UR - Gateway User Guide
Product Description
This chapter mainly introduces functions and structures of UC2000-VE/F/G.
Overview
UC2000-VE/F/G serials GSM/CDMA/WCDMA/LTE VoIP Gateway is full functions VoIP gateway based on IP and Mobile network, which provides a flexible network configuration, powerful features, and good voice quality. It works for carrier grade, enterprise, SOHO, residential users for cost-effective solution.
Scenario of Application
With the development of users and telecom service, mobile network and fixed network integration will be steadily increasing. UC2000-VE/F/G provides high quality VoIP service which perfectly meets the requirement.
This is a scenario shown as figure 1-2-1
Figure 1-2-1 Network scenario
Product Appearance
Product Appearance of UC2000-VE
The appearance of UC2000-VE shows as follow
Figure 1-3-1 Front view of UC2000-VE-8G/8C/8W/8T
Table 1-3-1 Description of Front view
Index Indicators Description
1
RUN
On: Starting Off: Abnormal
Blinking every 0.5s: Normal status
2
PWR On: Power on
Off: Power off
3
Signal
Signal strength indicators with green color
4
Channel
Use/Unuse indicator with Red color, ON is used, Off is unused
5
SIM Slots
SIM card slot
Figure 1-3-2 Rear view of UC2000-VE-8G/8C/8W/8T
Table 1-3-2 Description of Rear view
Index Interface Description 1 Power Connector
Power connector of DC power. Input: DC12V
2 Antenna Connector Mark as digits 0 to 7 3 Network FE0 and FE1, its default IP address 192.168.11.1 4 Console RS232 standard, band rate 115200bps
5
RST
Reset button to restore default IP and password or restore factory setting.
w Restore IP and Password: hold RST button 3~5 seconds, RUN LED being ON during this time
w Restore factory setting: Hold RST button 7 seconds, RUN LED being blink
1.3.2 Product Appearance of UC2000-VF
Front View
Indicators and connectors
Indicators Name Status Description
SIM Card Status Indicator
OFF
Indicates SIM is offline, SIM status may include
SIM card not inserted, SIM card not
available, SIM card unregistered
ON
SIM card is in use
Blinking
SIM card is registered but in IDLE
Antenna
Connector
-
Antenna connect, mark with 0-15
SIM Card Slot
-
SIM card slot, mark with 0-15
Back view
Indicators and connectors
Indicators Name Status Description
Power switch
-
Power on or power off the device
Power connect
-
AC Input 110-240V FE0-FE1 Network - Default IP is 192.168.11.1
Console
-
RS232 standard, band rate 115200bps
RST
RST
- Reset button to restore default IP and password or restore factory setting. w Restore IP and Password: hold RST button 3~5 seconds, RUN LED being ON during this time Restore factory setting: Hold RST button 7 seconds, RUN LED being blink
PWR
Power indicator OFF No power ON Power on
RUN
System indicator Blinking (0.5S)
Device is running normally
ON Device is booting up OFF Device is not booting up
1.3.3 Product appearance of UC2000-VG
Front view
Indicators Name Status Description
SIM Card Status Indicator
OFF
Indicates SIM is offline, SIM status may include
SIM card not inserted, SIM card not
available, SIM card unregistered
ON
SIM card is in use
Blinking
SIM card is registered but in IDLE
Antenna
Connector
-
Antenna connect, mark with 0-15
SIM Card Slot
-
SIM card slot, mark with 0-15
Back View
Indicators Name Status Description
Power switch
-
Power on or power off the device
Power connect
-
AC Input 110-240V
FE0-FE1
Network
-
Default IP is 192.168.11.1
Console
-
RS232 standard, band rate 115200bps
RST
RST
- Reset button to restore default IP and password or restore factory setting. w Restore IP and Password: hold RST button 3~5 seconds, RUN LED being ON during this time Restore factory setting: Hold RST button 7 seconds, RUN LED being blink
PWR
Power indicator OFF No power ON Power on
RUN
System indicator
Blinking
(0.5S)
Device is running normally ON Device is booting up OFF Device is not booting up
1.4 Functions and Features
1.4.1 Protocols l Standard SIP;
l Simple Traversal of UDP over NATs (STUN);
l Point-to-point protocol over Ethernet (PPPoE);
l Hypertext Transfer Protocol (HTTP);
l Dynamic Host Configuration Protocol (DHCP);
l Domain Name System (DNS);
l ITU-T G.711α-Law/μ-Law、G.723.1、G.729AB;
l PPTP(support on 8 channels gateway)
1.4.2 System Function l PLC: Packet loss concealment
l VAD: Voice activity detection
l CNG: Comfort Noise Generation
l Local/Remote SIM card work mode
l Adjustable gain of port
l DTMF adjustment
l Balance Check
l Lock/unlock SIM/UIM
l Mobile number display rejection
l Sending/receiving SMS
l Customize IVR Recording
l White and black list
l One number access
l Open API for SMS, support USSD
l Echo Cancellation (with ITU-T G.168/165 standard)
l Automatic negotiate network
l Hotline
l BCCH(Support on GSM Gateway only)
1.4.3 Industrial Standards Supported l Stationary use environment: EN 300 019: Class 3.1
l Storage environment: EN 300 019: Class 1.2
l Transportation environment: EN 300 019: Class 2.3
l Acoustic noise: EN 300 753
l CE EMC directive 2004/108/EC
l EN55022: 2006+A1:2007
l EN61000-3-2: 2006,
l EN61000-3-3: 1995+A1: 2001+A2: 2005
l EN55024: 1998+A1: 2001+A2: 2003
l Certifications: FCC, CE
1.4.4 General Hardware Specification l Power Supply
UC2000-VE:
Input: 100-240V, 50-60Hz
Output: DC12V 4.0A UC2000-VF/G: Input: 100-240VAC, 50-60Hz;
l Temperature (Operation): 0 ℃ ~ 45 ℃
(Storage): -20 ℃ ~80 ℃
l Operation Humidity: 10%-90% No Condensation
l Dimension(W/D/H): 250*156*32.5mm
l Weight: 1.069kg
l Package Weight: 2.05kg
2 Quick Installation
2.1 Attentions before Installation Please pay attention to the following before you install UC2000-VE/F/G/T include:
l DC power/AC power should be grounded well to ensure reliability and stability l Network interface should be standard RJ45 with 10Mbps or 100Mbps interfaces l GSM channels work properly and antennas should be well connected.
2.2 Installation Procedures l Connect antennas to the device; l Connect the power wire to the device; l Connect network cable to the device; l Insert SIM cards to SIM slots.
2.3 Network Connection
PC
192.168.11.10 Switch/Router
FE0
192.168.11.1
Note: UC2000-VE/F/G/T has two Ethernet ports (namely FE0 and FE1). The device can work normally when either of the ports is connected to PC. The IP address of device must be at the same network segment with that of PC.
3 Basic Operation
3.1 Feature Codes Users can do some basic system setting via dialing feature codes through a telephone.
The device has a built-in IVR navigator for local maintenance. In each step, if you hear an IVR message of “setting succeeds”, it means you have finished this step successfully. However, if you hear “setting fails”, please check and redo that step.
Code Corresponding Function
- 150*
Dial *150*1 to set the IP address of the gateway as static IP address; dial *150*2 to set the IP address as DHCP IP address
- 152*
Dial *152*192*168*1*10# to set the IP address of the device as 192.168.1.10.
( 192.168.1.10 is just an example)
- 156*
Dial *156*192*168*1*1# to set the default gateway of the network as 192.168.1.1.
( 192.168.1.1 is just an example)
- 153*
Dial *153*255*255*0*0*# to set the netmask of the network as 255.255.0.0
(255.255.0.0 is just an example)
- 158#
Dial *158 to inquiry IP address of the device
- 111#
Dial *111# to restart the device
3.2 Basic Operation 3.2.1 Check IP address Use a mobile phone to call a SIM card number of the device, then the device will answer and play a voice prompt of ‘dial the extension number’. Press *158# on mobile phone, then the device will report its local IP address automatically.
3.2.2 Restore factory setting via IVR
Use a mobile phone to call a SIM card number of the device, the device will answer and play a voice prompt of ‘dial the extension number’. Press *166*000000# on the mobile
phone, then you will hear ‘setting succeeds’, then the factory setting of the gateway will be restored.
3.2.3 Restore default IP and password
Press RST button for about 3 seconds, then the device will be rebooted and the IP address, username and password will be restored to factory default.
3.2.4 Restore factory setting
Press RST button for about 7 seconds, then gateway will be rebooted and restored to factory setting.
3.3 Local Maintenance through Console Port To ensure easy maintenance, the device provides a standard RS232 console port, which has a Baud rate of 115200bps. Users can log in the device to carry out maintenance-related configurations through the console port. Ø Example: Log in device via Console Port Step 1: Prepare a serial cable as follows (standard RS232, 115200bps);
Step 2: Connect the F port of the serial cable with the COM port of PC. If the PC does not have a COM port, please use a USB-to-COM converting tool to connect the serial cable with the PC. Step 3: Connect the M port of the serial cable with the console port of the device.
Step 4: Conduct configurations on login software.
Herein we take the PuTTY software as an example. Detailed configurations are as follows:
After finishing the above configurations, click the Open button to enter the maintenance interface of the console port. The username and password are the same with those of the web interface of device.
Commands for configuring the IP address of the device :
(In the following example, IP address of device needs to be configured as 172.30.66.100, and netmask is 255.255.0.0) > enable enable# configure config# interface ethernet config-if-br-lan# ip address 172.30.66.100 255.255.0.0 config-if-br-lan# exit config# ip default-gateway 172.30.0.1
Commands for inquiring the IP address of the device
> enable
enable#show interface
4 WEB Interface Configuration
UC2000-VF/G serials gateway has the same web interface. This chapter describes web configuration of UC2000-VE. The UC2000-VE contains an embedded web server to set parameters by using the HTTPS/HTTP protocol. We are strongly recommended to access device with Google Chrome or Firefox Browser. The configuration introduction also suitable for following models:
UC2000-VE-4G UC2000-VE-8G UC2000-VF-16G UC2000-VF-8G UC2000-VF-32G UC2000-VE-8C (8 Channels CDMA Gateway) UC2000-VE-4C (4 Channels CDMA Gateway) UC2000-VF-16C (16 Channels CDMA Gateway) UC2000-VF-32C (32 Channels CDMA Gateway) UC2000-VE-8W (8 Channels WCDMA Gateway) UC2000-VF-16W(16 Channels WCDMA Gateway) UC2000-VF-32W(32 Channels WCDMA Gateway) UC2000-VE-4T UC2000-VE-8T UC2000-VF-16T UC2000-VF-8T
4.1 Access UC2000 unit
Enter IP address of UC2000 in IE/Google Chrome. The default IP of LAN port is 192.168.11.1. and the GUI shows as below: Figure 4-1-1 WEB log interface
Enter username and password and then click “Login” in configuration interface. The default username and password are “admin/admin”. It is strongly recommended, change the default password to a new password for system security.
4.2 Parameters Configuration
UC2000 WEB configuration interface consists of the navigation tree and the detail configuration interfaces.
Figure 4-2-1 WEB introduce
Go through navigation tree, user can check, view modify, and set the device configuration on the right of configuration interface.
4.3 System Information
System information interface shows the basic information of status information, Mobile information, and SIP information. 4.3.1 System information
Figure 4-3-1 system Information
Table 4.3-1 System Information
Parameters Description MAC Address Displays the current MAC of the gateway, for example: 00-1F-D6-1B-3D-02 Network Mode UC2000-VE works as bridge mode by default Network Current IP address and subnet mask of gateway DNS Server Displays DNS server IP address in the same network with the gateway
Device ID A unique device ID which assigned in factory. This device ID to be used as register ID with Ultiroam SIM cloud.
Server Register status Its indicates communicate status with SIM Cloud server, there are two type of status: Registered Not Registered
Need Authentication
License It indicates device’s license status. Contact with support when it displays as Invalid System Up Time Shows the time period of the device running. For example:1h: 20m, 24s
Traffic Statistics Calculates the net flow, including the total bytes of message received and sent.
Version info shows the current firmware version l Device Model: Model name of the device l Package version: 02231301 2018-07-10 17:04:31 official
is the version number l Software version: 02231301 2018-07-10 17:04:31 official, 02231301 is the version number l Web version: the version number of web system. The web version must match with software l User board 0 Version: the firmware version of user board slot 0
l User board License ID: Contact with support when it displays as Invalid l Hardware version/DSP version/ SIM box version
4.3.2 Mobile Information
Figure 4.3-2 Mobile Information
Table 4.3-2 Mobile Information
Parameters Description Port Number of GSM/CDMA ports.
Type Indicates the current type of module. Such as GSM, CDMA, WCDMA, LTE
Network Mode Indicates the current type of network. Such as GSM, CDMA, WCDMA, LTE
IMSI International Mobile Subscriber Identity, it is the uniquely identifies of SIM card IMEI Module series NO
Status Indicates the connection status of current GSM/CDMA /WCDMA/LTE module
Credits It showing available total call credit or time of SIM card while Call Limit is enabled. Operator Displays the network carrier of current SIM card.
Signal Displays the signal strength of in each channel of GSM/CDMA /WCDMA/LTE
ASR
Answer Seizure Ratio is a measure of network quality. It’s calculated by taking the number of successfully answered calls and dividing by the total number of calls attempted. Since busy signals and other rejections by the called number count as call failures, the ASR value can vary depending on user behavior.
ACD The Average Call Duration (ACD) is calculated by taking the sum of billable seconds (bill sec) of answered calls and dividing it by the number of these answered calls.
PDD Post Dial Delay (PDD) is experienced by the originating customer as the time from the sending of the final dialed digit to the point at which they hear ring tone or other in-band information. Where the originating network is required to play an announcement before completing the call then this definition of PDD excludes the duration of such announcements.
Call Status Show the Status of port, include idle, active, alert and processing Idle means there is no call on this channel Processing means call is connecting Alerting means destination is ringing Active means the call is connected
Ringing means the gateway is answering incoming call from mobile
Calling Waiting means the gateway is receiving another call during conversation and implement call waiting service Call Hold means the call is hold by extension of IPPBX/SIP Server
4.3.3 SIP Information
Figure 4-3-3 SIP Information
Displays registration status information with Softswitch platform or SIP Server
Table 4-3-3 SIP information
Parameters Description Port The number of SIP channels
SIP User ID SIP registration account which are provided by the Softswitch and SIP server
Register Status Shows the registration status of VoIP channel, including registered and unregistered. Port Group The number of SIP channels Port List The ports of the port groups contain
4.4 Statistics
4.4.1 TCP/UDP
Figure 4-4-1 TCP/UDP Statistics
4.4.2 RTP
Figure 4-4-2 RTP
Table 4-4-1 Description of RTP Statistics
Parameters Description Port The port of RTP statistics Payload Type The voice code of this channel, Include G.723.1/PCMA/PCMU/ G.729AB Packet Period Time of packaging Local Port Local port of transmitting RTP packages Peer IP End of equipment IP address Peer Port Peer port of receiving RTP packages Send Packet Total of sending RTP packages Recv Packet Total of receiving RTP packages Loss Packet Total of losing RTP packages Jitter Length of delay jitter Duration Time(s) Both ends of the call time
4.4.3 SIP Call History
SIP Call History
SIP Call History
Parameters Description Port The port of Call statistics Incoming Received The amount of received incoming calls which coming from IP side Incoming connected
The amount of incoming calls which have connected Incoming Answered The amount of incoming calls which answered by modular Incoming Failed The amount of incoming calls which failed Outgoing Attempted
The amount of outgoing calls which attempted to IP side Outgoing Connected
The amount of outgoing calls which have connected Outgoing Answered The amount of outgoing calls which answered by IP side Outgoing Failed The amount of outgoing calls which failed
4.4.4 IP to GSM Call History
IP to GSM Call History
IP to GSM Call History
Parameters Description Port Device GSM/CMD/WCDMA/LTE port
Call Statistics the number of calls in this port Duration Statistics call total time Answered Statistics response times Call Failed Caused by SIP Statistics cause of call failure from SIP, include: canceled/ timeout/ not allowed/ Negotiation failed Call Failed Caused by GSM Statistics cause of call failure from GSM, include: Busy/ no answer/ no dial tone/ no carrier
4.4.5 CDR Report
Figure 4-4-5 CDR Report
It is support 10000 CDRs on gateway. The CDRs will lost after reboot while save CDR set to No.
Parameters Description Port GSM/CDMA /WCDMA/LTE port number Start Date/Answer Date start and end time of calls Call Direction IP to GSM:
outbound calls from softswitch/IPPBX to mobile network GSM to IP: incoming calls from mobile network to IPPBX/ Softswitch Source Calling number Source IP Calling ip address
Destination Called number Hang Side Who hang up the call, calling, called or gateway Reason The reason of the call hang up Duration(s) Call duration of the call Codec The voice code of this call, Include G.723.1/PCMA/PCMU/ G.729AB RTP send/recv/loss rate RTP Statistics of the call Jitter(s) Voice jitter BCCH Which bcch the call using, first you need enable check bcch
4.4.6 Lock BCCH History
Figure 4-4-6 Auto Lock BCCH History
It is record history of BCCH to help analysis SIM card register status.
4.4.7 Current call status On the Statistics à Current Call Status interface, status and detail of the current call are shown.
4.4.8 GSM Event
GSM event page will record all the logs of GSM modules such as IMEI change, replace new SIM card to specific port etc.
4.5 Network Configuration
4.5.1 Local Network
Figure 4-5-1 Local Network
Table 4-5-1 Local network
Parameters Description Obtain IP Address Automatically
Enable the device obtain IP Address automatically or not. Temporary IP address and Subnet Mask
When device can’t get the ip automatic, it will use the temporary IP
Use the Following IP Address Configure the "IP Address", "Subnet Mask" and "Default Gateway" by manual, default this is enable, and default ip is 192.168.11.1
PPPoE Need ISP offer the account and password, Use this mode when there is not router in the local network MTU Message transmit unit, default is 1400 Obtain DNS Server Address Automatically
When enable the WAN port option of "Obtain DNS Server Address Automatically”, which will be enabled subsequently. Use the Following DNS Server Addresses
Fill in the IP address of "Primary DNS Server" and "Secondary DNS Server"
4.5.2 ARP
The ARP function mainly used to query and add the map of IP and MAC. There are static or dynamic ARP entries.
Like other routers, the gateway can automatically find the network device on the same segment. But sometimes you don't want to use this automatic mapping; you'd rather have fixed (static) associations between an IP address and a MAC address. Gateway provides you the ability to add static ARP entries to:
l Protect your network against ARP spoofing
l Prevent network confusion as a result of misconfigured network device
Figure 4-5-3 Add ARP
Click Search All to check ARP buffer.
4.5.3 VPN Parameter
Parameters Description Server VPN Server IP or domain name (support PPTP only) Account VPN account which provides by server or VPN provider Password Password of VPN which provide by server or VPN provider Domain Follow VPN setting, can be null
Use MPPE Encryption parameter, support 40/128 bit, must be match with VPN server
Check VPN connecting status on system information
4.6 Security Center
4.6.1 Access Rules On the Access Rules interface, click Add, and you can set rules to accept or reject the calls from a specific port, the login of other people via Web or Telnet, or PIN packages.
TCP: accept or reject the login of other people via Web or Telnet; UDP: accept or reject the calls from a specific port;
ICMP: accept or reject PIN packages.
All: accept or reject all the above mentioned items.
4.7 Mobile Configuration
4.7.1 Basic Configuration
Basic Configuration
SIM Mode
Ultiroam gateway support two types of SIM card installation, which is local and remote SIM management.
Item Description Local To use local SIM card which install on gateway, this way is most common used by many of users SIM Box SIM Box is a small box which use for SIM card storage. It ideal for users who want replace SIM card frequently SIM Bank SIM Bank is use for SIM card storage and remote SIM management together with Ultiroam SIM Cloud
Introduction to API
The API protocol is used for external applications (for instance: SMS Server) to control the sending and receiving of SMS/USSD on the gateway. Old version:
To enable the API function of the GSM gateway, the IP address, port, user ID and password of SMS Sever must be correctly configured, and the TCP Intercept function of the SMS Server must be enabled. Once the connection between the gateway and TCP is established, the gateway will send user ID and password to the SMS Server, and then the SMS Server will send back a message which indicates successful authentication.
The API Server Address, API Server Port, User ID and API User Password on the above interface of Gateway must be the same with the IP Address, Port, Auth ID and Password on the setting interface of SMS Server. New version:
The API is based on HTTP and JSON. So please check how to send HTTP request and how to encode/decode JSON data before you write an application with this API. please contact support for the document. Smpp:
SMPP function support from client after binding to send text messages, length and mass text messages, and support to receive text messages, receive SMS receipt, send the query results, and other functions at present average. Configured SMPP listener port and the user password, and then restart the equipment. GSM gateway is smpp server, it can connect with smpp client. Introduction to GSM Audio Coding
There are eight formats for GSM Audio Coding, including Auto, FR, HR, EFR, AMR_FR, AMR_HR, FR and EFR, EFR and FR. Auto: it means GSM Audio Coding is automatic.
FR (Full Rate): the first digital speech coding speech standard used in the GSM digital mobile phone system. The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample (often padded out to 33 bytes/20 ms or 13.2 kbit/s).
HR (Half Rate): the bit rate of the codec is 6.5 kbit/s. It requires half the bandwidth of the Full Rate codec and network capacity for voice traffic is doubled, at the expense of audio quality. It is recommended to use this codec when the battery is low as it may consume up to 30% less energy.
EFR (Enhanced Full Rate): is a speech coding standard that was developed in order to improve the quite poor quality of Full Rate codec. Working at 12.2 kbit/s, the EFR provides good quality in any noise conditions. The EFR is compatible with the
highest AMR mode (both are ACELP). Although the EFR helps to improve call quality, this codec has higher computational complexity, which in a mobile device can potentially result in an increase in energy consumption as high as 5% compared to 'old' FR codec.
AMR (Adaptive Multi-Rate): is an audio compression format optimized for speech coding. AMR speech codec consists of a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality speech starting at 7.4 kbit/s.
There are two modes for the AMR codec in the device:
AMR_FR: the AMR codec in a full rate channel (FR)
AMR_HR: the AMR codec in a half rate channel (HR).
FR and EFR: GSM Audio Coding supports both FR and EFR, but FR is prior to EFR. EFR and FR: GSM Audio Coding supports both EFR and FR, but EFR is prior to FR. Example:
Configuration between SMS box and gateway Configure API parameters on gateway
The IP server which installed SMS box software is 172.16.221.221, pre-set Port 12000, User ID aabbcc and password abc123 as example.
Configure SMS box
Then click OK and start service, the gateway IP will be presented in device list of SMS box
4.7.2 Mobile Configuration
Description of Mobile Configuration
Parameter Description CLIR Calling Line Identification Restriction: If the CLIR function is enabled, the phone number of the caller will not be displayed on the called phone, this needs support by the operator, if operator not support, device also can’t do it. Detect Reverse Polarity If the function is enabled, the caller will learn whether the called person has got through the phone. Internet Access Allow the sim cards access internet or not.
For example, if you want enable auto Internet access, please enable this. Tx Gain Gain of voice sent
Rx Gain Gain of voice received APN/APN name/APN PSW APN refers to a network access technology, which is a parameter that must be configured when accessing the Internet via mobile phone. Band Type Choose from GSM850, GSM900, GSM1800, GSM1900, WCDMA800, WCDMA 850, WCDMA900, WCDMA1900, and WCDMA2100 Network Mode Select 2G ,3G or 4G SMSC Short Message Service Center Reset Click Reset, and the corresponding module will be reset. Block/Open Click Block or Unblock, the corresponding module will turn to the opposite status. Power On/Off Click On or Off, the power of the corresponding module will turn to the opposite status.
4.7.3 Phone Number Config
On the Phone Number Config interface, you can write a phone number into a specific memory card and SIM Card, and thus the phone number can be called in case that this SIM card has been pulled out and inserted into another port.
Select Yes on the right of ‘Write Phone Number to SIM Card’, enter a phone number and click Submit.
4.7.4 PIN Management PIN code is a combination of numbers used as an additional password to access the SIM card of the selected port. On the following interface, you can set a PIN code for the SIM card of the selected port.
PIN Management
Description of PIN Management
Parameters Description PIN Personal identification number of SIM card. In the status of SIM card locked, PIN can be modified to prevent SIM card from being stolen. Select Port Selects the GSM/CDMA channel number
4.7.5
IMEI
IMEI Modify: to change the IMEI code for specific port/ports
IMEI Auto Set: set some rules to change the IMEI code with some predefinited conditions
4.7.6 Operator
Click Search button while there is SIM card in that port, after a while, you will see Operator codes list under Operator List drop box. And then select correct operator code which match with the SIM card insert in the gateway. Finally, save the setting and reboot the device to make SIM card re-register again.
4.7.7 Operator Configuration
Operator configuration aim to set operator code for batch of SIMs. Inserted SIM cards will match with IMSI prefix and register SIM card to the code as per setting. 4.7.8
BCCH
BCCH (Broadcast Control Channel): BCCH is a logical broadcast channel used by the base station in a GSM/WCDMA network to send information about the identity of the network. The information is used by a mobile station to get access to the network. Information includes the Mobile Network Code (MNC), the Location Area Code (LAC) and a list of frequencies used by the neighboring cells. Configuration Procedures for BCCH:
Step 1. In the navigation tree on the left, click Mobile Configuration à BCCH.
Step 2. Drag the scroll bar on the bottom of the interface, and you will see buttons.
Click the button of a specific port, and you will see the following interface
Step 3. Click the drag-down box on the right of BCCH Mode, and select a mode.
Default: All frequencies will be automatically matched with the gateway. Fixed: You are required to set three fixed frequencies, and the frequencies will be matched with the gateway permanently.
Random: you are required to set some conditions, including minimum signal strength, the period for automatic frequency switch, and whether to switch frequency during calling.
Advanced: you are required to set some conditions, including minimum signal strength, minimum answer-seizure ratio(ASR), number of calls and number of failed calls.
Note: When the actual number of failed calls reaches the set number, frequencies will be switched or when the actual answer-seizure ratio is less than the minimum answer-seizure ratio, frequencies will be switched.
Note: The BCCH Whitelist only works at random mode and advanced mode.
Only GSM module support Fixed/Random/Advance mode, other modules don’t support. 4.7.9 Call Forwarding Calls can be forwarded unconditionally or under certain conditions.
Call forwarding is the same as mobile phone which to activate/deactivate supplementary service of SIM card. For more details of these services, please contact to local providers.
Parameter Explanation Call Unconditional Calls will be forwarded unconditionally Call Forwarding No Reply If there is no reply from the called number, calls will be forwarded. Call Forwarding Busy If the called number is busy, calls will be forwarded. Call Forward on Not Reachable If the called number is not reachable (for example, the called phone is power off), calls will be forwarded. Cancel All Calls will not be forwarded. Call Number The number where calls will be forwarded.
4.7.10 Call Waiting
On the Mobile Configuration àCall Waiting interface, the call waiting function can be disabled or enabled.
Call waiting is the same as mobile phone which to activate/deactivate supplementary service of SIM card. For more details of these services, please contact to local providers.
Notes: Call waiting only takes effective while “Do Not Answer GSM Incoming Call for Hotline” set to Yes.
Call Configuration -> Service Parameter
4.7.11 Cloud Server Users need to configure the cloud server when the gateway works with SIM Bank or centralized management is required for the gateway.
Item Description Primary Server The domain name of IP address of the primary Cloud
Domain server Primary Server Port The port of the primary Cloud server Secondary Server Domain The domain name of IP address of the secondary Cloud server. It can be null. Secondary Server Port The port of the secondary Cloud server. It can be null. Domain Name The name of the sub-domain used by the gateway under the Could server. Password The password of the sub-domain used by the gateway under the Could server. Local Port The port of the gateway connected to the Cloud server. SIM Transport Type The transmission type of phone numbers of the SIM card. Port State Control By The port state is controlled by cloud or the gateway. Anti Call Scanning This function must be enabled when the whitelist/blacklist function of the SIM card is enabled.
How to register gateway to SIM Cloud?
Example: add gateway on domain support.cloud.com
Device S/N is the device ID on gateway, find it on the page system information, as below:
4.7.12
MBN Config
This tool used to help update LTE module MBN file.
4.8 SMS and USSD
4.8.1 SMS Send Overview On the SMS Send Overview interface, you can see the number of SMS messages that have been sent via the ports of the gateway, as well as the daily limit and monthly limit of SMS messages that can be sent through the ports of the gateway.
4.8.2 SMS Send Limit Settings
On the SMS Limit Settings interface, click Add, and you can see the following interface.
4.8.3 SMS Routing You can set sms routing if you use gateway as sms terminal.
4.8.4 SMSC Switch Setting Every operator can set 8 SMSC, it will switchover by the number you set successful or fail. For example: Successful send set 5, fail send set 1,
When the port send 5 sms successful, it will switch to next SMSC. When the port send 1 sms fail, it will switch to next SMSC.
4.8.5 Send SMS
The GSM gateway can be used to send messages and receive massages.
Parameter Explanation Port The port through which SMS messages are sent To The number(s) where the SMS message will be sent. UCS2 UCS2: Support English and Chinese
GSM 7bit: Support English only Message The content of the message
SMS send report
4.8.6 SMS Outbox On the SMS Outbox interface, you can see the detailed information of each SMS message that has been sent, and can export the messages.
4.8.7 SMS Inbox
On the SMS Inbox interface, you can see the detailed information of each SMS message that has been received, and can export the messages.
4.8.8 USSD USSD (Unstructured Supplementary Service Data): is a service which is provided by a telecom operator and allows GSM/WCDMA/LTE mobile phones to interact with the telecom operator's computers. USSD messages travel over GSM/WCDMA/LTE signaling channels and are used to query information and trigger services. Unlike similar services (SMS and MMS), which are stored and forwarded, USSD is session-based. It establishes a real-time session between mobile phones and telecom operators’ computers or other devices.
4.8.9 Email
Destination Email Address 1/2/3: Enter the e-mail address to receive the SMS content.
Title: Configure the title of the e-mail, which will be used as the e-mail title when send the SMS to destination mail. Check Email Every: how long time to check the mailbox.
Subject: Mail subject, gateway will check mailbox subject, if same, it will forward to sms.
Email Account: Configure one e-mail address, which will be used for sending the SMS to destination e-mail. Outgoing: Configure the SMTP server domain here, different e-mail address server have different server addresses, please confirm this with your e-mail provider or search from Internet. configure the SMTP port, usually 25, please also confirm this with your e-mail address provider. Incoming: set the outgoing protocol, server, and port, different e-mail address server have different server addresses, please confirm this with your e-mail provider or search from Internet. TLS Enable: Enable the TLS or not. If your e-mail address server requires TLS, please enable it.
How to set Email to SMS
Description
GSM gateway can check the email inbox on time, when have unread email at list and size less 300 chars, will try to read it. When email use protocol IMAP, if email read successful, will set the email status to read. If read email failed, will try to read again (MAX 3 times). If failed final, the email status will be set to read also. When email use protocol POP3, if email read successful, will delete it. If email read failed, keep the email status, because the UC2000 will check again at next check time. After read the email, if the subject matched, will extract the context from key words: “To:”, “Encoding:”, “Message:” as SMS receive number, SMS encoding, SMS context. If the GSM gateway have not available channel at that time, it will keep the SMS in queue and waiting till have available one. The queue max has 5120 SMS. If the queue full, the UC2000 will stop to check the email. How does SMS to email works
1) Send Email
Email Format: Plain text Email subject: Test SMS Email contents:
To:1008611
Encoding:7Bit
Message: Hello this is test SMS from phoebe
2) mail server forward email to support@ultiroam.com 3) GSM gateway check the inbox of support@ultiroam.com, find the email subject with ‘Test SMS’ 4) GSM gateway send SMS to mobile 1008611
Notice: Don’t set signature at the end of email and make sure the received email is plain text format.
How to configure Email to SMS in GSM gateway
1) Open page SMS and USSD>>>>>>>Email. Email to SMS support both POP3 and IMAP protocol.
The “Server” means your email services server info, you can get it from your email provider. The “TLS Enable” means use Encrypt or not.
If use TLS, IMAP default server port is 993, POP3 default server port is 995.
If not use TLS, IMAP default server port is 25, POP3 default server port is 110.
The “Check Email Every” means how long the UC2000 will check the email inbox, the set range is 1-60. The “Subject” means when the UC2000 match the email subject, will use that email to SMS. Add the Email address info at UC2000 side.
2) Email must use fix format: Subject: this subject MUST be the same as email subject. Example, when you send email with subject “Test SMS”, the Subject s field in GSM gateway must be “Test SMS” also. Email contents usually include 3 parts:
The “To” means destination number you want send to, this option is obligatory. The format is: To:xxxxxxxxxxxx
The “Encoding” means which format of SMS used, the format include 7Bit and UCS2, UCS2 is default. This option is Optional. The format is: Encoding:7Bit
The “Message” means which content you want send out, this option is obligatory. The content length max 300 chars. The format is: Message: ……………………………….
Received email should be in the inbox of support@ultiroam.com .
Note:
1) Character set. The UC2000 support character set ASCII and UTF-8 only. 2) Encoding. The email encoding support 8Bit, Base64 and Quoted-Printable only. If the email senders use other encoding, like 7Bit, it will not support. 3) Email size. The email size can’t more than 300 chars, if more than it, the UC2000 will not try to read it.
How to set SMS to Email
The UC2000 series gateway support to send the SMS received on the gateway to user’s mail box. Login device’s web, go to SMS and USSD-->Email page, enable SMS to Email function, and configure the other parameters needed.
4.9 Call Configuration
4.9.1 SIP Configuration This section describes how to configure SIP server and SIP parameters.
Configure SIP server and Outbound Proxy server
SIP Server Address and Port
Used for configure SIP server address and port, the address can be IP Address, also can be a domain name which can be resolved by DNS server
Check NET Status
Default is No. if it set to Yes, the gateway will send SIP OPTION periodic to check health status between gateway and SIP server.
Outbound Proxy
Outbound proxy, it mainly used in firewall / NAT environment. That make the signaling and media streams able to penetrate the firewall.
Local SIP Port Configuration
In order to work different application scenarios, the gateway provides flexible configuration with local SIP port.
Random
The gateway will generate SIP port after each reboot by random. It is commonly used while 5060 is blocked or conflict with other devices.
Use the same SIP port
It is mostly used to SIP trunk interworking with SIP server so that the gateway able to deal with high performance concurrent calls. Use the same local SIP port and SIP User ID
Use the separate SIP port
Each channel has separate SIP port so that they could be handle SIP call separately. After Use Same Local SIP Port set to No
The Local SIP port will be changed on Port Parameter page.
Auto set SIP Account and Router
The gateway will generate sip account and router auto. After set, It must restart the device to take effect.
Auto SIP Account: Prefix+Port/Iccid/IMSI
Auto IP->TEL Route: Source/Destination + Iccid/IMSI/Number
Register Interval and DNS SRV Register Interval
This field specifies the value that the gateway will send in the Expires header of the REGISTER message. Its value ranges from 1-3600s. But in fact, the gateway will get 200OK response from SIP server after REGISTER request, and an Expires header will be included in 200 OK message body. This value in the 200OK determines the time, in seconds, after which the registration expires. The gateway will refresh the registration Timer Register Delta seconds before the end of this interval.
DNS query type
The DNS query type defines the type of information that will be requested from DNS server
DNS refresh interval
The interval of DNS refresh, Ranges from 0 to 60000 mins, 0 means disable default value is disable.
Configuring SIP Timers T1
This field specifies the lowest value, in milliseconds, of the retransmission timer for SIP messages. Default specifies 500.
T2
This field specifies retransmission timer for T1 timeout of SIP message, in milliseconds. Default specifies 4000.
T4
This field specifies retransmission timer for T2 timeout of SIP message, in milliseconds. Default specifies 5000.
TMAX
This field specifies maximum timeout value for SIP message. The SIP message will be dropped after TMAX. Default value is 32000
Keepalive Interval
The gateway can monitor the status of SIP server by sending periodic SIP OPTION messages. This field specifies transmission timer of OPTION message. Its range from 10-3600s.
Keepalive SIP ID
This field specifies SIP ID of OPTION. The format would be <xxx@host.com >, example: OPTIONS sip:heartbeat@172.16.0.8:2080 SIP/2.0
Via: SIP/2.0/UDP 172.16.222.22;branch=z9hG4bK45c4f8d2026d9eed8a0adcd533161efd; From: <sip:heartbeat@172.16.222.22:2080>;tag=6d48f0a169d33fe7b032c0fd895084fd To: <sip:heartbeat@172.16.0.8:2080> Call-ID: 8874a4e49f11af243c6b717c05a16e35@172.16.222.22 CSeq: 1804289386 OPTIONS Contact: <sip:31@172.16.222.22> Max-Forwards: 70
Accept: application/sdp Content-Length: 0
Keepalive Retry Count
How many counts will retry if no response. Its value ranges from 1-10 times.
Configuring Caller ID and 183 Mode From Mode when Caller ID Is Available
Used to configure "From" Mode when Caller ID Is Available when call from GSM to VoIP Tel/User: From: Caller ID <sip:3001@host.com>;tag=51088abb User/User: From: 3001 <sip:3001@host.com>;tag=51088abb Tel/Tel: From: Caller ID <sip: Caller ID@host.com>;tag=51088abb User/Tel: From: 3001 <sip: Caller ID @host.com>;tag=51088abb
From Mode when Caller ID Is Unavailable
Used to configure "From" Mode when Caller ID Is Unavailable Anonymous : From: <sip: Anonymous @host.com>;tag=51088abb Username : From: <sip: Username @host.com>;tag=51088abb
Answer Mode
Answered: Gateway will send SIP message "200 OK" to SIP Server after GSM/CDMA users answered the call.
Alerted: Gateway will send SIP message ‘200 OK’ to SIP Server immediately after 183 Ringing. In this situation, the called party possibly still in ringing status.
Delay Answer (range:0 - 10s)
Outgoing call from ip to gsm,when gsm side answered,device will delay the time response 200 OK to ip side.
183 Mode
Immediately: Gateway will send "183 RING" immediately to SIP Server while it receives “INVITE". In this situation, the called party possibly still not in ringing status. Alerted: Gateway will send "183 RING" after received exact ringing signal from GSM/CDMA network. In this situation, the called party is definite in ringing status.
Called Number Parse
Where get the called number, from Request-Line or To header.
Caller Number Source
Where get the called number, from User name or Display name.
Request Line
The request uri forced to use Remote Contact during the session. normal calling doesn’t need set this.
Session Timer SIP Session Timers which is an extension of SIP RFC 4028 that allows a periodic refreshing of a SIP session using the RE-INVITE/UPDATE message. The refreshing allows both the user agent and proxy to determine if the SIP session is still active. The SIP Session Timer is a keep alive mechanism for SIP sessions that allow User Agents (UA) or proxies to determine the status of a session and to release it if it is not active, even if a BYE has not been received. Session timer Interval
The initial INVITE request establishes the duration of the session and may include a Session-Expires header and a Min-SE header. These headers indicate the session timer value required by the user agent (UAC). A receiving user agent server (UAS) or proxy can lower the session timer value, but not lower than the value of the Min-SE header. If the session timer duration is lower than the configured minimum, the proxy or UAS can also send out a 422 response message. If the UAS or proxy finds that the session timer value is acceptable, it copies the Session-Expires header into the 2xx class response.
A UAS or proxy can insert a Session-Expires header in the INVITE if the UAC did not include one. Thus, a UAC can receive a Session-Expires header in a response even if none was present in the request. Its value ranges from 90-60000s.
Session Timer Refresher
It specifies refresher which including in SIP message body, user agent client (UAC) or user agent server (UAS). UPDATE sips:bob@192.0.2.4 SIP/2.0
Via: SIP/2.0 pc33.atlanta.example.com;branch=z9hG4bKnashds12 Route: sips:p1.atlanta.example.com;lr Supported: timer
Session-Expires: 4000;refresher=uac Max-Forwards: 70 To: Bob <sips:bob@biloxi.example.com>;tag=9as888nd
From: Alice <sips:alice@atlanta.example.com>;tag=1928301774 Call-ID: a84b4c76e66710 CSeq: 314162 UPDATE
Contact: <sips:alice@pc33.atlanta.example.com>
Configuring GSM-SIP Mapping Code
This part specifies response codes between GSM cause reason and SIP response code.
SIP Response
404 Not Found 408 Request Timeout 403 Forbidden 486 Busy Here 480 Temporarily unavailable Resource unavailable 503 Service Unavailable
Response Code switch
This part specifies response codes of SIP between gateway and SIP server. Refer to table SIP Response, the SIP server possibly needs some specific SIP response from the gateway. Example, SIP server needs SIP response 180 Ringing instead of 183 Ringing, the configuration should be as below:
Custom Extensions Header Customer Extensions Header
Send the IMEI/IMSI/Portno info to other in SIP Header.
SIP/RTP Encryption
When you use VOS as sip server, and you want SIP and RTP encryption, please enable this option. 4.9.2 SIP Trunk Configuration
Table 4-11-1 Description of IP Trunk
Parameters Description
SIP Trunk Add remote IP of Softswitch, SIP server which will send call traffics to gateway.
Index It uniquely identifies a trunk. Its value is assigned globally, ranging from 0 to 31.
Description It describes the trunk for the ease of identification. Its value is character string
IP It is an interworking parameter between the remote Softswitch and the SIP server. It specifies the IP address of the peer equipment.
Port It is an interworking parameter between the remote Softswitch and the SIP server. It specifies the SIP port number of the peer equipment
Keep alive Send OPTION to Softswitch/IPPBX to detect health status
Example
To add a remote IP of Softswitch, SIP trunk index is 31, SIP port number “5060”
4.9.3 SIP Trunk Group
Figure 4-11-3 IP Trunk Group
Table 4-11-2 Description of IP Trunk Group
Parameters Description
IP Trunk Group This configuration is optional, and is used to add the IP that have the same attributes to an IP group. The IP group will be referenced by IP->Tel routing and number manipulation.
Index It uniquely identifies a route. Its value is assigned globally, ranging from 0 to 31.
Description It describes the route for the ease of identification. Its value is character string IP It specifies the IP will add to IP group
Example
To add an IP group, set IP “10, 14, 17” to IP group 18
Figure 4-11-4 IP Trunk group modify
4.9.4 Port Configuration
Table 4-12-3 Description of Port Configuration
Parameters Description Port Configuration Used to configure ports’ gain, Auto-Dial, etc. ALL ports register used same user ID The default is no. If set to "yes", all the ports will use the same user ID to register to SIP server
SIP User ID It is the account used for registration which provide by SIP server, equipment port's unique identifier
Authenticate ID
The Authentication ID is used for authentication purposes. The SIP user ID is usually the phone number you received from the service provider. Often, the Authentication ID is the same as the user ID
Authenticate Password
Password of SIP User ID which provide by SIP server Local SIP Port The channel sip port Register to Register with which sip server Tx Gain Tx Gain value of chipset. Adjusting it will affect volume on GSM side. Rx Gain Rx Gain value of chipset. Adjusting it will affect volume on IP side.
To VoIP Hotline
When mobile / fixed line users make call to this port, gateway will auto forward to dedicate number. The hotline could be DID / Ring Group / Extension of SIP server / IP-PBX.
- Note: Please configure Tel->IP Operation if you need this function.
To PSTN Hotline
When VoIP users make calls to this port, gateway will auto forward to dedicate number. The Hotline number could be mobile / fixed line number. Leave it blank if you don’t need this function.
- Note: Please configure IP->Tel Operation if you need this function.
Auto-Dial Delay Time The auto-dial delay time of hotline, the range is 0-10 seconds
4.9.5 Port Group Configuration
Select ports for defined port group.
4.9.6
Digitmap
Digit Map Syntax:
1. Supported objects
Digit: A digit from "0" to "9".
Timer: The symbol "T" matching a timer expiry.
DTMF: A digit, a timer, or one of the symbols "A", "B", "C", "D", "#", or "*".
2. Range []
One or more DTMF symbols enclosed between square brackets ("[" and "]"), but only one can be selected. 3. Range ()
One or more expressions enclosed between round brackets ("(" and ")"), but only one can be selected. 4. Separator
|: Separated expressions or DTMF symbols.
5. Subrange
-: Two digits separated by hyphen ("-") which matches any digit between and including the two. The subrange construct can only be used inside a range construct, i.e., between "[" and "]". 6. Wildcard
x: matches any digit ("0" to "9").
7. Modifiers
.: Match 0 or more times.
8. Modifiers
+: Match 1 or more times.
9. Modifiers
?: Match 0 or 1 times.
Example:
Assume we have the following digit maps:
1. xxxxxxx | x11
and a current dial string of "41". Given the input "1" the current dial string becomes "411". We have a partial match with "xxxxxxx", but a complete match with "x11", and hence we send "411" to the Call Agent.
2. [2-8] xxxxxx | 13xxxxxxxxx
Means that first is "2","3","4","5","6","7" or "8", followed by 6 digits; or first is 13, followed by 9 digits.
3. (13 | 15 | 18)xxxxxxxxx
Means that first is "13","15" or "18", followed by 8 digits.
4. [1-357-9]xx
Means that first is "1","2","3" or "5" or "7","8","9", followed by 2 digits.
4.9.7 IP->Tel Routing
Add a new outgoing route rule, click Add button
Click to set caller and called prefix
Source: indicates call from which SIP server or SIP trunk Destination: indicates call to which port or port group Call Restriction: allow or forbid to call out
Source Prefix: to match with prefix of caller number Destination Prefix: to match with prefix of called number Prefix to add: to add a prefix in front of called number
Digits to be deleted: indicates how many digits to be deleted for called number Number of digits reserved: to definite the number of length of called number Examples:
Caller number 201 dial any number which will route to port 0.
Remove prefix 991 of called number.
Remove prefix 88 and then add 0 in front of called number
4.9.8
Tel->IP Routing
Add a new incoming route rule, click Add button
Click to set caller and called prefix
Source: indicates call from which SIP server or SIP trunk Destination: indicates call to which port or port group Call Restriction: allow or forbid to call in Source Prefix: to match with prefix of caller number Destination Prefix: to match with prefix of called number Prefix to add: to add a prefix in front of called number Digits to be deleted: indicates how many digits to be deleted for called number Number of digits reserved: to definite the number of length of called number 4.9.9 Gsm Calling Config Modify the caller number of the incoming call.
4.9.10 Service parameter
To configure dialing mode parameters Do Not Answer GSM Incoming Call for Hotline
When the gateway get incoming call from mobile network, the modular will answer the call then start to DTMF or route to destination hotline number. While this option enabled, the modular won’t answer the call but routing to destination hotline number till it getting answer.
Notes: Refer to Port Parameter page for Hotline configuration.
Enable GSM Incoming Configuration
Means when call from mobile side, you can dial the feature codes (Chapter 3 Basic Operation) to configure IP address and so on
Answer Delay
In most instances, Most of CDMA operators don't offer answer signal. The gateway doesn’t response SIP 200 OK to SIP server in case of missing answer signal from CDMA network. Answer delay is to fix this issue and generate SIP 200 OK to SIP server after answer delay timeout. Default value is 5 seconds. Moreover, it is available for CDMA gateway only.
Ringback Tone
Default device forward the Ringback to IP side from GSM operator. But sometimes GSM operator Ringback not clear or other issue,client want device or softswitch play ringback,you can set Fake Ringback or None.
RTP Detect
This option is to disconnect call when there is no RTP received. Default value is 90s
Auto CLIP Routing
Same callee route to same port, Force means if the port is busy, the call can’t call through the device even there is idle port.
Nat Traversal
Include Static NAT, Dynamic NAT and STUN
STUN (Simple Traversal of UDP over NATs) is a network protocol. It is allowed to stay behind the NAT (or multiple NAT) client part to identify their clients’ public address, found himself after what Type of NAT and NAT for a particular Channel is bound to a local Internet terminal Channel. This information is used for two host to set up UDP communication behind the same NAT router. The agreement defined by the RFC 3489
Other configuration Enable Private Service
To enable local services like *158# etc.
User ID Is Phone Number
Default is No. user=phone will be added in SIP message body when this option enabled.
Reject Anonymous call from IP to PSTN The incoming anonymous calls will be rejected Use # as End Key
In General, SIP phones are based on # as the end, if this option is set to No, the dial-up will end expires dial-up time
No Answer Timeout
How long time hang up the call if callee no answered.
Interdigit Timeout
Timeout without dialing
Reset ASR after SIM Switching
Reset ASR or not after SIM Switch
4.9.11 Media parameter
Local Start RTP Port
Means the initial port when RTP voice stream transmit in the IP network, in general, using the factory default values. When there are several Ultiroam units are deployed and they are in the same network or behind the same NAT, user can try to change it to avoid NAT traversal issue;
Enable Silence Suppression
Enable the "silence suppression" almost no impact on call quality, and can save about half of the bandwidth;
Enable Busy Tone Detect
As usual, we detect Reverse Polarity then hang up the call, if gsm don’t send Reverse Polarity, you can enable Busy Tone Detect.
Call Progress Tone
Each country has its different call progress tone required standards, such as busy tone, ring back tones and ring tone standards, users can select the area standard from here
USA Standard:
Ringback Tone: 440,280,480,280,2000,4000,0,0 frequency: 440/480Hz on:2000ms off:4000ms Busy Tone: 480, 330, 620, 330, 500, 500, 0, 0 frequency: 480/620Hz, on: 500ms off: 500ms
DTMF Parameter
UC2000-VE/F/G support RFC2833 and SIGNAL two ways. DTMF INTERVAL range is 50 ~ 800ms, DTMF VOLUME can use the default Configuration
System IVR
While you make call to SIM card of GSM gateway, you will hear default IVR prompts or customized IVR.
Configure codec list
4.9.12 DBO Parameter Enable DBO service
Configure DBO parameter
More parameter showing on the interface after enable DBO, the main interface as below:
Parameter Description:
Parameters Description
DBO Local Port (0 means Random Port) Which port use to conenct dbo in device Active DBO Server URL/IP Primary DBO server IP or domain for traffics Active DBO Server Port DBO service ports that dedicate by DBO server. There are 4 ports definite in the DBO server by default, 3479, 6479, 12479 and 24479, any one of this 4 ports will work with the DBO server. Active DBO Server Username The authenticate username which provide by DBO server. The gateway will not allow to pass the traffics if the username and password doesn’t match with the server. The username with the format as x.x.x.x_3479 by default. x.x.x.x is the IP of DBO server. Active DBO Server Password
The authenticate password which provide by DBO server. The gateway will not allow to pass the traffics if the username and password match with the server. Standby DBO Server URL/IP Secondary DBO server IP or domain. Standby DBO Server Port DBO service ports that dedicate by DBO server. There are 4 ports definite in the DBO server by default, 3479, 6479, 12479 and 24479, any one of this 4 ports will work with the DBO server. Standby DBO Server Username The authenticate username which provide by DBO server. The gateway will not allow to pass the traffics if the username and password match with the server. The username with the format as x.x.x.x_3479 by default. x.x.x.x is the IP of DBO server. Standby DBO Server Password
The authenticate password which provide by DBO server. The gateway will not allow to pass the traffics if the username and password match with the server. Enable SIP Forwarding Enable SIP signaling encryption and forward by DBO server. The SIP signaling will forward by
DBO server after this option enable.
Enable RTP Forwarding
Enable RTP encryption and forward by DBO server. The RTP will forward by DBO server after this option enable.
Enable Bandwidth Compressed
Enable bandwidth saving function. This feature works after uploading proper license.
4.10 Human behavior
4.10.1 Overview On the Overview interview, you can see the number, last matched balance (the balance that is assigned last time), calculated balance (the remaining balance), remaining total, monthly, daily credits and remaining daily, hourly call counts of a SIM card.
4.10.2 Basic Configuration On the Basic Configuration interface, you can set how long an IP àTel call or a TelàIP call will be delayed, as well as call interval. The ‘set call volume threshold function’ is mainly used for anti-blocked (such as some operators
launched special call testing for the detection of the VoIP equipment, call volume may is mute or great noise) .
Tel to IP Call Delay
Incoming call reach device, device will delay the secs to send to IP side.
Startup Interval
Module power on time interval. when device power on, all module won’t power on at same time, they will power on one by one.
IP to Tel Call Delay
Outgoing call reach device, device will delay the secs to send to GSM side.
Call Interval
When one call end, the port will rest the time, if you set 5-120secs, it means the port will rest min 5secs, max 120secs.
No Alerting Call Handle
Outgoing call don’t have alerting before receive Reverse Polarity, we can choose Normal Handle, Hang Up or Not Answer. Normal Handle: Call will normal active.
Hang Up: Call will hang up by device.
Not Answer: Call won’t connect, call will timeout or caller cancel it.
IP to TEL Processing Timeout Handle
Enable Processing Timeout Handle, you can set timeout time.
Set Call Volume Threshold
Enable the Call Volume Threshold, if the Volume is lower or higher than the threshold you set, call will be hanged up by device.
SMS Sending Delay
SMS send interval, when one SMS send out, next one will delay send out.
Numeric Scale
How many digits displayed after the decimal point in balance.
GSM incoming call limit Limit the incoming call duration.
Setting of Multi-SIM SIM Switching Setting
This setting for Multi-SIM device, like 8/32,16/64,32/128,four slots for one module, you can set switchover card by SIM running time, call counts, call time, and sms counts. Enable Query SIM information during initiation, when device power on, all solts cards will register one by one to get their info, like sim number, sim balance, if you have set auto balance check and number study.
4.10.3 Phone Number Learning If you want to learn the SIM card number and used for auto call. The GSM gateway provide 3 modes to learn SIM card number: USSD/SMS/Call.
1) USSD. Send USSD to carrier and get the response. For example, send *156#, get response: “Your number is 8618344144906”. So, configured the Keywords to “Your number is”, the gateway will take the number 8618344144906, but local number is 18344144906, you need delete the 86
For make sure the configuration work, we can use the Matching Test. Input the “Your number is 8618344144906” at Test SMS Text, press the Test, you will get the match result.
2) SMS.
Send SMS to carrier and get the response. For example, send SMS “My Number” to 10086, the carrier reply SMS: “Your number is 8618344144906”. So, configured the Dest Number to 10086, the Send Text to “My Number”, the Check SMS From Number to 10086, the Keywords to “Your number is:”, the gateway will take the number 8618344144906, you can delete or add prefix.
For make sure the configuration work, we can use the Matching Test. Input the “Your number is 8618344144906” at Test SMS Text, press the Test, you will get the match result.
3) Call.
Call to carrier and get the response. For example, call 10086, after call connected, it will play IVR “welcome to use China Mobile, recharge, press 1; check balance, press 2; other services, press 3 ...” press 3, it will play IVR “check current package, press 1; check phone number, press 2;…”, press 2, the carrier reply MSG: “Your number is 8618344144906”. So, configured the Dest Number to 10086, the Send Text to p5,3,p3,2 that means after call connected wait 5s, then press 3, then wait 3s, then press 2. the Check SMS From Number to Null, the Keywords to “Your number is”, the gateway will take the number 8618344144906.
For make sure the configuration work, we can use the Matching Test. Input the “Your number is 8618344144906” at Test SMS Text, press the Test, you will get the match result.
4.10.4 Balance Check On the Balance Check interface, you can check the balance of a SIM card.
If you want to check balance automatically and block SIM card when it is low balance. The UC2000 have 3 modes to check balance: USSD/SMS/Call.
1) Check balance by USSD Send USSD to carrier and get the response. For example, send *101#, get response: “Your balance is 73.40$”. So configured the Keywords to “Your balance is”, the gateway will take the number 73.40. Balance check condition can be time, balance threshold and call counts.
For make sure the configuration work, we can use the Matching Test. Input the “Your balance is 73.40$” at Test SMS Text, press the Test, you will get the match result.
(next page)
2) Check balance by SMS.
Send SMS to carrier and get the response. For example, send SMS “My balance” to 10086, the carrier reply SMS: “Your balance is 73.40$”. So configured the Dest Number to 10086, the Send Text to “My balance”, the Check SMS From Number 10086, the Keywords to “Your balance is”, the gateway will take the number 73.40.
For make sure the configuration work, we can use the Matching Test. Input the “Your balance is 73.40$” at Test SMS Text, press the Test, you will get the match result.
3) Check balance by Call. Call to carrier and get the response. For example, call 10086, after call connected, it will play IVR “welcome to use China Mobile, recharge, press 1; check phone
number, press 2; other services, press 3 ...” press 3, it will play IVR “check current package, press 1; check balance, press 2;…”, press 2, the carrier reply MSG: “Your balance is 73.40$”. So, configured the Dest Number to 10086, the Send Text to p5,3,p3,2 that means after call connected wait 5s, then press 3, then wait 3s, then press 2. the Check SMS From Number to Null, the Keywords to “Your balance is”, the gateway will take the number 73.40.
For make sure the configuration work, we can use the Matching Test. Input the “Your balance is 73.40$” at Test SMS Text, press the Test, you will get the match result.
4.10.5 Billing setting Billing setting mainly use to limit call time of SIM cards, see also call limit.
Minimum Charging Time: set minimum charging time, some operator does not charge if the call is less than some seconds when call is connected, user can set that value here. If the operator starts billing once the call is connected, please set 0 here. In this example: set 1$ per 60s for port group 0.
4.10.6 Call limit
Single Call Duration: set single call duration, it defines the maximum duration every single call can take, 0 means no limit. If you set 40, it means every call can last 40secs at most, and call will be disconnected if gets the limit. Total Credits: set total credits, it defines the maximum credit the port can use, 0 means no limit. If you set 600, it means the port can use 600 credits at most. Monthly Credits: set monthly credits, it defines the maximum credit the port can use in one day, 0 means no limit. If you set 300, it means the port can use 300 at most one month, and the data will be cleared at Reset Monthly Date. Daily Credits: set daily credits, it defines the maximum credit the port can use in one day, 0 means no limit. If you set 30, it means the port can use 30 at most one day, and the data will be cleared at 0’clock of everyday. Daily Calls: set daily calls, it defines the maximum counts the port can use in one day, 0 means no limit. If you set 30, it means the port can call 30 counts at most one day, and the data will be cleared at 0’clock of everyday Hourly Calls: set hourly calls, it defines the maximum counts the port can use in one hour, 0 means no limit. If you set 10, it means the port can call 10 counts at most one hour, and the data will be cleared next hour. Daily Connected Counts: set daily Daily Connected Counts, it defines the maximum counts the port can use in one day, 0 means no limit. If you set 20, it means the port can call 20 connect calls at most one day, and the data will be cleared at 0’clock of everyday Adjust Credits Automatically: If enable adjust credits automatically or not, Yes means enable, No means disable. This option is used to work together with Balance Check function, when enable both balance check function and billing, the gateway will automatically regulate the balance. Low Credits Warning: When the total credit reaches the setting, it will send sms to the cell phone number you set. In this case, billing unit = 1$/60s, total credits = 300 Call limitation = 300/1 = 300 minutes
4.10.7
Exception Event Handling
Call Event
1. Definitions For the purpose of the present document, the following terms and definitions apply:
ACD: The Average Call Duration is a measurement in telecommunications that reflects an average length of telephone calls transmitted on telecommunication networks. ASR: The Answer-seizure ratio is a call success rate in telecommunications; it is the percentage of answered telephone calls with respect to the total call volume. CDR: The Call Detail Record is a data record produced by a telephone exchange or other telecommunication equipment that documents the details of a telephone call that passes through the facility or the device. 2. Configurations Low ASR Handling The ASR is equal to: the answered call, divided by the total attempts of calls. That is,
ASR = answered call/total attempts of calls. To calculate the ASR, the gateway checks the CDRs. Because the CDRs on the gateway is disabled by default, you need to enable the CDR before you apply the Low ASR handling. 3. Enable CDRs on the gateway Open the web of the gateway, and then click “Statistics” and “CDR Report”. Then enable the CDR as the below figure shows:
Don’t forget to click “save” after selecting “Yes” on Enable CDR.
4. Configure the Low abnormal call handle Click “Human Behavior” and “exception event handle”, then select “yes”, the configuration page will be displayed:
Low ASR Less Than: This value is the threshold of the ASR, once the exact ASR is lower than this value, the UC2000 port will be considered as the low ASR.
Low ACD Less Than: define the low ACD value threshold once the exact ACD is lower than this value, the UC2000 port will be considered as low ACD. Counts of Recent Call: This value defines how many recent calls will be counted to calculate the ASR/ACD. Counts of failed calls: This value defines how many failed calls. This feature is used to detect the failure calls, once there are certain counts of call failure consecutively, the gateway port will be considered abnormal.
Low Balance Than: define the low balance value threshold. To apply the Low Balance Handle, it is required to configure the Balance check properly; please refer to the FAQ of balance check for more details.
GSM Network side error code handle When the gateway makes an outgoing call, GSM network side will respond a code which indicates the cause of the call of failure; gateway will record these error codes until the gateway was restarted. The error code 8, meaning of “Operator determined barring”, indicates precisely that the SIM was blocked by operator; so, we provide this feature to detect the error code and then blocked gateway module. Follow these steps to use this feature:
a) Enable the error code record
The GSM network side error code record is disabled by default, you need to enable the record before you use this feature. Click “System Configuration” and “SIP Parameter”, then select “yes” for “GSM-Sip Code Map GSM Code Enable”.
Don’t forget to save the configuration.
b) Configure the GSM code monitor
By GSM Code: The GSM network side error code
Counts of consecutive GSM Code: The counts of the error code consecutively.
As the figure shows above, once the GSM error code 8 is detected, the gateway will be blocked the gateway module. PDD Less Than (1-30): define the value of abnormal PDD. You can check PDD value under system information page.
Handle abnormal event
Once one of the above abnormal conditions is detected, gateway could: Reset the specified GSM module
Block the specified GSM module
Block the SIM, this setting only available while remote SIM mode is in using or multiple SIM device. SMS Test, send a SMS through specific port to verify if the SIM card works properly Sending SMS to a phone number for alerting, this is optional.
USSD Event
Reset module/block Port/Block SIM card in case of USSD failed more than defined value threshold.
USSD/SMS Monitor
This parameter be used to Monitor SMS/USSD response contents, which helps gateway to know SIM card is blocked.
USSD/SMS URL Monitor
This parameter be used to Monitor SMS/USSD response contents, which gets the right keywords, access the internet.
SIM Register Fail
This parameter be used to Monitor the cards register status.
Abnormal BCCH
This parameter be used to Monitor the BCCH, the bcch isn’t in the whitelist, the module will power down.
SMS Test
When sim blocked by Call Event Monitor, it will send sms to confirm again. Send sms success, the sim will be unblocked.
Alerting Setting
Set SMS or Email Alerting send.
Email Alerting: Before using this function, please confirm the setting in "SMS and USSD - Email " has been set correctly.
4.10.8 Auto generation
Auto Generation mainly used to make calls and SMS between SIM cards which in same device, also you can make call or send SMS to other numbers. Why need Auto Generation?
Because the device used to call out as a landing, the large number of outgoing easily be detected abnormality, so we need auto generation incoming calls, outgoing calls which between different operators. Basic Settings:
Auto Call:
Prefix to Add and Deleted: when call the other ports, modify the prefix you want. Called by other ports: Auto call between the same device ports. Note: Auto Generation between SIM cards must learn number at first, please refer to Learn SIM Card Number section. Number Length: The number valid digits from right. for example, when you call in, device know the number 18612345678 will dial in, but caller show +8618612345678, device will reject the call, if we set the valid number length as 11, it will only check the last 11 digits, the call will allow pass.
Call Out: We can set to call fixed numbers Import Numbers: Choose the file, then save the text file as .txt format. How to make the txt file? The length of each number up to 22 digits, use “,” separated, we can only input 600 digits in one file (Include commas)
Number of retries after call failure: After the automatic call failure, whether to retry.
Call Duration: you can set any time you want, Automatic call duration will between the Min and Max. Auto Send SMS: Auto SMS between the same device ports
Auto Internet access: Auto access internet to anti-block. How to enable Internet access? Enable Internet Access and set VPN to active this feature, Gateway won't enable this feature successfully if APN is blank or wrong. YOU can set url as you want, but support https or not decided by the module type, you can check access internet success or not under GSM Event.
Conditions Settings: define the value when auto SMS/Call generation start to work
1) By Device Online Time: SIM cards register in device time, every 30-120mins, it will make call or send SMS, Random intervals between 30-120minutes. 2) By Total Call Durations: When call out time reach 60mins, there will generate an automatic call or SMS. 3) By Consecutive Calls: There are 20 consecutive outgoing calls, there will generate an automatic call or SMS. But if there are 19 consecutive outgoing calls, the SIM card receive an incoming call, it will be re-count.
4.11 Diagnostic
4.11.1 Syslog
Syslog is a standard for network device data logging. It allows separation of the software that generates messages from the system that stores them and the software that reports and analyzes them. It also provides devices which would otherwise be unable to communicate a means to notify administrators of problems or performance. There are 5 levels of syslog, Including NONE, DEBUG, NOTICE, WARNING and ERROR. The Signal Log is including following traces which defined in system by default
- SD, hardware debug
- SIP, SIP signaling trace
- STUN, STUN logs
- ECC, detail information of call control modular
- RE, the common communication modular for SCP and SIM
- SCP, the communication protocol between gateway and cloud server
The media log is including following traces which defined in system by default
- RTP, RTP stream info collection
- SIM, to output traces between gateway and remote SIM cards
The System Log is including following traces which mainly used by developer
- SYS, system log
- TIMER, system process
- TASK, system task process
- CFM, system process
- NTP
The Management Log is including following traces which defined in system by default
- CLI, command line
- TEL,
- LOAD, firmware upload
- SNMP
- WEBS, embedded web server
- PROV, provisioning
Server Syslog:
When the gateway registers to SIM Cloud server, the option will be changed to un-configurable and all logs to be storage on server. 4.11.2 Filelog
The filelog includes signal log, media log and system log, you can enable it if you want to do some troubleshooting. Click download button to save the filelog. 4.11.3
Summary
Summary file is enabled by default. Just click download button in case of some of system error happened.
4.11.4 SIM card debug
Enable trace while remote SIM card used in this device.
4.11.5 Ping test you can use Ping to check whether the network is working or not.
4.11.6 Tracert Test You can check the routes of the tracert destination.
4.11.7 Network Capture Network capture is a very important diagnostic tool for maintenance. This section describes how to enable network capture. Voice stream transmit path of the gateway as below:
Getting start to PCM capture
PCM capture is help to analysis voice stream between GSM/CDMA modular and DSP chipset.
To enable PCM capture
w Select ‘PCM only’ on Network Capture page
w Click “Start’ to enable PCM capture
w Dialing out through gateway, start talking a short while then hangup the call.
w Click ‘Stop’ to disable network capture
w Save the capture file to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The sample of PCM capture as below:
Getting start to Syslog capture
Syslog capture is another way to obtain syslog which the same as remote syslog server and filelog. The capture file is saving as pcap format so that it can be opened in some of capture software like Wireshark, Ethereal software etc.
To enable syslog capture
w Select Syslog special only on Network Capture page
w Click “Start’ to enable syslog capture w Dialing out through gateway, start talking a short while then hangup the call. w Click ‘Stop’ to disable syslog capture w Save the capture to local computer The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The sample of syslog capture as below:
Getting start to RTP capture
PCM capture is help to analysis voice stream between gateway and remote IPPBX/SIP Server.
To enable RTP capture:
w Select RTP special on Network Capture page
w Click Start to enable RTP capture
w Dialing out through gateway, start talking a short while then hangup the call.
w Click Stop to disable RTP capture
w Save the capture to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The sample of RTP capture as below:
Getting start to DSP capture
DSP capture is help to analysis voice stream inside DSP chipset. The DSP chipset will handle RTP from IP network as well as voice stream from GSM/CDMA modular.
To enable DSP capture:
w Select DSP only on Network Capture page
w Click Start to enable DSP capture w Dialing out through gateway, start talking a short while then hangup the call. w Click Stop to disable DSP capture w Save the capture to local computer
The capture is named to ‘capture(x).pcap’, x is serial number of capture and will be added 1 in next time. The sample of RTP capture as below: Configurable capture options Getting start to custom capture
This menu provides more options to capture specific packets as actually needs.
4.11.8 Voice Loopback Test
Voice Loopback test should be done on call status. Each call can do one kind of test. After each test, please hang up and call again, refresh web interface and go on the other tests.
Voice stream patch on gateway:
DSP TDM Test
DSP TDM Test is the loopback of GSM side.
VoIP <-----DSP<----- Modular<------------------------------- Mobile
> ------> ------->
To start DSP TDM Test:
w Make a call test through gateway, the call can be initialed by IPPHONE. Keep the conversation after call establish w Click DSP TDM Test to start test w Check the voice on both sides. VoIP side become silence and echo should be generated on Mobile phone side w Hang up
To start DSP IP Test:
DSP IP Test is the loopback of VoIP side.
IPHONE------>VoIP----- > DSP
<------ <-------
To start DSP IP Test:
w Make a call test through gateway, the call can be initialed by IPPHONE. Keep the conversation after call establish w Click DSP IP Test to start test w Check the voice on both sides. GSM side become silence and echo should be generated on IPPHONE side w Hang up
4.11.9 Mobile Network Test GSM or WCDMA module test the call or register on web.
4.11.10 Module Recovery When module unknow or version lower, we can update directly from network, but not all module version can update, please contact technical support to help you if you need.
4.11.11 Web Operation Log Show the web operator log.
4.12 tools
4.12.1 File Upload
On the Tools à Firmware Upload, you can upload a firmware to upgrade the device. But you need to restart the device for the change to take effect. 4.12.2
Userboard Upgrade
Click Upgrade button while Status show as “FAULT”. This page mainly uses to reload the userboard firmware. 4.12.3 Config Restore and Backup
Backup or restore config file of the device.
You can restore this configuration in case the unit loses it for any reason or to clone a unit with the configuration of another unit. The configuration backup configurations are in txt format. Please note that you can use a backup file from an older firmware version and use it in a unit with a more recent firmware version. However, a backup file from a newer firmware version than the one actual in the unit cannot be used for a restore operation on the unit.
4.12.4
Management Parameter
Parameters
Description
NTP Parameter
The Network Time Protocol (NTP) is a protocol and software implementation for synchronizing the clocks of computer systems over packet-switched, variable-latency data networks. User need to fill the NTP Server Address and select Time Zone
Web Port
Default is 80
Telnet Port
Default is 23
4.12.5 Remote Server
While devices deployed behind router/firewall. Users can’t access the device remotely. With the Remote Server, device can register to it and access web/telnet through Remote Server remotely. Register Remote Server account from web site server02.dmcld.com:3000
4.12.6
Email Account Setting
Please refer to section. SMS and USSD -> Email.
4.12.7 Username and Password
When using web or telnet Configuration, please enter default user name and password. User can modify the login name and password.
4.12.8 Access control
You need to set a new password to control access level of web links. After set password, you can set which page is allow/disallow to access by default user.
4.12.9 Factory Reset
Be careful do this operation, after restore factory setting, all the parameters will be changed to the factory default.
4.12.10 Auto Restart Configure auto restart at pre-defined HH/MM
4.12.11 Restart
5 Troubleshooting and Command Line
5.1 Login UC2000 & General Knowledge of UC2000 Command
This is a document for some customers who need more details of Ultiroam’s products with command lines. To make sure the system runs successfully, we suggest customers setting UC2000 by GUI. In this manual, some topics such as how to check the IP, signaling and call conversation are covered.
Tips: The document is fit for all UC2000-VE/F/G models.
Run system tool Telnet to login UC2000. The default username and password is "admin". C:\Users\Administrator>telnet 172.16.101.142
Input "?" to show the all commands and its information.
Abbreviation is supported in UC2000 command. For example you can input "en" substitute for "enable", input "sh" substitute for "show", input "cl" substitute for "clock",
5.2. Commands in "ROS#" Mode
There is only a litter commands in "ROS>" mode. If you need more commands you must enter the "ROS#" mode. Input "enable" to enter "ROS#" mode if you have in the "ROS>" mode.
5.2.1 Summarize of commands in "ROS#" mode Input "?" to get the information of all commands in "ROS#" mode.
5.2.2 General Purpose Commands in "ROS#" mode
Show IP address (show int) Show Time (show clock) Show version (show version)
Show sip Information (show sip config)
Show memory status (show memory detail) Show SIP port status (show sip all) Show Current calls (sh ecc call)
Show RTP session ( sho rtp se) Show ASR/ACD statistics (show ecc state)
5.3 COMMANDS in "Config" Mode
5.3.1 Summarize of commands in "config" mode Input "^config" in the "ROS# " to enter "config" mode.
Input "?" to show the all commands and its information.
5.3.2 General Purpose Commands in "Config" mode
Set time (clock set) Save the configuration (save) Restart device (reset eia) Enable debug
The command format is deb port + port number, to enable port 0 debug, as below:
To enable all ports debug, with the command “deb port all”
Without these steps, no trace logs will display on output window
Enable SIP debug (deb sip msg all)
5.4 How to trace SIP logs Create telnet session to gateway, the main steps as below: Welcome to Command Shell! Username:admin Password:***** ROS>en ROS#
ROS#^config ROS(config)#deb sip msg all ROS(config)#ex ROS#
ROS#^ada
ROS(ada)#ADA CONNECTED ...,WELCOME!
ROS(ada)# ROS(ada)#turnon 71 Disable sip trace: ROS(ada)#turnoff 71 5.5 How to trace ECC logs (Call Details)
Welcome to Command Shell! Username:admin Password:*****
ROS>en
ROS#
ROS#^config ROS(config)#deb port all Debug All!!. //enable trace on all port ROS(config)# ROS(config)#deb port 0 Succ! Debug PortNo:0 // enable trace port 0 ROS(config)# ROS(config)#no deb port all ROS(config)# ROS(config)#ex ROS#^ada
ROS(ada)#ADA CONNECTED ...,WELCOME!
ROS(ada)#turnon 84 Disable trace: ROS(ada)#turnoff 84
5.6 How to trace Modular logs
Welcome to Command Shell! Username:admin Password:*****
ROS>en ROS#^ada
ROS(ada)#ADA CONNECTED ...,WELCOME!
ROS(ada)#cmd 53 19 0 0 1
// enable trace. 0 0 means port range 0 to 0, 0 8 means port range from 0 to 8; 1 means enable modular trace ROS(ada)#cmd 53 19 0 0 0 //disable modular trace
6 Glossary
GSM: Global System for Mobile Communications CDMA: Code Division Multiple Access FMC: Fixed Mobile Convergence SIP: Session Initiation Protocol MGCP: Media Gateway Control Protocol DTMF: Dual Tone Multi Frequency USSD: Unstructured Supplementary Service Data PSTN: Public Switched Telephone Network STUN: Simple Traversal of UDP over NAT IVR: Interactive Voice Response
IMSI: International Mobile Subscriber Identification Number IMEI: International Mobile Equipment Identity DMZ: Demilitarized Zone
API: Application programming Interface BCCH: Broadcast Control Channel LAC: Location Area Code CID: Cell ID
BTS: Base Transceiver Station DTMF: Dual-Tone Multifrequency IVR: Interactive Voice Response
NAT: Network Address Translation RTP: Real-time Transport Protocol VoIP: Voice over IP