Alphatech BlueGate SIP 1 Assembly instructions

ALPHATECH TECHNOLOGIES s.r.o.
Jeremenkova 88, Praha 4, Czech Republic
www.alphatechtechnologies.cz
BlueGate SIP 1
VoIP GSM GATE
Installation and setup guide V 1.0

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1. General description
1.1 Technical parametres
Dimensions 100 x 130 x 37 mm
Weight 850 g
Operating position various
Operating condition temperature: +5°C ÷ +40° C
humidity: 10% ÷ 80% při 30° C
Power supply 9-15V dc or 8-12 Vac, 1.5 A
VoIP
Ethernet –10/100Mb with standard BaseT and 100BaseTx,
connector RJ45
SIP connection P2P or IP PBX net unit –tested with Cisco
Call Manager, Alcatel OMNI PCX, Asterisk, Nexspan,
Panasonic…
1 VoIP channel (1 IP adress)
Codecs: G711u, G711a, G726, GSM
VAD (Echo cancellation)
Protocols: IP, TCP, UDP, HTTP, TELNET, SIP, RTP
Web server for remote management –BOA
Software for internet phone (VoIP) –Linphone
WEB –firmware upgrade
WEB –interface for parametres programming
Connection with DTMF dial in
connection directly to the operator or designated line
connection to operator after time out (adjustable) for direct
dialing in
transmit of calling part number (CLIP)
incoming call restriction from GSM network
outgoing call restriction from GSM network
Smart Callback –automatic incoming calls routing up CLIP
Echo canceller –switching ON/OFF
PIN protection of SIM card

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GSM
GSM 900 (class 4 –2 W)
GSM 1800 (class 1 –1 W)
Antenna connector SME/SMA, 50Ω
SIM card: 3/1,8 V
1.2 Installation
Insert SIM cards.
Before SIM card inserting check their status in mobile phone
(starting with PIN or without PIN). When you want to use the
starting with PIN you must firstly program this PIN to Blue
Gate SIP. Without this setting the Blue Gate SIP will not
registrate to GSM network!
SIM holders will be released after
pressing of yellow button
Connect GSM antenna

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Connect the LAN
cable
Finally, connect
the power
adapter
Error status indication:
LED is off –stand by
LED flashing –failure
Incoming and outgoing call
GSM operation:
LED flashing –modul registrate to
GSM net, stand by
LED lighting –connection (call)

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On PC which is connected to LAN, you can control connected
Gateway via various web broser. Insert IP address of gate (default
192.168.1.250) and you can continue in setting of parametrs.
1.3 DIP switch
The program reads the DIP Switch state after
restart. The DIP switch has 4 switches.
1. switch is for default IP address. For normal
Gateway operation set this switch to ON position.
2. switch is for default VoIP parameters. For normal
Gateway operation set this switch to ON position
3. switch for P2P or SIP Server mode. For SIP
Server mode set the switch to off position, for P2P
set the switch to on position
4. switch is for default GSM parameters. For
normal Gateway operation set this switch to ON
position.
Note: ON position is on the right side

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1.4 choosing a mode and login
It is important to choose a gate mode first. The gate can work in the
PeerToPeer mode or SIP server mode. The mode setting can be
made by a relevant switch. In the SIP server mode is possible to
choose internal or external SIP server. It can be set in a
configuration interface of the gate.
In your web browser enter IP address of the gate, default is
192.168.1.250. Enter user name and password. User name is
admin, default password is 1234.

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1.5 language option
Language - can be chosen in the menu on the left panel.

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2. Gateway setup
After making changes click on a save and restart or save changes
button. For default data restoring click on a default values button.
2.1 Network setting
Setup via DHCP - it is possible to use DHCP service or you can
enter IP address manualy.
DHCP client ID - host identification for DHCP server address
allocation
IP address - address of GSM gates. In P2P mode make a call to
this address. If use internal or external SIP server, make calls to
address of SIP server.
IP address SIP server –is used for internal SIP server. If it’s
enabled all clients (phones) make calls via this address.
Network gateway and DNS - servers are needed only if you make
connections or register via internet. If GSM Gateway is used only
on local network, leave them empty. Network gateway is not
address of internal or external SIP server usually.

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2.2 GSM Gate
DISA - if a number of digits in the direct dialling code is greater
than zero, the callers may dial directly using tone selection to the
required extension.
Wait - sets, how many seconds the automatic operator waits for
direct dialling code selection. If it is not selected, the pre-selected
number of the operator is dialed again.
Erase Clip - parameter for deleting the initial digits from CLIP (e.g.
deleting +42 in order not to "confuse" LCR exchanges when calling
back to the displayed number).
Echo canceller - this feature restricts ECHO.
Add zero before CLIP - it adds 0 before each incomming number.

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2.3 GSM Gate - module
PIN - set up the PIN of SIM card in modul. When you use SIM card
without PIN then this window is not used..
Subscriber number - here is set a customer number of operator
which will be dialed by Gateway at an incoming calling.
Volume GSM, Volume VoIP - allows call volume adjusting in both
direction.
Incoming calls, Outgoing calls - permission of
incoming/outgoing calls
Smart callback - store all outgoing calls which has been missed
or refused. When called part is calling back then the call is
automatically routed to extension which made the call. When
connection was succesfull data are erased from memory.
0, CLIR - it adds 0 or code for Switch OFF outgoing CLIP before
each outgoing dialled number
Roaming - enable Roaming.

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2.4 Accepted calls
If the table is not filled, all calls are enabled. Up to 12 blocks of
permitted calls can be selected. 1- up to 8-digit numbers can be
written in the first column of the table. The program substitutes any
digits instead of not filled digits from the right when is checking the
enabled calls. This means, instead of writing the numbers
601,602,…609, it is possible to write only 60. If the operator has
three successive numbers differing from each other only by the last
digit, this calls can be written as a group on a single line: the first
column contains up to 8-digit number of the first call (“from”) as in
the case mentioned above and the last digit of the last call is stored
in the second column (i.e. the last digit of “to”). A group may be
created only within the range of one order of the last position - i.e.
from 0 to 9!

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2.5 GSM Gate status
Operator –name of network operator
Signal strength - 113 to -99 dBm is very bad signal, -98 to -83
dBm is bad signal, -82 to -71 dBm is good signal and -70 to -51
dBm is very good signal.
Bit error rate - less value means the signal is better.
IMEI - unique number assigned by GSM part producer.
GSM Firmware version - name of SW version for GSM part.

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2.6 Number translation
The gate must be set to the P2P mode. In this mode it is important
to set a translation of numbers into IP addresses. Item „Number“ is
a phone number, IP address is an address of the selected phone.

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2.7 SIP parameters
The module must be set to the mode with SIP server. Next, internal SIP
server must be deactivated. After the saving of entered data, registration
attempt is executed (if name is not empty) and result is shown. If the
registration is not successfull, you will see the reason in Registration log in
menu Services
SIP proxy server - enter IP address or the name and port (if differ from
default 5060) of SIP server to send registration or route calls.
SIP registrar server - it isn't necessary fill if it is identical with SIP proxy
server
Name and Password - are not mandatory but must be set exactly same
as on SIP server machine.
Use (180 Ringing) - signalling of incomming call by default Ringing.
Use (183 Session progress) –the change on Session progress is
added for SIP proxy servers which require it.
Enable Simmetric RTP - added for SIP proxy servers which require it.
Send DTMF as SIP INFO - Outband DTMF as SIP INFO.
Send DTMF according to RFC2833 - Outband DTMF in "RTP channel
".

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2.8 SIP Server (internal)
A module must be set to the mode with SIP server.
Enable SIP server - activation/deactivation of the internal SIP server
Server name (realm) - the name is sent from SIP server to client as the
SIP server name at make registration or call.
Prefix - 1 to 3 digit number which is prepended before called GSM
number from IP phone. If leave it empty, call via appropriate GSM
Module will not be possible.
Client numbers and passwords - must be entered the same as in client
settings, in other case client will not be able to register itself on internal
SIP server and make a call.

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2.9 Setting audio
In case of poor audio quality try another codec priority. In menu
'Setting audio' choose G711µ as first, G711a as second, ... Check
client (IP Phone ) audio settings. At least one audio codec must be
chosen.

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2.10 User interface
Web interface tcp port - possibility change Web interface network
port.
Enable telnet - possibility disable / enable access over telnet.

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2.11 Admin services
VoIP version –name of SW version for VoIP part
Download log file - if you are in troubles and need technical
support, you will need this file. Follow these steps:
1. press 'start enhanced log' button
2. do the problematic action, this action will be recorded step
by step into log file
3. click on 'Download log file', save the file and send it to
technical support.
Show call log - record of few last incomming and outgoing calls.
In case of error, reason is also shown.
Show register log - registration on SIP server, in case of error
reason is shown. Registration is always done in two steps. Firstly a
client sends the request to server and server responds with its
realm, in second step the client sends identity based on realm of
server and server responds with success or access denied.
Show VoIP log - start VoIP monitor (online log file).
Time server - IP adress of NTP server with time data. (If doesn't
know address of NTP, you use * and system automatic choose
acceptable.)
Syslog server - computer able of receiving internal messages
from device
Firmware upgrade - customer is allowed to load new versions of
firmware into device. Here you also load customizations. If
upgrading firmware is in progress do not power off device, you risk
the device gets unusable! After firmware upgrade the pressing of
reset button is allways required.
Upload language - possibility to supply custom language. Name
of this file appears in Language combo box. HTML entities in file
name are allowed. Language file must start with line cs or en - the
language from which thist customization is derived.
Save configuration - save actual setting of all gateway
parameters to file
Upload configuration - restores setting of all gateway parameters
from file.
Service password - possibility to change the manufacturer default
password.

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2.12 Help
Here you can find informations about Gateway configuration and
problem solving

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3. Troubleshooting
Many of problems is possible to successfully solve by cooperation
with technical support. In this case tech. support needs exact and
clean description of your problem and log file downloadable from
menu Support. Follow these steps:
1. press 'start enhanced log' button
2. do the problematic action, this action will be recorded step
by step into log file
3. click on 'Download log file', save the file and send it to
technical support
3.1 Registration
registration is unsuccessful
- go to menu Service and click 'Show register log', record of
registration attempts and results will appear
registration is unsuccessful - in registration log is reason:
Timeout
- SIP server is unreachable, check SIP server address setting
in menu 'SIP parameters'
- check network connection and SIP server running
registration is unsuccessful - in registration log is reason: 404
(Not found)
- check IP address of SIP server, port and registration name
in menu 'SIP parameters'
registration is unsuccessful - in registration log is reason:
'Unauthorized' or 'Access denied'
- registration is allways done in two steps, in first step client
obtain server's realm and result is 'Unauthorized', in second
step client sends authorisation and should be successful.
Two step registration is OK.
- registration name (number) and password are not
mandatory, but must be set exactly same as on SIP server
- look into SIP server machine log, you may find interesting
things