Configure Polycom IP Phone With Asterisk PBX

Introductory information on how to use Polycom® SoundPoint® IP phones and Polycom® SoundStation® IP conference phones with the Asterisk PBX.

This document assumes you have an installed and functioning Asterisk server.

In this document, the terms SoundPoint IP phone, SoundStation IP phone, and Asterisk/Asterisk server refer to the following:
SoundPoint IP phone—Polycom SoundPoint IP 301, 320, 330, 430, 450, 501, 550, 560, 601, 650, or 670 desktop phone SoundStation IP phone—Polycom SoundStation IP 4000, 6000, or 7000 conference phone. Asterisk or Asterisk server—a functioning Asterisk PBX installation.

Creating a Basic Configuration
The configuration file fragments in the following figure show the basic settings required to configure both Asterisk and the SoundPoint IP or SoundStation IP phones to successfully interoperate.
Note that these settings are minimal configurations and only show changes from default factory settings on the phone. If you are not sure of the current settings and their impact on the phone in your configuration, restore the phone to the factory default settings (see Removing Configuration Overrides).
To create a basic configuration:
1. For each phone to be connected, define the phone extension number in the Asterisk sip.conf file, as shown in the example below.

In the above example, the extension number 8055 is defined. This extension:

Is defined by the number inside square brackets [8055]

Has a SIP registration username of 8055 username

Has a SIP registration password of 8055 password

Uses the Asterisk mailbox number 8055 in the default context

Is an Asterisk type friend, meaning it is both a user and a peer

Is a dynamic host according to Asterisk

Sends DTMF tones as special RTP packets according to RFC2833

Exists in the Asterisk context Polycom

Allows RTP traffic to bypass the Asterisk server

Sends a caller ID with text as Polycom Demo and a number 8055


2. Define the dial plan in the Asterisk extensions.conf file for the context declared above, as shown in the example below.

[polycom]
exten =>8055,1,Dial (SIP/8055,25)
exten =>8055,2,Voicemail,u8055
exten =>8055,102,Voicemail,b8055

In the above example, the extension number 8055 was added to the dial plan.

When someone dials this extension from within the context :

Asterisk will first try to ring the extension 8055 using the SIP channel, waiting up to 25 seconds for the extension to answer.

If the extension is unanswered, Asterisk will direct it to mailbox 8055.

If the extension is busy, Asterisk will direct it to mailbox 8055.


3. Once the Asterisk configuration is complete, configure the SoundPoint IP or SoundStation IP phone. You can do this in one of three ways:


1. Via a centralized provisioning (or boot) server using configuration files. This method offers the most flexibility. This is the recommended method of configuring Polycom phones and the method that is used throughout this document.
2. Via the phone’s built-in web server using a web browser. Not all options are available using this method, but you can use this method to configure the phone to register with Asterisk.

3. Via the phone’s menu system (using the phone’s buttons). Not all options are available using this method, but you can use this method to configure the phone to register with Asterisk.