Unified Networks IP Stimulus (UNIStim) Channel Driver for Asterisk¶

This is a channel driver for Unistim protocol. You can use a least a Nortel i2002, i2004 and i2050.

Following features are supported :

Power on the phone
Wait for message "Nortel Networks"
Press quickly the four buttons just below the LCD screen, in sequence from left to right
If you see "Locating server", power off or reboot the phone and try again
DHCP : 0
SET IP : a free ip of your network
NETMSK / DEF GW : netmask and default gateway
S1 IP : ip of the asterisk server
S1 PORT : 5000
S1 ACTION : 1
S1 RETRY COUNT : 10
S2 : same as S1

The line=> entry in unistim.conf does not add an extension in asterisk by default. If you want to do that, add extension=line in your phone context.

If you have this entry on unistim.conf :

[violet]
device=006038abcdef
line => 102

then use:

exten => 2100,1,Dial(USTM/102@violet)

You can display a text with :

exten => 555,1,SendText(Sends text to client. Greetings)

You need to append /r to the dial string.
The first digit must be from 0 to 7 (inclusive). It's the 'melody' selection.
The second digit (optional) must be from 0 to 3 (inclusive). It's the ring volume. 0 still produce a sound.

Select the ring style #1 and the default volume :

exten => 2100,1,Dial(USTM/102@violet/r1)

Select the ring style #4 with a very loud volume :

exten => 2100,1,Dial(USTM/102@violet/r43)

You can use the following codes for country= (used for dial tone) - us fr au nl uk fi es jp no at nz tw cl se be sg il br hu lt pl za pt ee mx in de ch dk cn
If you want a correct ring, busy and congestion tone, you also need a valid entry in indications.conf and check if res_indications.so is loaded.
language= is also supported but it's only used by Asterisk (for more information see http://www.voip-info.org/wiki/view/Asterisk+multi-language ). The end user interface of the phone will stay in english.

Layout

 |--------------------|
 | 5 2 |
 | 4 1 |
 | 3 0 |

When the second letter of bookmark= is @, then the first character is used for positioning this entry
If this option is omitted, the bookmark will be added to the next available sofkey
Also work for linelabel (example : linelabel="5@Line 123")
You can change a softkey programmatically with SendText(@position@icon@label@extension) ex: SendText(@1@55@Stop Forwd@908)

This feature must only be used on a trusted network. It's very insecure : all unistim phones will be able to use your asterisk pbx.
You must add an entry called [template]. Each new phones will be based on this profile.
You must set a least line=>. This value will be incremented when a new phone is registered. device= must not be specified. By default, the phone will asks for a number. It will be added into the dialplan. Add extension=line for using the generated line number instead.

Example :

[general]
port=5000
autoprovisioning=yes

[template]
line => 100
bookmark=Support@123 ; Every phone will have a softkey Support

If a first phone have a mac = 006038abcdef, a new device named USTM/100@006038abcdef will be created.
If a second phone have a mac = 006038000000, it will be named USTM/101@006038000000 and so on.
When autoprovisioning=tn, new phones will ask for a tn, if this number match a tn= entry in a device, this phone will be mapped into.

Example:

[black]
tn=1234
line => 100

Use the two keys located in the middle of the Fixed feature keys row (on the bottom of the phone) to enter call history.
By default, chan_unistim add any incoming and outgoing calls in files (/var/log/asterisk/unistimHistory). It can be a privacy issue, you can disable this feature by adding callhistory=0. If history files were created, you also need to delete them. callhistory=0 will NOT disable normal asterisk CDR logs.

You can use the following entries in unistim.conf

It's not possible to connect a Nortel Succession/Meridian/BCM to Asterisk via chan_unistim. Use either E1/T1 trunks, or buy UTPS (UNISTIM Terminal Proxy Server) from Nortel.

As always, NAT can be tricky. If a phone is behind a NAT, you should port forward UDP 5000 (or change [general] port= in unistim.conf) and UDP 10000 (or change [yourphone] rtp_port=)
Only one phone per public IP (multiple phones behind the same NAT don't work). You can either :
- Setup a VPN
- Install asterisk inside your NAT. You can use IAX2 trunking if you're master asterisk is outside.
- If asterisk is behind a NAT, you must set [general] public_ip= with your public IP. If you don't do that or the bindaddr is invalid (or no longer valid, eg dynamic IP), phones should be able to display messages but will be unable to send/receive RTP packets (no sound)
Don't forget : this work is based entirely on a reverse engineering, so you may encounter compatibility issues. At this time, I know three ways to establish a RTP session. You can modify [yourphone] rtp_method= with 0, 1, 2 or 3. 0 is the default method, should work. 1 can be used on new firmware (black i2004) and 2 on old violet i2004. 3 can be used on black i2004 with chrome.
If you have difficulties, try unistim debug and set verbose 3 on the asterisk CLI. For extra debug, uncomment #define DUMP_PACKET 1 and recompile chan_unistim.