Configuring the Asterisk Zap channel

The Zap module is what Asterisk uses to interface with the Zaptel device driver. In this file, you'll need to add your definitions for the context of the PRI connection, signaling, and available channels. There are many configuration parameters available for configuring the Zap device, including automatic fax detection, caller ID details, echo cancellation, and default ring cadences. These details, however, are going to be either outside the scope of this article or may be covered in later articles in this series.

Signalling

The important parameters for the Zap device are going to be under the [channels] context. Using a text editor, modify /etc/asterisk/zapata.conf, find the option signalling, and make it signalling=fxo_ls. For the options that will work with your specific configuration, you need to speak with your provider; but here's the list of available options, as provided by http://www.voip-info.org/:

• em: E & M Immediate Start
• em_w: E & M Wink Start
• em_e1: E & M CAS signalling for E1 lines
• featd: Feature Group D (The fake, Adtran style, DTMF)
• featdmf_ta: Feature Group D (The real thing, MF [domestic, US]) through a Tandem Access point
• fgccama: Feature Group C-CAMA (DP DNIS, MF ANI)
• fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI)
• featdmf: Feature Group D (The real thing, MF [domestic, US])
• featb: Feature Group B (MF [domestic, US])
• fxs_ls: FXS (Loop Start)
• fxs_gs: FXS (Ground Start)
• fxs_ks: FXS (Kewl Start)
• fxo_ls: FXO (Loop Start)
• fxo_gs: FXO (Ground Start)
• fxo_ks: FXO (Kewl Start)
• pri_cpe: PRI signalling, CPE side
• pri_net: PRI signalling, Network side (for instance, side that provides the dialtone)
• sf: SF (Inband Tone) Signalling
• sf_w: SF Wink
• sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
• sf_featdmf: SF Feature Group D (The real thing, MF [domestic, US])
• sf_featb: SF Feature Group B (MF [domestic, US])
• e911: E911 (MF) style signalling. Originating switch goes off-hook, far-end winks, originating sends KP-911-ST, far-end gives answer supervision, Originating-end sends KP-0-ANI-ST

The following are used for Radio interfaces:

• fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank)
• fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank)
• fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank)
• fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank)
• em_rx: Receive audio/COR on an E&M interface (1-way)
• em_tx: Transmit audio/PTT on an E&M interface (1-way)
• em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface (2-way)
• em_rxtx: Same as em_txrx (for our dyslexic friends)
• sf_rx: Receive audio/COR on an SF interface (1-way)
• sf_tx: Transmit audio/PTT on an SF interface (1-way)
• sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface (2-way)
• sf_rxtx: Same as sf_txrx (for our dyslexic friends)

Switchtype

The next option you want to look at is your PRI switch type. This is controlled through the switchtype option. The default is national; however, here is a list of available options:

• national: National ISDN type2 (common in the US)
• ni1: National ISDN type 1
• dms100: Nortel DMS100
• 4ess: AT&T 4ESS
• 5ess: Lucent 5ESS
• euroisdn: EuroISDN
• qsig: Minimalistic protocol to build a "network" with two or more PBX of different vendors

Channels and context

After the signalling and switchtype are defined, the next step is to define the channels that the Zap channel works on. These translate to the D channels of the PRI device, which is defined through the option channel => 1-23.

You'll need to define the context that these channels with operate in context = remote. Optionally, if your configuration will be splitting your T1 channels for dedicated DID's to groups/spans of channels, it's important to note that your context is defined after the channel group. So if you need channels 1-12 to be in context A and channels 13-23 to be in context B, you can do it.

The dial plan

Now that your Zap channel in Asterisk is configured to work with the Zaptel device, it's time to modify the dial plan to allow for incoming and outgoing calls on your PRI trunk. You'll want to have the outside world call you on your DID numbers that will connect to your PRI card, then to Asterisk, then on to your SIP client phone inside your office.

You should remember from the previous section that we set Zap channels 1 through 23 in the context remote. This is to signify that all communications on these channels connect directly to the PSTN (Public Switched Telephone Network), and thus are remote connections.

The first thing to do is to open /etc/asterisk/extensions.conf with your favorite text editor. At the bottom of the file, add the extensions that will describe the remote context and our PRI connections. Figure C shows our remote context in action.

Figure C

Add the extensions you need to the /etc/asterisk/extensions.conf file.

In this example, you can see that the extension is the DID (Direct Inward Dialing) number (your telephone number). It's picked up on the first step using the Goto function. In Asterisk, it forwards the call to the office context at extension 212 priority 1. This translates to 555-1234 being the direct number to extension 212 in the office. Alternatively you can treat the call like any other extension, including being able to run an AGI script, or forward the call to a call queue which will be covered in depth in later articles.