
| 60 | 60 | * `<SECRET_PASS>`: A strong password |
| 61 | 61 | |
| 62 | -## PJSIP (`pjsip.conf`) |
|
| 62 | +### PJSIP (`pjsip.conf`) |
|
| 63 | 63 | |
| 64 | 64 | This file defines how Asterisk listens to the network, sets up templates for devices, creates your local extension, and prepares an anonymous endpoint to catch incoming ENUM calls. |
| 65 | 65 | |
| ... | ... | @@ -155,7 +155,7 @@ allow=!all,ulaw,alaw ; Use standard audio formats |
| 155 | 155 | send_pai=yes |
| 156 | 156 | ``` |
| 157 | 157 | |
| 158 | -## Dialplan (`extensions.conf`) |
|
| 158 | +### Dialplan (`extensions.conf`) |
|
| 159 | 159 | |
| 160 | 160 | The file tells Asterisk how to route numbers. It is divided into serveral contexts. |
| 161 | 161 | |
| ... | ... | @@ -226,7 +226,7 @@ Apply your configuration by entering the Asterisk CLI: |
| 226 | 226 | asterisk -rx 'core reload' |
| 227 | 227 | ``` |
| 228 | 228 | |
| 229 | -# Setting up your extension |
|
| 229 | +## Setting up your extension |
|
| 230 | 230 | |
| 231 | 231 | You can now connect a softphone (like MicroSIP, Zoiper, or Linphone) or a hardware IP Phone/ATA using the following details: |
| 232 | 232 | |
| ... | ... | @@ -238,3 +238,4 @@ You can now connect a softphone (like MicroSIP, Zoiper, or Linphone) or a hardwa |
| 238 | 238 | Once connected, ask a friend to call your `<FULL_NUMBER>`. |
| 239 | 239 | |
| 240 | 240 | *Tip: Install and use `sngrep -c` in your PBX's terminal to visually monitor SIP traffic if calls fail.* |
| 241 | + |