VoIP Phones - Grandstream HT8xx: Difference between revisions

Back up to the VoIP Configuring page
From AAISP Support Site
(First draft of page)
 
m ('IPv6 Only' works fine)
(13 intermediate revisions by the same user not shown)
Line 1: Line 1:
<indicator name="VoIPConfiguring">[[File:menu-voip.svg|link=:Category:VoIP Phones|30px|Back up to the VoIP Configuring page]]</indicator>
<indicator name="VoIPConfiguring">[[File:menu-voip.svg|link=:Category:VoIP Phones|30px|Back up to the VoIP Configuring page]]</indicator>
[[file:Grandstream_801.png]]
[[file:Grandstream_HT801.png|300px]]




Line 16: Line 16:
|}
|}


This is based on the author's experience of the HT801 (bought new in Sep 2018), which is a small unit (roughly 4" x 4" x 1") which provides 1 FXS port and no FXO ports. It does IPv6.
*See https://www.ukvoipforums.com/viewtopic.php?t=1061 for UK settings
Similar units are the HT802 (which has 2 FXS ports), and the HT81x series (which adds a NAT router, and is available with different
numbers of FXS ports).


If supplied without a Quick Start Guide, connect to a router which acts as a DHCP server and connect a telephone to the unit. Then dial '***' to access the internal Interactive Voice Response (IVR) system. '02' will read out the unit's IP address(es). Point a web browser at one of those addresses, and login as 'admin','admin'.

*''Basic Settings'', ''Internet Protocol'', select ''Both, prefer IPv6'' (or, if you're brave, select ''IPv6 Only'')
*See https://www.ukvoipforums.com/viewtopic.php?t=1061 for UK settings (although you might want to set the ''NTP Server'' to ''time.aa.net.uk'' as that has IPv6 addresses)
*Can support pulse dialling handsets - ''FXS Port'', ''Enable Pulse Dialing'', select ''Yes''. As a bonus, this allows a pulse dialling handset to interact with IVR systems - at least until the system asks you to press '*' or '#'.
*Can send BT CLID to handsets, see UK settings link above

A couple of personal favourite config options:
* There's a confusingly explained option under ''FXS Port'' labelled ''Use # as Dial Key'', explained as ''if set to Yes, "#" will function as the "(Re-)Dial" key''. If set to ''No'' the # key works as a normal key, if set to ''Yes'' it acts as a special key variously named ''Send'' or ''Dial'' which can be used as ''Redial last number called'' - but that'll be the last number dialled by the unit, not the last number dialled by the phone concerned. I set it to ''No'' to avoid surprises.
* Also under ''FXS Port'' are some options regarding Hook Flash. This corresponds to what the UK calls ''Timed Break Recall''. I set the option ''Enable Hook Flash'' to ''No'' - otherwise you can think you've hung up on a call by tapping the hook switch but you haven't (and your phones ring to remind you - so you pick the handset up and find the other party is confused too).

You can set up distinctive ring tones based on the calling number, and it even allows matching on part of a number. I use '020836xxxxx' to match calls from a number of exchanges on the northern edge of London (which is where your author lives).

==IPv6 Only==
I finally bit the bullet and configured the unit for IPv6 only. It needs to know about IPv6 DNS and NTP servers, and needs to use an IPv6 syslog server, but it just works.
==Firewall & Security==
==Firewall & Security==
*You will also want to set up firewall rules, as per the [[VoIP Firewall]] page.
*You will also want to set up firewall rules, as per the [[VoIP Firewall]] page.
*Also see the [[VoIP Security]] page for information about securing your VoIP service.
*Also see the [[VoIP Security]] page for information about securing your VoIP service.
*You can have an IP address whitelist and blacklist for web, telnet, and ssh access to the ATA. Unfortunately it only accepts IPv4 address ranges (I have a support ticket open with Grandstream about this). ''Basic Settings'', ''Web/SSH Access'', ''White List for WAN Side'', similar for blacklist.


[[Category:VoIP Phones|Grandstream]]
[[Category:VoIP Phones|Grandstream]]

Revision as of 16:38, 29 November 2018

Grandstream HT801.png


Feature Notes
Supports 302 Redirect ?
Tested on FireBrick SIP Server No
IPv6 Support Yes

This is based on the author's experience of the HT801 (bought new in Sep 2018), which is a small unit (roughly 4" x 4" x 1") which provides 1 FXS port and no FXO ports. It does IPv6. Similar units are the HT802 (which has 2 FXS ports), and the HT81x series (which adds a NAT router, and is available with different numbers of FXS ports).

If supplied without a Quick Start Guide, connect to a router which acts as a DHCP server and connect a telephone to the unit. Then dial '***' to access the internal Interactive Voice Response (IVR) system. '02' will read out the unit's IP address(es). Point a web browser at one of those addresses, and login as 'admin','admin'.

  • Basic Settings, Internet Protocol, select Both, prefer IPv6 (or, if you're brave, select IPv6 Only)
  • See https://www.ukvoipforums.com/viewtopic.php?t=1061 for UK settings (although you might want to set the NTP Server to time.aa.net.uk as that has IPv6 addresses)
  • Can support pulse dialling handsets - FXS Port, Enable Pulse Dialing, select Yes. As a bonus, this allows a pulse dialling handset to interact with IVR systems - at least until the system asks you to press '*' or '#'.
  • Can send BT CLID to handsets, see UK settings link above

A couple of personal favourite config options:

  • There's a confusingly explained option under FXS Port labelled Use # as Dial Key, explained as if set to Yes, "#" will function as the "(Re-)Dial" key. If set to No the # key works as a normal key, if set to Yes it acts as a special key variously named Send or Dial which can be used as Redial last number called - but that'll be the last number dialled by the unit, not the last number dialled by the phone concerned. I set it to No to avoid surprises.
  • Also under FXS Port are some options regarding Hook Flash. This corresponds to what the UK calls Timed Break Recall. I set the option Enable Hook Flash to No - otherwise you can think you've hung up on a call by tapping the hook switch but you haven't (and your phones ring to remind you - so you pick the handset up and find the other party is confused too).

You can set up distinctive ring tones based on the calling number, and it even allows matching on part of a number. I use '020836xxxxx' to match calls from a number of exchanges on the northern edge of London (which is where your author lives).

IPv6 Only

I finally bit the bullet and configured the unit for IPv6 only. It needs to know about IPv6 DNS and NTP servers, and needs to use an IPv6 syslog server, but it just works.

Firewall & Security

  • You will also want to set up firewall rules, as per the VoIP Firewall page.
  • Also see the VoIP Security page for information about securing your VoIP service.
  • You can have an IP address whitelist and blacklist for web, telnet, and ssh access to the ATA. Unfortunately it only accepts IPv4 address ranges (I have a support ticket open with Grandstream about this). Basic Settings, Web/SSH Access, White List for WAN Side, similar for blacklist.