Mobile VoIP -2

User Manual
 【Content】

[Link]................................................................................................................. 1

[Link] DESCRIPTION............................................................................................... 1

[Link] LIST.......................................................................................................................... 1

[Link] ......................................................................................................................... 2

[Link] OF THE DEVICE.................................................................................................. 3

[Link] .............................................................................................................................. 4

[Link] PAGE SETTING......................................................................................................... 5

[Link] INFORMATION.................................................................................................. 6

9. ROUTE.................................................................................................................................. 7

[Link] ............................................................................................................................ 12

[Link]........................................................................................................................ 15

[Link] SETTING.................................................................................................................... 19

13. NAT TRANS..................................................................................................................... 27

[Link] AUTH................................................................................................................ 28

[Link] CHANGE................................................................................................................ 29

[Link] ............................................................................................................................ 30

[Link]............................................................................................................................ 32

18. IP SETTING ..................................................................................................................... 33

[Link] ............................................................................................................. 35

[Link] MOBILE VOIP-2 WITH ASTERISK............................................................... 36

21. SIMPLE STEPS ............................................................................................................... 36

[Link]
Mobile VoIP -2 is a 2 channels VoIP GSM Gateway for call termination
(VoIP to GSM ) and origination (GSM to VoIP). It is SIP based and
compatible with Asterisk. It can enable to make 2 calls simultaneously
from IP phones to GSM networks and GSM network to IP phone.

[Link] description
2.1 VoIP(SIP)、GSM(MOBILE VOIP) conversion.
2.2 50 sets of LAN->MOBILE routes setting，50 sets of MOBILE->LAN
routes setting.
2.3 Voice response for setting and status (dial in from mobile).
2.4 Series connections to save bills.
2.5 Standard SIP(RFC2543,RFC3261) protocol，
Communicates with other gateway or PC.

[Link] list
Please check the parts for any missing parts. If do, please contact
our agents：
3.1 「MOBILE VOIP-2」main body
3.2 Power adaptor AC-DC (110V AC – 12V DC) or (220V AC – 12V DC)
3.3 Network cable
3.4 Antenna
3.5 User Manual

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 (1)
(2)

(3) (4)

[Link]

4.1cm

17cm
14.5cm

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[Link] of the device

5.1

5.2 5.3 5.4 5.5 5.6 5.7 5.8

5.1 Antenna：Antenna connector.

5.2 DC 12V：Power input.
5.3 LAN：LAN port. It also can be DHCP Server.
5.4 WAN: RJ-45 internet connector，standard RJ-45 socket，connect to
HUB.
5.5 PWR (Power LED)：Light up when power is normal.
5.5 VoIP1：an indicator light of VoIP1
5.6 VoIP2：an indicator light of VoIP2
5.8 LINK Indicator：Light up when network is connected.

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[Link]
6.1 Connect the internet cable from HUB to the ‘WAN’ connector of the
Mobile VoIP-2.
*If you need to stack up more Mobile VoIP-2,you can stack up as
follows.

6.2 Connect the antenna and put it in proper position to get the best
signal reception.
6.3 Insert the SIM card from back of the main body. (take the slide off
first).

6.4 Connect the power adaptor. The ‘POWER’ LED should be light up.

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[Link] Page Setting
When the IP setting is done, the operator may setup all the rest
parameters via web page. Browse the IP address from Internet
Explorer (e.g. [Link] following page shows up：

Enter the username and password for authentication. (default

username=voip, password=1234). The page follows when the
username and password are correct.

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[Link] Information.
8.1 When you login the web page, you can see the demo system current
system information like firmware version, company… etc in this
page.
8.2 Also you can see the function lists in the left side. You can use
mouse to click the function you want to set up.

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9. Route
9.1 Mobile TO LAN Settings
The operator may assign 50 sets of routing rule to transfer the call
incoming from MOBILE to LAN.

The MOBILE VOIP will transfer to the URL according to the caller ID of
the Mobile.
*CID：
(1) may enter the whole number, e.g. 0911111111
(2) only part of the number (prefix) e.g. 0911* means any number
starting with 0911 will be accepted

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 (3) * means all numbers can be accepted
(4) N means the calls without the CID
Please note the priority of the rules. The item which has more digits will
have higher priority. If the digits are the same, then former one gets the
higher priority.
*URL：The IP address to transfer this call
(1) may enter the whole IP address, e.g. [Link] or proxy
extension or phone number.
(2) If this field is blank or simply ‘N’, it means refuse to transfer.
(3) If an ‘*’ entered, it means 2-stages-dialing. The call will be
answered and prompt dial tone again to receive the IP address/sip
extension or any phone number as the destination. The caller may
enter the IP such as 192*168*0*101#.
*If the device have register proxy server/Asterisk ,you can enter any
destination phone number. Please note the proxy server/Asterisk
need to set the route of destination phone number.
Example:
(1) Mobile to Lan: 0932*,0911123456
Mobile VoIP have register proxy server/Asterisk
The proxy server/Asterisk have the route “09”
When the caller’s prefix number is 0932,Mobile VoIP will connect
0911123456 automaticlly
(2) Mobile to Lan: *,*
Any caller call the mobile voip’s sim,mobile voip will prompt dial
[Link] can enter IP or sip extension or phone number.
*sip extension or phone number both need to register SIP Proxy
Server or Asterisk.
*Phone number, SIP Proxy Server or Asterisk need to set the route
of this phone number.

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9.2 Mobile to LAN Speed Dial Settings
When you set Mobile to LAN Speed Dial Settings and Mobile to
LAN at the same time,Mobile VoIP-2 will give priority to Mobile to LAN
Speed Dial Settings.

*The call will be answered and prompt dial tone again. When the caller
may enter the “Num”, system will connect the “URL” as destination.

E.g Num:0 Name:test URL:[Link]

When the caller hear dial tone and enter 0, system will connect
[Link]

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9.3 LAN to Mobile Settings
The operator may assign 50 sets of routing rule to transfer the call
incoming from LAN to MOBILE.

The MOBILE VOIP will transfer to the mobile number according to the
incoming URL
*URL：The IP address of the incoming call.
may enter the whole IP address, e.g. [Link] or proxy server’s
extension. If a simple ‘*’ is entered, means no restriction for the
incoming IP address.

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*Call Num：
[Link] enter the whole number, e.g. 0911111111
2.a simple *”means 2-stages-dialing. The call will be answered and
prompt dial tone again to receive the called number as the
destination, e.g. 0911111111 or 0911111111#
3.#['d'n]['a'ppp] for one-stage dialing
[...] is option
'd'n means to delete the beginning n codes,
'a'ppp means to add 'ppp' in front.
for example #d2a09 means one-stage dialing,
delete the first 2 codes from your destination number,
then add 09 in front as the new destination number.
Example:
Lan to Mobile: *, #
(1)Mobile VoIP and Lan Phone both need to register proxy server or Asterisk.
(2)Proxy server/asterisk set the route that the prefix of destination number
(3)When you dial any destination phone number from lan phone,Mobile VoIP will
connect this call auto.

Example of Application:
When you call the ch.1 Mobile VoIP-2 gsm number,it will provide dial tone and you
enter a destination number.
Then ch.2 Mobile VoIP-2 will dial this number and connect.
ch.1 Mobile VoIP-2: mobile to lan set route table *,*
ch.2 Mobile VoIP-2:lan to mobile set route table *,#
Additionally, two channels Mobile VoIP-2 both need to register proxy server or
Asterisk.
And proxy server/asterisk set the route that the prefix of destination number dial out
from ch.2 Mobile VoIP-2.
*The channel 2 Mobile VoIP-2's ip: the first ip + :5062 (e.g [Link]

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[Link]
10.1 Mobile Status

(1)Network Registration：The telecom carrier which the SIM card been

registered.
(2)SIM Card ID：SIM card ID.
(3)Signal Quality：Signal quality.
(4)Incoming IP：The IP address of the last incoming call from LAN.
(5)Incoming IP Name: proxy server name
(6)Outgoing IP：The IP address of the last outgoing call to LAN.
(7)Incoming Mob：The caller ID of the last incoming call from MOBILE.
(8)Outgoing Mob：The called number of the last outgoing call to MOBILE.

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10.2 Mobile Setting

(1) (2)
(3)

(4) (5)
(6)
(7)
(8)
(9)
(10)

Mobile 1:
LAN (5)Rx
VoIP Codec GSM
(4) Tx

DTMF
(1)VoIP Tx Gain Mobile 2:
Rx
(2) VoIP Rx Gain Codec GSM
Tx

DTMF

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(3)LAN Dialtone Gain: DTMF Reciver is not good,you can adjust gain
down.
(4)CODEC Tx Gain: as above
(5)CODEC Rx Gain: as above
(6)Caller ID: You may select to display the Caller ID from GSM incoming
call, or fixed SIP user name.
(7)Presentation CLIR : If you need to block the Caller Id for call
termination,please choose Suppression
(8)Mobile PIN Code:If you need to unlock pin code via MOBILE
VOIP,you can click “On” and enter pin code.
(9)LAN Answer Mode:
Answered : when mobile answer,then connect the call
Alerted : when the mobile is ringing back tone,then connect the call
Income : when lan dial out,then connect soon
(10)Band Type:When you buy Quad band,you need to choose your
GSM frequency

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[Link]
In Network you can check the Network status, configure the WLAN
Settings , LAN Setting and SNTP settings.

11.1 Network Status: You can check the current Network setting in this
page.

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11.2 WAN Settings: You can check the current Network setting in this
page.
(1) The TCP/IP Configuration item is to setup the WAN port’s network
environment. You may refer to your current network environment to
configure the system properly.
(2) The PPPoE Configuration item is to setup the PPPoE Username and
Password. If you have the PPPoE account from your Service
Provider, please input the Username and the Password correctly.
(3) The Bridge Item is to setuo the system Bridge mode Enable/Disable.
If you set the Bridge On, then the two Fast Ethernet ports will be
transparent.
(4) When you finished the setting, please click the Submit button.

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11.3 LAN Settings: You can check the current Network setting in this
page.
(1) The TCP/IP Configuration item is to setup the WAN port’s network
environment. You may refer to your current network environment to
configure the system properly.
(2)DHCP Server: You may refer to your current network environment to
configure the system properly

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11.4 SNTP Settings:

SNTP Setting function: you can setup the primary and second SNTP
Server IP Address, to get the date/time information. Also you can base
on your location to set the Time Zone, and how long need to synchronize
again. When you finished the setting, please click the Submit button.

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[Link] Setting
In SIP Setting you can setup the Service Domain,Port Settings,Codec
Settings,RTP setting,RPort Setting and Other SettingS. If the VoIP
service is provided by ISP,you need to setup the related informations
correctly then you can register to SIP Proxy Server correctly.

12.1 In Servcie Domain Function you need to input the account and the
related informations in this page,please refer to your ISP Provider.
You can register three SIP accounts . You can dial the VoIP phone
to your friends via first enable SIP account and receive the phone
from the tree SIP account.

First you need to click Active to enable the Service Domain,then you can
input the following items.
(1)No.,: choose Mobile 1 or Mobile 2
(2) Display name: you can input the name you want to display.
(3) User name: you need to input the User Name get from your ISP.
(4) Register Name: you need to input the Register Name get from your
ISP.
(5) Register Password: you need to input the Register Password get
from ISP.
(6) Domain Server:you need to input the Domain Server get from your
ISP.
(7) Proxy Server:you need to input the Proxy Server get from your ISP.
(8) Outbound Proxy: you need to input the Outbound Proxy get from your
ISP. If your ISP does not provide the information,then you can skip
this item.
(9) You can see the Register Status in the Status item.
(10) When you finished the setting,please click the Submit button.
Remember to click “Save Charge”

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Example:
Register VoipBuster

Your Voipbuster username

Your Voipbuster password

Proxy Server’s IP

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12.2 Port Setting
You can setup the SIP and RTP port number in this page. Each ISP
provider will have different SIP/RTPport setting, please refer to the ISP
to setup the port number correctly. When you finished the setting, please
click the Submit button.

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12.3 Codec Settings:
You can setup the Codec priority, RTP packet length in this page. You
need to follow the ISP suggestion to setup these items. When you
finished the setting, please click the Submit button.

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12.4 Codec ID Setting
You can setup the Codec ID in this page.

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12.5 DTMF Setting
You can setup the DTMF Setting in this page.

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12.6 RPort Function:
You can setup the RPort Enable/Disable in this page. To change this
setting, please following your ISP information. When you finished the
setting, please click the Submit button.

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12.7 Other Settings
Other Settings: you can setup the Hold by RFC and QoS in this page. To
change these settings. please following your ISP information. When you
finished the setting, please click the Submit button. The QoS setting is to
set the voice packets’ priority. If you set the value higher than 0, then the
voice packets will get the higher priority to the Internet. But the QoS
function still need to cooperate with the others Internet devices.

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13. NAT Trans
In NAT Trans. you can setup STUN and uPnP function. These functions
can help your VoIP device working properly behind NAT.

13.1 STUN Setting: you can setup the STUN Enable/Disable and STUN
Server IP address in this page. This function can help your VoIP
device working properly behind NAT. To change these settings
please following your ISP information. When you finished the
setting, please click the Submit button.

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[Link] Auth.
In System Authority you can change your login name and password.

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[Link] Change
In Save Change you can save the changes you have done. If you want to
use new setting in the VoIP system, You have to click the Save button.
After you click the Save button, the system will automatically restart and
the new setting will effect.

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[Link]
In Update you can update the system’s firmware to the new one or do the
factory reset to let the system back to default setting.

16.1 Update firmware

(1) In New Firmware function you can update new firmware via HTTP in
this page. You can upgrade the firmware by the following steps:
(2)Select the firmware code type, Risc code.
(3)Click the “Browse” button in the right side of the File Location or you
can type the correct path and the filename in File Location blank.
(4)Select the correct file you want to download to the system then click
the Update button.

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16.2 Restore Default Settings

Default Setting you can restore the system to factory default in this page.
You can just click the Restore button, then the system will restore to
default and automatically restart again.

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[Link]
Reboot function you can restart the system. If you want to restart the
system, you can just click the Reboor button, then the system will
automatically.

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18. IP Setting
The operator can setup or query the network parameters by dialing in the
mobile number which it SIM card has been put in the main body. The
status or result is response by voice. In the first 20 seconds after
power-on, the VoIP GSM Gateway enters the IP setting mode. The
operator may dial in the mobile number during this period to set or query
the network parameters.
Item IVR Action IVR Menu Choice Notes
1 Reboot #195# After you hear “Option
Successful,” hang-up. Unit will
reboot automatically.
2 Factory Reset #198# System will automatically
[Link]: ALL
User-Changeable”
NONDEFAULT SETTINGS
WILL BE LOST! This will
include network and service
provider data.
3 Check IP Address #120# IVR will announce the current
IP
address , Default :
[Link]
4 Check IP Type #121# IVR will announce if DHCP in
enabled or disabled.
default : OFF
5 Check Network #123# IVR will announce the current
Mask network [Link] :
[Link]
6 Check Gateway #124# IVR will announce the current
IP gateway IP address,
Address Default : [Link]
7 Check Primary #125# IVR will announce the current

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 DNS Server setting in the Primary DNS
field.
Default : [Link]
8 Check Firmware #128# IVR will announce the version
Version of the firmware running
9 Set as DHCP #111# The system will change to
client DHCP
Client type
10 Set Static IP #112xxx*xxx*xxx DHCP will be disabled and
Address *xxx# system will change to the
Static IP type.
Enter IP address using
numbers on the telephone key
pad. Use the * (star) key when
entering a decimal point.
11 Set Network Mask #113xxx*xxx*xxx Must set Static IP first.
*xxx# Enter value using numbers on
the telephone key pad. Use
the * (star) key when entering
a decimal point.
12 Set Gateway IP #114xxx*xxx*xxx Must set Static IP first.
Address *xxx# Enter IP address using
numbers on the telephone key
pad. Use the * (star) key
when entering a decimal
point.
13 Set Primary DNS #115xxx*xxx*xxx Must set Static IP first.
Server *xxx# Enter IP address using
numbers on the telephone key
pad. Use the * (star) key
when entering a decimal
point.

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[Link]
19.1 Protocols
SIP (RFC2543,RFC3261)
19.2 TCP/IP
IP/TCP/UDP/RTP/RTCP/
CMP/ARP/RARP/SNTP
DHCP/DNS Client
IEEE802.1P/Q
ToS/DiffServ
NAT Traversal
STUN
uPnP
IP Assignment
Static IP
DHCP
PPPoE
19.3 Codec
G.711 u-Law
G.711 a-Law
G.723.1 (5.3k)
G.723.1 (6.3k)
G.729A
G.729A/B
19.4 Voice Quality
VAD
CNG

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 AEC, LEC
Packet loss
19.5 GSM (MOBILE VOIP)
Dual BAND: 900/1800 MHZ
Tri BAND: 900/1800/1900 MHZ
Quad BAND: 900/1800/1900/850 MHZ

[Link] Mobile VoIP-2 with Asterisk

Test version
trixbox-2.2

SIP Softphone

SJPhone 1.60.289a

X-Lite 1105x

Modify file

Add the following setting to/etc/asterisk/[Link]
[1000]
type=friend
secret=1000
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=internal

[1001]
type=friend
secret=1001
qualify=yes

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nat=yes
host=dynamic
canreinvite=no
context=internal

[1002]
type=friend
secret=1002
qualify=yes
nat=yes
host=dynamic
canreinvite=no
context=internal

Add the following setting to /etc/asterisk/[Link]

[internal]
exten => 1000,1,Dial(SIP/1000)
exten => 1001,1,Dial(SIP/1001)
exten => 1002,1,Dial(SIP/1002)

configure:
trixbox-2.2: address=[Link]:5060
SJPhone: address=[Link]:5060; username=1000,
displayname=user_1000
X-Lite: address=[Link]:7331; username=1001, displayname=user_1001
MOBILE VOIP-2: address=[Link]:5060; username=1002,
displayname=user_1002

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test1
pstn  call 0928492911(mobile number)  MOBILE VOIP-2  hear the second dial
tone,call SoftPhone’s number  SoftPhone  show pstn caller id

This Is X-Lite receiving packet, red word is pstn number. Test ok.

INVITE sip:1001@[Link]:7331 SIP/2.0

Via: SIP/2.0/UDP [Link]:5060;branch=z9hG4bK3d0bbaf7;rport
From: "035678238" <sip:1002@[Link]>;tag=as580472a7
To: <sip:1001@[Link]:7331>
Contact: <sip:1002@[Link]>
Call-ID: 20fa417265e6a26d0b0aae4f551f06f3@[Link]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 May 2007 [Link] GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

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Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2737 2737 IN IP4 [Link]
s=session
c=IN IP4 [Link]
t=0 0
m=audio 15852 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

SIP/2.0 200 Ok
Via: SIP/2.0/UDP [Link]:5060;branch=z9hG4bK3d0bbaf7;rport
From: "035678238" <sip:1002@[Link]>;tag=as580472a7
To: <sip:1001@[Link]:7331>;tag=677373503
Contact: <sip:1001@[Link]:7331>
Call-ID: 20fa417265e6a26d0b0aae4f551f06f3@[Link]
CSeq: 102 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105x
Content-Length: 254

v=0
o=1001 4804366 4807851 IN IP4 [Link]
s=X-Lite
c=IN IP4 [Link]
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000

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a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

test 2
SoftPhone  call 1002  MOBILE VOIP-2  hear second dial tone and call pstn 
pstn answer  show caller id-mobile number 0928492911

This Is X-Lite receiving packet. Test ok.

INVITE sip:1002@[Link] SIP/2.0

Via: SIP/2.0/UDP
[Link]:7331;rport;branch=z9hG4bK4C4315351FC84CA582D14FB8C25F
C3BF
From: user_1001 <sip:1001@[Link]:7331>;tag=1121869743
To: <sip:1002@[Link]>
Contact: <sip:1001@[Link]:7331>
Call-ID: F4B32CA6-1835-4E68-941A-C685B39C43FF@[Link]
CSeq: 63148 INVITE
Proxy-Authorization: Digest
username="1001",realm="asterisk",nonce="0d3b2879",response="8aaaaa5b5ad53
654bf0a2ab0fa9bb118",uri="sip:1002@[Link]",algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 254

v=0
o=1001 5111461 5111501 IN IP4 [Link]
s=X-Lite

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c=IN IP4 [Link]
t=0 0
m=audio 8000 RTP/AVP 0 8 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

SIP/2.0 200 OK
Via: SIP/2.0/UDP
[Link]:7331;branch=z9hG4bK4C4315351FC84CA582D14FB8C25FC3BF
;received=[Link];rport=7331
From: user_1001 <sip:1001@[Link]:7331>;tag=1121869743
To: <sip:1002@[Link]>;tag=as2a2fbf98
Call-ID: F4B32CA6-1835-4E68-941A-C685B39C43FF@[Link]
CSeq: 63148 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1002@[Link]>
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2737 2737 IN IP4 [Link]
s=session
c=IN IP4 [Link]
t=0 0
m=audio 13798 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

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a=fmtp:101 0-16
a=silenceSupp:off - - - -

register issue
The packet date from Asterisk as follows.
Please note, user_1002’s display name don’t appear
So the website’s Display Name is not available

<-- SIP read from [Link]:5060:

REGISTER sip:[Link] SIP/2.0
Via: SIP/2.0/UDP
[Link]:5060;rport;branch=z9hG4bK590e92b551233a10a0ae71944c19b5
aa
From: <sip:1002@[Link]>;tag=4e36d8f1
To: <sip:1002@[Link]>
Call-ID: 7e45b773130f1fc945efcee502f84042@[Link]
Contact: <sip:1002@[Link]:5060>
CSeq: 10 REGISTER
Expires: 300
Authorization: Digest
username="1002",realm="asterisk",nonce="3ca93a1e",response="4d39ccb0dae64
bb2f1341e9896ac1ea7",uri="sip:[Link]",algorithm=MD5
User-Agent: CMI CM5K
Content-Length: 0

--- (11 headers 0 lines) ---

Using latest REGISTER request as basis request
Sending to [Link] : 5060 (NAT)
Transmitting (NAT) to [Link]:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
[Link]:5060;branch=z9hG4bK590e92b551233a10a0ae71944c19b5aa;rec

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eived=[Link];rport=5060
From: <sip:1002@[Link]>;tag=4e36d8f1
To: <sip:1002@[Link]>
Call-ID: 7e45b773130f1fc945efcee502f84042@[Link]
CSeq: 10 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1002@[Link]>
Content-Length: 0

---
Transmitting (NAT) to [Link]:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
[Link]:5060;branch=z9hG4bK590e92b551233a10a0ae71944c19b5aa;rec
eived=[Link];rport=5060
From: <sip:1002@[Link]>;tag=4e36d8f1
To: <sip:1002@[Link]>;tag=as13a32ae8
Call-ID: 7e45b773130f1fc945efcee502f84042@[Link]
CSeq: 10 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5def9231"
Content-Length: 0

---
Scheduling destruction of call
'7e45b773130f1fc945efcee502f84042@[Link]' in 15000 ms
asterisk1*CLI>
<-- SIP read from [Link]:5060:
REGISTER sip:[Link] SIP/2.0

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Via: SIP/2.0/UDP
[Link]:5060;rport;branch=z9hG4bK672fa67f59c2223275f5ee286d27597a
From: <sip:1002@[Link]>;tag=4e36d8f1
To: <sip:1002@[Link]>
Call-ID: 7e45b773130f1fc945efcee502f84042@[Link]
Contact: <sip:1002@[Link]:5060>
CSeq: 11 REGISTER
Expires: 300
Authorization: Digest
username="1002",realm="asterisk",nonce="5def9231",response="046a412f4e7ed4
e98fd507416994a80a",uri="sip:[Link]",algorithm=MD5
User-Agent: CMI CM5K
Content-Length: 0

--- (11 headers 0 lines) ---

Using latest REGISTER request as basis request
Sending to [Link] : 5060 (NAT)
Transmitting (NAT) to [Link]:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
[Link]:5060;branch=z9hG4bK672fa67f59c2223275f5ee286d27597a;recei
ved=[Link];rport=5060
From: <sip:1002@[Link]>;tag=4e36d8f1
To: <sip:1002@[Link]>
Call-ID: 7e45b773130f1fc945efcee502f84042@[Link]
CSeq: 11 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:1002@[Link]>
Content-Length: 0

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---
12 headers, 0 lines
Reliably Transmitting (NAT) to [Link]:5060:
OPTIONS sip:1002@[Link]:5060 SIP/2.0
Via: SIP/2.0/UDP [Link]:5060;branch=z9hG4bK7b92dd8a;rport
From: "Unknown" <sip:Unknown@[Link]>;tag=as5dee3942
To: <sip:1002@[Link]:5060>
Contact: <sip:Unknown@[Link]>
Call-ID: 5ebc2211278e2cb7699911ad39454d4e@[Link]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 May 2007 [Link] GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Transmitting (NAT) to [Link]:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
[Link]:5060;branch=z9hG4bK672fa67f59c2223275f5ee286d27597a;recei
ved=[Link];rport=5060
From: <sip:1002@[Link]>;tag=4e36d8f1
To: <sip:1002@[Link]>;tag=as13a32ae8
Call-ID: 7e45b773130f1fc945efcee502f84042@[Link]
CSeq: 11 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 300
Contact: <sip:1002@[Link]:5060>;expires=300
Date: Tue, 22 May 2007 [Link] GMT
Content-Length: 0

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21. Simple Steps
Step 1. Change the Network setting if you need (Network/network setting)
Step 2. Register SIP proxy Server or Asterisk or VoipBuster if you need
(sip setting/service domain)
Step 3. Set Route ( request )
mobile to lan:
(1) *,* --->it is two stage dialing.
when mobile call in,MOBILE VOIP-2 will provide dial tone and you
can enter ip or asterisk extension or phone number.
* If you want to enter phone number,please note your asterisk need
to have route of destination number.
(2) *, specific extension or IP or phone number
when mobile call in,MOBILE VOIP-2 will connect with this specific
extension or IP or phone number auto
* If you want to set specific phone number,please note your asterisk
need to have route of destination number.

Lan to Mobile:
(1) *,* --->it is two stage dialing.
when lan phone call in,MOBILE VOIP-2 will provide dial tone and
you can enter mobile number.
(2) *, specific mobile number
when lan phone call in,MOBILE VOIP-2 will connect with the specific
mobile number auto.
(3) *,#--->It is 1 stage dialing
When lan phone and MOBILE VOIP-2 both register Asterisk,
you can dial any destination number from lan phone directly.
* Please note:Asterisk need to set route of destination number that
dial out from MOBILE VOIP-2
* All changes both need to click "save and change"

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