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For use G668 with asterisk you need:
Enter the superpassword in te webconf of phone.
Go to "protocol settings"
protocol: sip
service: inphonex
service addr: name/ip_of_your_asterisk
service id: your number
nat traversal : disable (if you do not need)
nat addr: empty
phone number: your number
acount: your username in sip.conf
pin: your password in sip.conf
register port: your sip port (default 5060)
signal port: 0
control port: 0
rtp port: 5040
local tipe: acount


with this setting the soyo g668 works

I don't have these working with Asterisk, but here is how you unlock the phone for sip configuration (existing configuration information will be lost):

1) Log into the phone using web browser (1234 is default password)
2) Input this ip address 165.236.243.102 where says "upgrade addr"
3) Click "Update", and wait for the phone boot up correctly
4) Press "Speaker", "12346", and then "Local IP". You should see the update running on the phone display.
5) Log back in again and you'll see all possible settings.
6) You'll probably want to put the ip address above in again; if you don't and you try to "upgrade", you'll be back to being locked in.

Now if I could only get the phone to register with asterisk...

I have tested a phone from www.vlines.de with asterisk (model VD120s running version 1.40 for IAX) with native IAX2 trunking to the PBX. I have no commercial connection with either asterisk or Vlines. Everything seemed to work OK, and the quality was fine. The manual was pretty poor, as the IAX version seems to be a hacked version that used to run SIP, so some of the settings are described a bit vaguely. Settings that I had to change: use service = yes. service addr = IP_address_of_your_asterisk_box. service_id = not_important. account= my_IAX_extension_number. pin=my_IAX_shared_secret. dtmf = 'control string'. codec1='gsm' codec2='g711a' codec3='g711u'

In iax.conf:
2200 ; 2200 matches my_IAX_extension_number set in the phone
host=dynamic
auth=md5
secret=my_IAX_shared_secret
context=local
type=friend
callerid=2200
mailbox=2200

in extensions.conf
; assuming that you have all IAX phones in the range 2200-2299
exten => 22XX,1,Dial(IAX2/${EXTEN}:@${EXTEN}/${EXTEN},15)
Hopefully this helps.

Does anyone have experience in configuring the VoIP phone MAXlink 201 to work with Asterisk?

Thank you.

-- sip.conf --

username1
context=default
context=nikotel-phil
type=friend
username=username1
callerid="Ulysses User"
secret=CapodociaTopSecret
host=dynamic
auth=md5
nat=yes
canreinvite=no
tmfmode=info
;dtmfmode=rfc2833
mailbox=2101
disallow=all
allow=g732.1
allow=g729
allow=ulaw
allow=alaw


-- extensions.conf --

exten => 2101,1,Dial(SIP/username1,20,rT)


-- Grandstream Budgetone 101 setup --

Software Version: Program--1.0.4.17 Bootloader--1.0.0.11 HTML--1.0.0.19

IP address: DHCP
SIP server: hostname.provider.com
Outbound proxy:
SIP User ID: username1
Authenticate ID: username1
pw: CapodociaTopSecret
Name: Ulysses User

vocoder choice 1: 729
vocoder choice 2: 723
vocoder choice 3: ulaw
vocoder choice 4: alaw

Early dial: No
Use # as dial key: Yes
local SIP port: 5060 (--> install port forwarding on your router if you can !)
local RTP port: 5004 (--> install port forwarding on your router if you can !)
Use random port: No
NAT traversal: Yes
STUN server: stun.xten.net
Voicemail user ID: 201 (any extension that leads to VoiceMailMain2(${CALLERIDNUM})
Send DTMF: via SIP INFO
DTMF payload: 101
Send Flash event: No
NTP server: time.nist.gov

About Swissvoice ip10s (MGCP version):
MGCP phone with graphical display (no backlight)

From what I found the MGCP channel supports the following service codes:

1. - blind transfer

FLASH - consultative transfer (buggy!)

• 67 - Calling Number Delivery Blocking
• 70 - Cancel Call Waiting
• 72 - Call Forwarding Activation
• 73 - Call Forwarding Deactivation
• 78 - Do Not Disturb Activation
• 79 - Do Not Disturb Deactivation
• 8 - Call pick-up

Notes:
- read the handbook draft offered by digium to learn about the three
different types of .conf files. Be aware that mgcp.conf does not work
the same way as sip.conf... so place your line= statement at the end...
- you must restart (!) Asterisk to activate any changes in mgcp.conf.
A simple "reload" will not do.
- the only way I found to set the "phone name" was to change the user's
display settings "idle text": that idle text IS the phone name it seems.
That'll matter for any TFTP files that you might plan to load at boot.

Question:
- does anyone have access to the "MGCP XML" guide or any description
of the other configuration files that can be loaded via TFTP or FTP?


Sample entry for mgcp.conf:

[192.168.0.123]]
context=from-mgcp
host=192.168.0.123
threewaycalling=yes
; note that transfer=yes requires threewaycalling
; transfer=yes permits FLASH transfers, assign FLASH to the key F1!
transfer=yes
; we have too much trouble with the ip10s and callwaiting
; so we better turn this feature off (sniff sniff)
callwaiting=no
; not sure what nat=yes actually does for MGCP devices
nat=no
; we prefer canreinvite=no so that ASTERISK can do codec translation
; the ip10s doesn't provide GSM, so we cannot talk directly to X-Lite
; using the GSM codec
canreinvite=no
callgroup=0,2-5
pickupgroup=0,1
cancallforward=yes
; not sure why anyone would want to use dtmf=inband ...
;dtmf=inband
callerid = Sven Svoboda
; if you configure the F4 key with the "voicemail" function then
; this key will light up in the event of vm box 5444 having msgs
; note, however, that you'll have to lift and replace the handset
; in order to update the blinking status of the F4 button
mailbox=5444
line => aaln/1


Sample extensions.conf entry:

exten => 1000,1,Dial(MGCP/aaln/1@192.168.0.123,20,rt)


Hope this helps,
Philipp von Klitzing