80 lines
3.8 KiB
Markdown
80 lines
3.8 KiB
Markdown
# Analog phones (ATAs)
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An analog telephone adapter (ATA) connects ordinary phones, with a phone cord, to SelfPhoneNG.
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Each port of the ATA is one user's phone. Calls, voicemail (with the stutter dial tone when a
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message waits), `*` codes, transfer with the hook flash, and 911 all work.
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For now you set an ATA up by hand, in its own web page. Automatic setup, like the desk phones
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get, is planned for the Grandstream HT802 V2 first.
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Field names below are from Grandstream's HT802 V2 guide; other HT80x/HT81x models and firmware
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versions may name them a little differently.
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## 1. In SelfPhoneNG
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1. **Users**: one user for each port you'll use (port 2 can stay empty).
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2. Open the user, **Credentials** tab, **Reveal password**. Keep the server, port, username,
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password and **dial plan** at hand. Viewing it is recorded in the audit log.
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You don't need to add the ATA on the Phones page: "Other (set up by hand)" there shows the same
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settings, and is only for one line per device.
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## 2. On the ATA (Grandstream HT802 V2)
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Plug the ATA into the office network and find its address (in the router's list of devices, or
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dial `***` then `02` on the analog phone to hear it). Open `http://<its address>`. The admin
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password is on the label underneath; change it after you sign in.
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On **FXS PORT 1** (and **FXS PORT 2** with the second user):
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| Field | Value |
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|---|---|
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| Profile Active | Yes |
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| Primary SIP Server | the server and port from the Credentials tab, for example `pbx.example.com:5060` |
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| Outbound Proxy | empty |
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| SIP Transport | UDP |
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| SIP User ID | the username (the extension number) |
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| Authenticate ID | the username again |
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| Authenticate Password | the password |
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| Name | the person's name (shown to the people they call) |
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| SIP Registration | Yes |
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| Allow Incoming SIP Messages from SIP Proxy Only | Yes: nobody but SelfPhoneNG can ring the phone |
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| DTMF | RFC2833 first |
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| Preferred Vocoder | PCMU first |
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| Dial Plan | see below |
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Click **Apply**. On the **Status** page the port should say **Registered** within a minute, and on
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the server `pjsip show contacts` ([Troubleshooting](troubleshooting.md)) lists the extension.
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### Dial plan
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An analog phone has no send button: the ATA decides when you've finished dialing. Its default
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dial plan, `{ x+ | … }`, always waits about 4 seconds after the last digit, even for 911. Paste
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the **Dial plan** from the Credentials tab into **Dial Plan** instead. For the usual setup it
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looks like:
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```
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{ 911 | 9911 | [2-9]xx[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | xxx | *x+ | **x+ }
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```
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- 911 and 9911 go out at once (US law requires 911 to work without a prefix).
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- 10- and 11-digit numbers go out at once. International numbers (`011…`, when that route is on)
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go after a pause.
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- Internal numbers and `*` codes go after a pause (an internal number could be the start of an
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outside one), or press **#** to send now.
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- Numbers that match no rule can't be dialed from the ATA. It's made from your outbound routes
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and the lengths of your internal numbers: after changing either (a new route, or Settings →
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Numbering), copy it again.
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## 3. Things to know
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- **911:** each port's user has its own emergency caller ID, like any phone (Users →
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Emergency). An ATA that moves to another address needs the E911 address updated at voip.ms.
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- **Fax:** not supported. Many fax machines still work through an ATA with Fax Mode set to
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**Pass-Through** and only PCMU allowed, but expect failures on long faxes. T.38 isn't set up on
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the SIP trunk.
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- **Not on analog phones:** busy lights, the company phonebook, the browser phone features.
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- **Caller ID** shows on analog phones that support it (US: Bellcore, the ATA's default).
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- **Away from the office:** an ATA works anywhere a desk phone works. The phone network limit
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(Settings → Security) applies to it the same way.
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