v 3.9 2015
GSM Gateway (hereinafter G20) functions as a multichannel GSM bridge between mobile communication operators and PBXs or IP subscriber networks via so called soft switches. The gateway SW operates in Linux OS with 2.6.33 core. Exchange interaction link is PRI30 (EDSS1). IP subscribers interaction link is LAN (protocol – SIP, codecs – G711a/u, G729, G723, G726-40, G726-32, G726-24, G726-16).
Joint operation of IP and PRI links is supported, however only by individual GSM channel daughter cards (a daughter card has 4 GSM channels and one DSP voice signal processor). The configuration of the daughter cards modes of operation is specified in /vrem/cards.ini file.
G20 may be supplied with different numbers of GSM radio channels – 4 to 24 channels.
The gateway back panel has the following connectors, from left to right:
The gateway front panel has:
The gateway is controlled via LAN using an Internet browser.
It is recommended to use the latest versions of Google Chrome or Opera browser.
Usually, for restarting the operating SW of the gateway (unlike the whole system), it is sufficient:
To get started, use the following procedure:
Then (approximately in 5 seconds) the operational system and the unit operating SW launch process starts. If launching process does not start automatically, Reset button (recessed in the hole) may be pressed.
Then, to set up the unit (incoming and outgoing calls routing, radio channels (RC) parameters and voice processors (DSP)), follow the recommendations on the next page.
Please, find FAQ Section in the end of this manual.
The unit supports SMSs sending/receiving. The critical interfaces helping to process SMSs are:
If an SMS is received from the GSM network in portions, the unit "patches" them together.
To get access to the gateway control and setup, enter http://192.168.2.77 into the address line of the browser where 192.168.2.77 is the IP address of the gateway (please, ask your supplier about the specific address when purchasing the equipment). You can change the address. For details see /etc/rc.d/rc.network.
When the gateway SW is launched (the process starts automatically after the power is switched on), the main gateway control page is displayed after you enter the login and password. With PRI link, this page may look like this:
With SIP server, the page looks like this:
Login/password for web interface:
The passwords may be changed using a standard htpasswd utility included in HTTPD web server package (naturally, only the 'root' user can get access to his utility).
Login/password for ssh:
user - /please, ask your equipment supplier/
sysadmin - /please, ask your equipment supplier/
Note:
access to the gateway (via SSH) using a sysadmin password is PROHIBITED (that is you cannot do this, even don’t try), therefore you need to access the system with user password first, and then, using su - command, to switch to the administrator mode with 'root' user password.
The passwords may be changed using standard Linux operators. However, after changing the passwords, you'll need to reset the unit.
MAIN PAGE
This page displays the radio channels status and PRI channel status - PRI STATUS or the status of registration with an external SIP server.
Select Command (Выбор команды) – menu for selecting a command transmitted to the device.
These commands are listed below:
Lock All RCs (Блокировать все РК) – this command locks all the RCs in the gateway so that no communication is possible (all the channels are switched off).
Unlock All RCs (Разблокировать все РК) – this command sets all the radio channels to their initial position – Call Waiting (ожидание вызова).
Enable Callback Channels Use (Разрешить занятие CALLBACK каналов) – this command enables using the GSM channels dedicated for receiving the incoming callback mode calls for outgoing calls from PRI to GSM.
Disable Callback Channels Use (Запретить занятие CALLBACK каналов) – this command disables using the GSM channels dedicated for receiving the incoming callback mode calls for outgoing calls from PRI to GSM.
Restart Vinetic 1, ..., Restart Vinetic 6 (Перезапустить Vinetic 1, ..., Перезапустить Vinetic 6) – these commands allow for restarting DSP processors on radio channels daughter cards (each daughter card has one DSP and 4 RCs). This command does not result in termination of all the connections at the respective RC daughter card, however, the conversation in the voice path stops for the time of the DSP restarting (~2-5 seconds) then the voice path becomes available again and the conversation may be continued.
Stop Net Thread (Остановить Net Thread) – this command stops the program SMS server in the gateway that is used for sending, receiving SMSs and for reading and replenishing the balances of the gateway SIM cards (the server serves a specialized SMS client described in the respective document of the equipment manufacturer).
Start Net Thread (Стартовать Net Thread) – this command starts the program SMS server in the gateway.
Balance Request (Запрос баланса) – the request of the account status of the SIM card installed in the RC for which the protocol recording is activated – P (П) checkbox of the channel is ticked.
Balance Request by all the Radio Channels (Запрос баланса по всем радиоканалам) – helps to request the balance status by all the radio channels at a time, the received amounts may be viewed in the side SIM Card menu. If no balance request code was assigned to the channel (Routing - PSTN->GSM and Channel Parameters pages), the command for this channel won't be executed.
Replenish Account (Пополнить счет) – the account is replenished for the SIM card installed in the RC for which the protocol recording is activated – P (П) checkbox of the channel is ticked; the replenishment code and voucher code shall be prior specified for the radio channel on SIM Cards (voucher code), Routing - PSTN->GSM (replenishment code for the GSM operator), Channel Parameters (assignment of the GSM operator to the radio channel) pages.
Module_Off, Module_On, Module_Pwr_Off, Module_Pwr_On – these commands are used for switching on and off the modules of the radio channel with enabled protocol logging – P (П) checkbox of the channel is ticked. These commands are applied by the MANUFACTURER SPECIALISTS ONLY.
Enable AT Commands Protocol (Включить протокол АТ-команд) – detailed recording of AT commands protocol for the RC modules is enabled. The commands are recorded to at_log.txt (in text or HEX format). The protocol is logged by the RC with enabled protocol logging – P (П) checkbox of the channel is ticked
Disable AT Commands Protocol (Выключить протокол АТ-команд) – cancels the previous command (this is a default setting).
RC Gain Ratio Request (Запрос коэф. усиления в РК) – the command requests the RC module about the voice gain ratios for the mobile subscriber end and for PRI/SIP subscriber end.
Set RC Gain (Установить усиление в РК) – the command sets the voice gain factor values (the gain factor values are set on Channel Parameters page).
Read SIM Card Number (Читать номер SIM-карты) – the command reads the subscriber number of the SIM card installed in the RC (please, note that not all the SIM card allow for this command completion). The number itself may be viewed on SIM Cards page or on the main page by placing the mouse cursor on the SIM card item number.
Enable DSS1/SIP Protocol (Включить протокол DSS1/SIP) – this command enables the logging of PRI link commands into dss_log.txt / sip_log.txt protocol file.
Disable DSS1/SIP Protocol (Выключить протокол DSS1/SIP) – this command cancels the previous command (this is a default setting).
Enable Protocol Log for RC (Включить запись протока по РК) – the protocol of commands for the RC is recorded in rk_log.txt file.
Disable Protocol Log for RC (Выключить запись протока по РК) – cancels the previous command (this is a default setting).
Store Settings to BACKUP.BIN (Сбросить настройки в BACKUP.BIN) – this command stores the current gateway settings to backup.bin file. If the gateway is restarted correctly the settings are stored to this file automatically.
Enable 425 Hz To RC End (Включить 425 Гц в сторону РК) – this commands allows for transmitting a continuous signal 425 Hz to the mobile subscriber end via the voice path. This is a maintenance command not used under the normal gateway operation conditions.
Disable 425 Hz To RC End (Выключить 425 Гц в сторону РК) – this commands cancels the previous command, i. e. terminates the transmission of a continuous signal 425 Hz to the mobile subscriber end via the voice path. This is a maintenance command not used under the normal gateway operation conditions.
Include 425 Hz To Path (Включить 425 Гц в тракт) – this command allows for transmitting a continuous signal 425 Hz to both ends of the voice path – APPLIED ONLY to PRI. This is a maintenance command not used under the normal gateway operation conditions.
Disable 425 Hz To Path (Выключить 425 Гц в тракт) – this command cancels the previous command, i. e. terminates the transmission a continuous signal 425 Hz to the voice path – APPLIED ONLY to PRI. This is a maintenance command not used under the normal gateway operation conditions.
Enable SMS Server Net Protocol (Включить NET протокол SMS сервера) – this command enables recording of exchange protocol between the gateway SMS server and SMS client to net_log.txt file.
Disable SMS Server Net Protocol (Выключить NET протокол SMS сервера) – this command cancels the previous command, i. e. disables recording of exchange protocol between the gateway SMS server and SMS client to net_log.txt file (the default setting is option enabled).
Enable SMS Protocol (Включить протокол SMS сообщений) – this command enables recording of SMS receiving and decoding protocol to sms_log.txt.
Disable SMS Protocol (Выключить протокол SMS сообщений) – this command cancels the previous command, i. e. disables recording of SMS receiving and decoding protocol to sms_log.txt (the default setting is option disabled).
Enable Echo Canceler On /SIP (Near End) (Включить эхоподавитель на PCM/SIP (near end)) – the command enables echo canceling function at PRI/SIP subscriber end (the default setting is echo canceler enabled).
Disable Echo Canceler On PCM/SIP (Near End) (Выключить эхоподавитель на PCM/SIP (near end)) – the command cancels the previous command, i. e. disables echo canceling function at PRI/SIP end.
Enable Echo Canceler On GSM (Far End) (Включить эхоподавитель на GSM (far end)) – the command enables echo canceling function at GSM subscriber end (the default setting is echo canceler disabled).
Disable Echo Canceler On GSM (Far End) (Выключить эхоподавитель на GSM (far end)) – the command cancels the previous command, i. e. disables echo canceling function at GSM subscriber end.
Reload DSS chip – the command allows for resetting and rebooting the PRI channel (WARNING: if executed, this command breaks all the existing links and result in rebooting all the DSP processors in the unit).
Enable Detailed SIP/DSP Protocol (Включить детальный протокол SIP/DSP) – this command allows for recording the protocol of additional process information into a file. This is a maintenance command not used under the normal gateway operation conditions.
Disable Detailed SIP/DSP Protocol (Выключить детальный протокол SIP/DSP) – this command cancels the previous command. This is a maintenance command not used under the normal gateway operation conditions.
Reset UDP Socket Counter (Сбросить счетчик UDP сокет) – this command SHALL NOT be applied in the normal course of the gateway operation and is used ONLY for maintenance purposes.
Reset RC Minute Counter To Zero (Сбросить счетчик минут на РК в нуль) – this command resets the counters of conversation minutes of one RC to all the ends.
Reset ALL RC Minute Counters To Zero (Сбросить счетчик минут на ВСЕХ РК в нуль) – this command resets the counters of conversation minutes of all the RCs to all the ends.
Read ROM CROSS_BOARD Version (Прочитать версию ROM CROSS_BOARD) – this command reads the firmware version on the gateway cross-card. The result of this command execution may be viewed on AT Commands page in Response Text window.
Set Delete Mail From Server Flag (Установить флаг удаления почты с сервера) – this command allows for deleting messages from the mail server even if the message was not sent by SMS due to the absence of available radio channels for SMS sending.
Remove Delete Mail From Server Flag (Сбросить флаг удаления почты с сервера) – this command cancels the previous command. That is the flag is removed and a message will NOT be deleted from a mail server if it was not sent by SMS due to the absence of available radio channels for SMS sending (channels for SMS sending are set on Channel Parameters page – For SMS parameter). Please, note that the gateway will still request this undeleted message and try to send it as soon as at least one available channel appears. The frequency of the mail server addressing is set on General Settings page (timer 7 - TimeWaitCheck).
Enable successful connections data logging to cdr_bill.csv file.
Disable successful connections data logging to cdr_bill.csv file (this is the default setting).
Enable detailed TCP protocol logging (AllNetLog) to net_log.txt file.
Disable detailed TCP protocol logging (AllNetLog) to net_log.txt file (this is the default setting).
Enable detailed SMTP protocol logging (AllSmtpLog) to smtp_log.txt file.
Disable detailed SMTP protocol logging (AllSmtpLog) to smtp_log.txt file.
Permit links interlock control (when working with a bank).
Forbid links interlock control (when working with a bank).
Enable FS and ATR storing in files (when working with a bank).
Disable FS and ATR storing in files (when working with a bank).
Enable SIM card protocol recording (when working with a bank).
Disable SIM card protocol recording (when working with a bank).
Enable recording to human_log.txt file (when working with a bank).
Disable recording to human_log.txt file (when working with a bank).
Enable recording to rtp_log.txt file.
Disable recording to rtp_log.txt file.
GSM operator base stations info.
Operators info.
Disable recording of all the assigned protocols.
Enable recording of all the assigned protocols.
Enable SMPPDUMP option (Включить SMPPDUMP опцию) – permits recording to /vrem/smpp_log.txt protocol file the gateway and SMPP client detailed communication info.
Disable SMPPDUMP option (Выключить SMPPDUMP опцию) – permits termination of recording to /vrem/smpp_log.txt protocol file the gateway and SMPP client detailed communication info.
Forbid SMS sending in SMPP/POP3 (Запретить отправку SMS в режиме SMPP/POP3) – only SMS receiving will be possible.
Permit SMS sending in SMPP/POP3 (Разрешить отправку SMS в режиме SMPP/POP3) – both SMS receiving and sending will be possible.
Reread IP addresses and port numbers (Перечитать IP адреса и номера портов) (from /etc/sipua.conf , ./vrem/net.in files).
Restart Gateway (Перезагрузка шлюза) – this command allows for correct restarting the gateway SW and OS.
Stop Gateway For Power Off (Остановить шлюз для выключения) – this command allows for terminating the gateway operation for failsafe power disconnection (after ~20 sec.).
Note:
1. The commands marked BLUE are applied to the radio channel with enabled protocol logging - P checkbox of the channel is ticked.
2. The commands marked BROWN are used with external SMS client.
3. The commands are executed after SELECT button is clicked.
4. The commands marked GREEN are used when using received SMSs sending to mail server over SMTP protocol.
5. The commands marked GRAY are used only for SMS receiving/sending over SMPP protocol.
This page is used for viewing and changing some general time and other gateway operation parameters.
PARAMETERS:
WaitBeforeMakeCallBack – delay time before a callback is made.
TimeLimit – maximum time of a conversation.
WaitFirstDigit – waiting time for the first digit of extension dialing.
WaitNextDigit – waiting time for the next digit of extension dialing.
CountLastDigit – the number of last digits of number A (a calling subscriber number). This parameter is required for correct CALLBACK mode operation.
MaxWaitAnswer – maximum waiting time for a mobile subscriber answer.
TimeWaitCheck – delay time before mail is requested from POP3 server.
PorogNumber – the threshold length of the called subscriber number (number B); the incoming calls from the GSM network with numbers shorter or equal to the threshold are directed over the PRI/SIP link, the numbers longer than the threshold are directed to the GSM network over another radio channel of the gateway.
MaxDigitInfoDSS – the expected number of digits in number B (the called subscriber number) when PRI link is operated in overlap mode (receiving the number digits in information elements).
Remove N digits from CalledNumber – cuts off the set number of digits (from the beginning) in the called subscriber number when called from PRI/SIP to GSM.
VoiceActivityDetection (for SIP only) – enabling (1) /disabling (0) of "silence" packages generation in case of no voice at the mobile subscriber end (this function is available for SIP mode only).
ComfortNoise (for SIP only) – enabling (1) /disabling (0) of "comfort noise" generation in case of no voice (this function is available for SIP mode only).
IK_With_RK (for DSS only) – enabling (1) /disabling (0) of GSM channels assigning to PCM channels (GSM-PCM Link page assessed from the web interface main page). The function is available for DSS mode only and is used for calls routing from PRI to GSM and back from GSM to PRI.
First select mode – enabling (1) /disabling (0) of the free radio channel search mode using "always first free starting from the beginning" algorithm.
ring183 (for SIP only) – voice path through connection mode when calling from SIP to GSM:
0 – if the mobile subscriber is found (their phone rings),
1 – after the mobile subscriber's number is dialed,
2 – after the mobile subscriber answers.
This page is used for managing the mobile subscribers data base (DB). The DB is used for offering CALLBACK service via a dedicated RC (the channel is assigned in CALLBACK field on Channel Parameters page). The dedicated RC receives an incoming call from a subscriber, then the subscriber number is checked with the DB and the subscriber is sent “canceled” signal. In case of successful check, the callback is initiated and, after the subscriber answers the call, he is sent an invitation signal to dial either a number of another mobile subscriber (if * is placed before the number), or an exchange subscriber number; after the full number is dialed enter #. In case of a mistake, you can press * and start dialing again.
The DB contains the numbers of mobile subscribers (up to 128) and option flags for them.
The DB operation commands are:
Number and name of each subscriber is entered in Number:name field with ":" (two-spot) as a separation symbol; the command is entered to Command field.
The checkboxes are used for placing and removal of the option flags
CONF ? - conference flag
GSM ? - GSM to GSM communication flag
PRI ? - GSM to PRI communication flag
SIP ? - GSM to SIP communication flag
Black ? - lock any communication – “black number”
Base (База) window displays the numbers of mobile subscribers with their options.
Currently, CALLBACK service by option flags is not completely available. Only Black flag is an exception: if this option is selected for a subscriber, no calls from his/her number will be accepted and CALLBACK service won’t be provided.
NOTE.
1. Two permission flags are supported:
06756842** – matches the range of numbers from 0675684200 to 0675684299
or
067568**** – matches the range of numbers from 0675680000 to 0675689999
CALLBACK service is offered subject to respective settings are made on Channel Parameters page for the RC.
SW versions 14.2.6 and higher support callback function by SMS
if the following format SMS is sent to the channel assigned for CALLBACK:
callback:first_number:second_number
where
first_number is the first number of a mobile subscriber
second_number is the number of an internal (SIP or PRI) subscriber if its length does not exceed PorogNumber value on General Settings page; otherwise, it's the number of the second mobile subscriber the first mobile subscriber will be connected with, the second mobile subscriber will be called via another gateway radio channel or via the same radio channel as the first subscriber (CONF function) if * symbol is placed before the number, for example -
callback:0675551234:*0675551104 is the conference call of two mobile subscribers over one and the same channel (in this case, the channel should be flagged as "Conf+CBack" on Channel Parameters page),
callback:0675551234:0675551104 is the connection of two mobile subscribers using two radio channels.
If first_number is not found in the gateway database or is marked as "black number", CALLBACK serive is not provided.
If first_number is missing in the callback request like this:
callback::0675551104
the number the SMS was received from will be used as the first_number.
If second_number is missing in the callback request like this:
callback:0675551234:
IVR number assinged for this channel on Routing - GSM->PSTN page will be used as second_number.
If DTMF, extension dialing (Enable DTMF is ticked on Channel Parameters page), is permitted for the channel, second_number will be dialed by the first mobile subscriber when making the connection and the second subscriber, either internal or mobile (by comparing the dialed number will length with the PorogNumber value on General Settings page), will be called only after the extension is dialed (second_number).
The data base is stored in /vrem/ra_stat.ini text file. IT IS NOT RECOMMENDED to edit this file as it has a specific structure. Please, add/remove the numbers ONLY via the web interface.
This page is used for viewing info codes (IDs) of RC modules.
The current channel code numbers are displayed in the Current ID (Текущий ID) column only after the channel has passed initialization phases and has FREE (СВОБОДЕН) status.
The page is used for setting numbers of GSM to PRI/SIP calls routing for each RC if extension dialing (DTMF) option is not selected for the RC.
DTMF option is accessible from Channel Parameters page by Enable DTMF button.
With incoming call from GSM subscriber, the call will be directed to the number indicated at the existing radio channel provided DTMF receiving service is not enabled on this radio channel and this channel is not used for receiving calls for CallBack service.
In addition, here (only in SIP mode), IP address of SIP client the call is routed to may be set as a number, for example:
192.168.2.129 or: 8002@192.168.2.129
but not more than 18 symbols (for operational SW version 12.X the number of symbols shall not exceed 30).
Calls routing from GSM to SIP is also possible when SIM card number installed in this radio channel (the number may be viewed on SIM Cards page or on the main page) is set instead of the destination number (or destination address). This mode of calls routing form GSM subscribers end is set at the software launch using special licensed parameter (optional mode, license is required).
This page is used for assigning GSM operators to RCs, assigning codes for SIM card balances reading and replenishment, and for entering outgoing calls routing settings (to the GSM network end). In SW version 14.8.1 and higher, the page looks as follows:
As you can see, the page shows only the prefixes of communication operators with their assigning presented on page GSM Operator.
Note:
This page is used for replenishing the balances of the SIM cards installed in the RC. This page also shows GSM and IMCI numbers of the SIM cards.
You enter replenishment voucher code in Account Replenishment Code (Код пополнения счета) column, the operation code is set on Routing - PSTN->GSM Page page. The gateway stores the account replenishment codes when Send button is clicked. The accounts are replenished from the main page with Replenish Account (Пополнить счет) command.
SIM Card GSM Number (GSM номер SIM карты) column displays the card number in the mobile communication operator network. This number is requested by the gateway software automatically. However, not all the SIM cards “show” their numbers, therefore you can just enter the number in the respective line to view the SIM card number afterwards (naturally, you need to know the number).
In PIN Code (PIN код) column, the PIN code to be entered in reply to the respective inquiry is displayed if PIN code inquiry function is not disabled on the SIM card.
In Operator Code (Код оператора) column, mobile operator network code is displayed (such as for MTC – 25501, for KyivStar – 25503) and a checkbox of forced registration in the set operator network. These parameters are sored in the configuration file and allow, when the unit operational SW is started, for forced registration in the specific mobile communication operator's network. If the checkbox is not ticked, such forced registration is not performed, the operator's network is selected by GSM module based on the SIM card installed in this channel.
The page is used for adding and removing the GSM operators and, in SW version 14.8.1 and higher, for specifying direction prefixes by operator and rating type.
~ by seconds type: the conversation minutes are registered by simple summarizing the traffic in seconds,
~ by minutes type: each connection duration is analyzed using the following pattern: duration 1..60 sec. – 1 minute, 61..120 sec. – 2 minutes, 121..180 sec – 3 minutes and so on.
P.S. Remove Operator – this results in removing all the operator's prefixes from the routing system and in resetting all the call minutes counts in all the radio channels of the removed operator.
This page is used for setting various parameters of radio channels (RC) and voice processors (DSP) operation.
RC: GAIN to PSTN, RC: GAIN to GSM, DSP: GAIN to PSTN, DSP: GAIN to GSM.
This page is used for executing AT commands in the RC.
This is a maintenance page that is not used under t0he normal gateway operation conditions.
The page is used for sending an AT command to the GSM module and receiving a response from it.
The commands are sent to the GSM module that serves the radio channel with ticked P checkbox (Protocol) on the main page of the web-interface.
This page is used for sending SMS messages and for reading SMS messages from the SIM card.
Sending SMS:
The device has a buffer for each radio channel where the incoming SMS messages are stored automatically (128 records per a channel). The buffer has a ring structure and may be read (message by message) via TCP socket at a specified port according to the in-house network client-server exchange protocol. The protocol itself may be presented to the Customer in electronic format (if the Customer wishes to write his own application for SMS messages reading). This page allows for reading SMS messages from the buffer.
Select the radio channel (Channel) number and click Send, specify the buffer to be called; there are two of them in SMTP mode: main (net) buffer and smtp (smtp) buffer; click Send. Then, in Text window you can view the SMS message read from the radio channel buffer (of course, if the buffer is not empty), and, in Result window, you will view one of the following messages:
This reading mode allows for reading SMS messages “acknowledgments”. If you sent an SMS message with “CDS acknowledgment” flag (SMS Sending page), such acknowledgment will be received by the device as another SMS message and will be automatically placed into the buffer (such acknowledgment cannot be stored on the SIM card).
This page is used for setting time limits (in minutes) (for each of the described GSM operators) for each radio channel in the device and for viewing the current used talk minutes by each radio channel.
For each radio channel you set a time limit (not more than 9999 minutes) for each of the described directions per month.
The gateway records the limits after clicking Send button.
The radio channel operation by limits is enabled on Channel Parameters page using Limit Mode parameter. Used talk minutes are registered by all the described prefixes of each GSM operator according to the rating type selected on GSM Operators page.
Limits are also monitored during the connection and if the limit is reached, connection will be forced to disconnect.
This page is used for setting limits of SMSs sending per day and per month.
The SMS limits are set for each channel per day/month; in addition, the number of already sent SMSs (per day/month) is also displayed here.
Checkboxes to the left from "set" entry fields enable/disable the limited operation, and checkboxes to the right from "sent" entry fields are used for resetting the number of sent messages.
The gateway records the limits after clicking Send button.
This page is used for setting limits of outgoing calls per day with the breakdown by hours.
The number of outgoing calls is set for each channel and for each hour during a day. The maximum possible value is 255 calls per hour. If the limit is set to zero, the number of outgoing calls is unlimited. If the respective current count reaches the set limit, the channel is switched to "Limited Free" status and is not used for any outgoing calls till the end of the current hour of the day.
The gateway records the limits after clicking Send button.
This page is used for setting currency units patterns.
This patterns (character lines) are used when checking SIM card balance.
Maximum number of patterns is 8. The account is searched for the balance figures to the left of the pattern detected. Any fraction of the amount detected is rounded down.
This table lists RCs (GSM channels) in lines and PCM links of PRI digital flow in columns.
The page is used for assigning GSM channels to specific PCM links of PRI digital flow. Here, you can arrange for the call routing from PRI to GSM via GSM channels assigned to PCM links. This mode is enabled from General Settings page: IK_With_RK=1, and disabled by IK_With_RK=0.
With incoming call at GSM network end, the call will be directed to PRI flow using the first free PCM link assigned to this RC. In the absence of free links, the mobile subscriber gets 'canceled'. If an RC was not assigned to any of the PCM links, the call will be sent to PRI flow using the first free PCM link if this PCM link is not assigned to any RC. In the absence of free links, the mobile subscriber gets 'canceled'.
So, the call routing rule in this mode is as follows:
NOTE: 1. Outgoing calls routing by the called subscriber number is NOT available for this mode
(Routing - PSTN->GSM page).
2. Incoming calls from GSM are NOT received in this mode if Only to GSM function is not assigned to the RC (Channel Parameters page)
This table lists radio channel groups in lines and radio channels in columns. The table is used only for SIP mode.
The page is used for joining the radio channels in groups.
Here, you can arrange for routing of the calls from SIP subscribers end to GSM end by indicating the group number before the number of the called mobile subscriber as follows:
*03*0675684032 , where
* are symbols separating the radio channel group numbers,
03 is the number of a radio channel group (two symbols: 01, 02,...,16),
0675684032 is the number of the called mobile subscriber.
Therefore, when this number is called, the unit will search for a free link for calling only those channels from the group numbered 3. Inside the group, the channels are selected according to the call direction (based on the called subscriber number).
One radio channel may be assigned to several or none of the groups (in latter case, no calls will be made to the GSM subscribers end).
NOTE:
If this routing option is used for calls to GSM end, no other routing options will be supported (including calls routing specifying the radio channel number before the number of the called mobile subscriber).
When working with IP subscribers (SIP mode), auxiliary SW, SipAgent (sipua), needs to be launched on the gateway. SipAgent performs the function of a buffer interface between the GSM gateway and the external SIP server, which may be both software, like Asterisk, 3CX, etc., and hardware based such as PUGW CoralFlexicom, Cisco, etc.
SipAgent (sipua), just like the gateway operation SW, is located in /vrem directory; it is launched automatically when switching on the power to the gateway via sipuactl script.
SipAgent start line is: ./sipuactl start
SipAgent stop line is: ./sipuactl stop
SipAgent restart line is: ./sipuactl restart
Sip Agent configuration file – /etc/sipua.conf
Audio codecs configuration file – /etc/sipua/codec.cfg
NOTE: The gateway SW supports the prefix function for daughter card operation in SIP mode: the outgoing call from the SIP subscriber end to the GSM subscriber is directed over the radio channel, which number is specified before the GSM subscriber number and is separated with * symbol, for example: 12*0501230012 where
12 – is the radio channel number the mobile subscriber with 0501230012 number will be called over, * symbol is used for separation between the radio channel number and the number of the called mobile subscriber. So, when placing prefix number with the separator before the called subscriber number, you select the radio channel the outgoing call will be made over; the routing system described on Routing - PSTN->GSM page is ignored in this case.
NOTE:
[general]
deregistering=no ; deregistering at exit
netif=auto; first good interface
netif.in4=auto; first good interface
udp.port=5062is the port sipua operates at (may be different)
[user]
name=4001is the name under which sipua is logged in to SIP server (may be different)
password=quit4001is the password used for logging in to SIP server (may be different)
registrar=sip:192.168.2.77:5060 is the address and port of the SIP server (may be different)
expires=120is the time interval when checking the authorization with SIP server (may be different)
registering=enable is the value specifying whether to log in to SIP server or not (enable – log in, disable – do not log in)
server=udp:192.168.2.77[5060]is the address and port of SIP server (as in 'registrar' line)
local=192.168.2.77the gateway address (may be different)
rtpaddr=192.168.2.77the gateway address (may be different)
While in operation, sipua generates a protocol file /var/log/sipua.log.
The gateway SW supports SIP-call both at SIP and GSM subscribers ends.
At the GSM subscriber end, the address of the called SIP subscriber is set on Routing - GSM->PSTN Page page of web interface:
for example, 192.168.2.12 or 2017@192.168.2.12 , however, this address shall be not longer than 30 symbols.
I. The gateway software supports SMS function with the data for sending being received from an external mail server using POP3 protocol.
The server details are given in /vrem/net.ini configuration file.
COMM_PORT:9005 – not used in this mode;
DATA_PORT:9006 – not used in this mode;
BANK_USER:k16 – user name when used with a SIM Bank;
BANK_PASS: elgato – user password when used with a SIM Bank;
POP3_SERV:pop.gmail.com – mail server address;
POP3_PORT:995 – server prot;
POP3_USER:gsm – user name registered on the mail server;
POP3_PASS: gateway – user password for logging in to the server ;
POP3_CRYP:ssl – encrypting method when establishing the connection with POP3 server.
BRFC_PORT:9100 – base number of TCP port for operation with AT commands.
DRFC_TIME:16 – RFC2217 mode timer.
@
The last line of the file shall contain just one @ symbol.
This is MANDATORY.
To work in this mode with the external mail server, you need to create a mail box for the gateway, i. e. register a new user as specified in /vrem/net.ini file (for example, gsm/gateway).
Server authorization method – open text, encoding – disabled, ssl or tls.
When a mail is sent to a registered user address (hereinafter – the gateway mailbox), the following rules shall be observed:
NOTE:
Other options of mail sending will NOT be served by the gateway.
In this mode, the gateway uses the following operation algorithm:
In case of any problems with this mode, you may enable the option of logging a detailed exchange protocol between the device and the mail server. The protocol is enabled/disabled on the main page by selecting the following commands in Command Select (Выбор команды) menu:
Enable SMS Protocol (Включить протокол SMS сообщений) and Disable SMS Protocol (Выключить протокол SMS сообщений).
The detailed information will be logged to the exchange protocol file.
Messages can be sent even though the channel is currently used for conversation!
II. The gateway software supports received SMSs sending to e-mail addresses using SMTP protocol.
The mail details for every radio channel are set in configuration file /vrem/smtp.ini
#to_addr:from_addr:pwd:smtp_server:port:crypto: ->no,ssl,tls
koe_kto@gmail.com:koe_kto@gmail.com:shadow:smtp.gmail.com:465:ssl:
kto_to@mail.ru:kto_to@mail.ru:none:mail.ru:25:no:
kto_to@mail.ru:kto_to@mail.ru:none:mail.ru:25:no:
kto_to@mail.ru:kto_to@mail.ru:none:mail.ru:25:no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
:::::no:
@
Each line shall end with " : " symbol (semicolumn).
The last line of the file shall contain just one @ symbol.
You need to fill all the 24 info lines of this file (if the line starts with #, it is a remark line, if it start with @ symbol, it is the file end)
It is recommended to edit this file using web interface. Just enter your GSM gateway address from you browser indicating suffix /smtp,
such as: http://192.168.2.77/smtp
The table has 24 lines, each line is for one unit radio channel (the values of the table fields are read from /vrem/smtp.ini).
To addr column – e-mail address the SMSs received over this channel to be sent to.
From addr column – sender's e-mail (actually, this is a login to the mail server e-mails will be sent through).
Pwd column – the sender's password.
Smtp server column – server address e-mail will be sent through.
Port column – port where the mail server "listens" to the client.
Crypto column – defines the presence of absence of encoding when sending mail:
To assign a specific radio channel for SMSs sending through SMTP, flag For SMS only and AutoDel SMS functions for this radio channel on Channel Parameters page!!!
In case of any problems when sending SMS to the addressee e-mail, the table line for this radio channel will be highlighted red indicating the operator intrusion is required. Possible causes of problems when sending e-mails:
If so, the web page screen may look like this:
Here, we can see, there is a problem on the second radio channel when sending e-mail. The problem source will be described in /vrem/smtp_log.txt file like in the following example:
24.10 15:27:36 | rk=02: TCP_Connect to smtp server 94.100.191.205:25 ERROR: Connection refused
24.10 15:27:46 | rk=02: TCP_Connect to smtp server 94.100.191.205:25 ERROR: Connection refused
24.10 3:27:56 PM | rk=02: TCP_Connect to smtp server 94.100.191.205:25 ERROR: Connection refused
24.10 3:28:06 PM | rk=02: FATAL ERROR: WRONG SMTP_SEVER_ADDR. SLEEP 60 sec.
Here, we can see several unsuccessful attempts to establish communication with SMTP server for radio channel no. 2, then the software switches to standby mode (SLEEP 60 sec.).
As soon as you enter correct e-mail details for this radio channel and press SUBMIT key, new details will be read from /vrem/smtp.ini file and a new attempt to establish communication with the mail server will be made. E-mailing log in /vrem/smtp_log.txt file is recorded only for the radio channel with ticked P checkbox (see the main page of the web interface). The log detailing may be selected on the main page of the web interface using Enable Detailed SMTP Protocol Logging (AllSmtpLog) to smtp_log.txt file (Включить запись детального SMTP протокола (AllSmtpLog) в файл smtp_log.txt), Disable Detailed SMTP Protocol Logging (AllSmtpLog) to smtp_log.txt file (Выключить запись детального SMTP протокола (AllSmtpLog) в файл smtp_log.txt). The log file itself (its 'tail' in particular) may be viewed on Statistics – Protocol Files page of the web interface by selecting the file name for SMTP monitoring and pressing SELECT key.
The telephone exchange operation parameters for PRI-30 link are described in two configuration files: /vrem/dss.ini
/vrem/pcm.ini
/vrem/dss.ini file:
MODE_ALERT:PROGRESS; 0-PROGRESS, 1-ALERT
MODE__ACKN:CALL_PROCEED; 0-CALL_PROCEED, 1-SETUP_ACK
MODE_HICOM:OTHER; 1-HICOM, 0-OTHER
MODE_SNDRR:YES; 1-YES, 0-NO
MOD_NORTEL:YES; 1-YES, 0-NO
@
where each line describes one parameter of the channel settings, namely:
MODE_ALERT : either PROGRESS or ALERT message is displayed
MODE__ACKN: the response to SETUP incoming message may be either CALL_PROCEED – Enblock mode (if the number
of the called subscriber is full) or SETUP_ACK – Overlap mode (if the number is incomplete or absent);
the gateway, in its turn, always generates SETUP message with full number B
MODE_HICOM: a specific parameter – the telephone exchange we connect with – HICOM or other
MODE_SNDRR: this parameter specifies if RR message is required in response to each received message
MOD_NORTEL: a specific parameter for Meridian PBX if the PBX is in USER mode (i. e. signaling slave)
The last line of the file shall contain just one @ symbol
This is MANDATORY.
/vrem/pcm.ini file:
SYNC:SLAVE; 0-SLAVE, 1-MASTER
SIDE:USER; 0-USER, 1-NETWORK
RR_M:SLAVE; 0-SLAVE, 1-MASTER
@
where each line describes one parameter of the channel settings, namely:
SYNC : the gateway position in the channel timing: SLAVE or MASTER
SIDE: the gateway position in the channel signaling: USER or NETWORK
RR_M: the gateway position in the channel RR messaging: SLAVE or MASTER
The last line of the file shall contain just one @ symbol.
This is MANDATORY.
RJ-45 connector for PRI-30 link cable connection:
1-2 - transmission from the gateway
4-5 - receiving from the exchange
If PRI-30 physical link is not raised or you have doubts as for the correct cable connection, use an ordinary LED to see where the unit transmission is.
The gateway transmission shall be closed on exchange receiving and vice versa.
For the gateway, a simplified billing system is developed to allow for accounting of the talks by directions and billing plans.
http://192.168.0.77/db
Billing system is built based on sqlite3 data base and PHP scripts that provide the data base access and management. The PHP version on the gateway shall be not less than 5.3.2. With older PHP version, the billing system operation is IMPOSSIBLE. To accelerate the system operation, the data base (hereinafter – the DB) is created on an electronic disk (/disk directory). There is the possibility of creating DB copies in the file system located on a USB-disk (/www/root/db/db_backup/ directory). All working PHP scrips are located in /www/root/db/ directory. With correct gateway restarting, the DB is stored automatically, and with the system launch, it is automatically copied to RAMDisk (/disk directory). The RAMDisk size is limited to 8 MB. So, the size of the created DB is restricted (for example, DB containing 18,000 records is sized a bit more than 2 MB).
The DB is created by the user (i. e. by you) using log file generated by the gateway software.
The data to the log file are collected during a month and the file is stored in /vrem directory. Usually, it is named stat_log.txt. The user (i. e. you) must create the DB from the log file using a respective billing plan menu option – Create DB (Создать БД). Then, you can retrieve the calls.
The Main menu is located in the left part of the window and consists of the following options:
This is the starting page of the complex where the telephone talks are retrieved by a group of criteria:
A combination of the criteria is also possible. There is also the possibility to write the selected data to file (html file).
This option allows for viewing, creating and editing the groups of subscribers.
A new group of subscribers may be created or any existing group may be edited.
When selecting one of the groups, you get access to its subscribers, i. e. you can add a subscriber to the group or remove a subscriber from the group.
Clicking Remove Record (Удалить запись) button when editing the group results in the whole group removal. If you want to remove one or several subscribers, tick their checkboxes and click Refresh Record (Обновить запись). When adding a new subscriber to the group, enter his/her number into one of 20 entry fields and click Refresh Record (Обновить запись).
I believe, the meaning of Exit Without Changes (Закрыть окно без обновления) function is clear without comments – clicking it returns you to the previous menu page.
Creating subscriber groups helps to detail outgoing calls by a company or an organization division thus giving a useful tool for budgeting this unit communication costs.
This option is used for creating, viewing and editing the directions for accessing the mobile communication operators' networks.
Each direction is described by its name, dial code, external radio channel for outgoing calls, billing plan and the number of digits in the dialed number. The direction of each call is stored in the data base and is always specified when generating a detailed outgoing calls report.
The operator (i. e. you) creates, edits or removes the directions at his/her own discretion.
Items:
New directions may be created or any existing direction may be edited.
When creating a new direction, it is necessary to specify:
The information of a new direction is stored in the DB when Refresh Record (Обновить запись) button is clicked.
When editing an existing direction, the following field may be changed:
The changes are stored when Refresh Record (Обновить запись) button is clicked. Exit Without Changes (Закрыть окно без обновления) function allows to exit, storing the old data in the data base. Remove Record (Удалить запись) button removes the current direction from the date base.
This menu option allows for creating, editing, and removing billing plans. The billing plans are created taking into account the charged time, day period (business time - 8:00 a.m to 8:00 p.m, except for Saturday, Sunday and holidays) and the cost of connection.
To create a new billing plan, click Add New Billing Plan (Ввести новый тариф) link.
Having filled in the respective fields and clicked Refresh Record (Обновить запись) button, you store the data into the data base. Exit Without Changes Закрыть окно без обновления function allows to exit, storing the old billing plan data in the data base.
Remove Record (Удалить запись) button removes the current billing plan from the data base.
When editing the billing plan, make changes in the required field and click Refresh Record (Обновить запись) button. Remove Record (Удалить запись) button removes the current billing plan from the data base. Exit Without Changes Закрыть окно без обновления function allows to exit, storing the old billing plan data in the data base.
Create DB Option
The name of the file, such as st_stat_log.txt is the identification of the script that forms RAMDisk data base from the file. After a script is executed, the display is like this:
The name of thus created DB will be constructed of the log file name + extension .s3db
As you can see from the screenshot, the process of a data base creation from a log file may take quite a long time. For log files with many records, up to 50,000, this may take several minutes. However, after the DB is created from a monthly log file, then you can make retrieves as many times as necessary, and make a backup copy of the DB.
Remember, the DB is created on a limited size RAMDisk (8М). Average DB size will be approximately equal to the size of the log file the DB is created from.
Here you can view the content of backup directory (/www/root/db/db_backup), i. e. the data bases that were stored in Select DB menu option.
The DB name, such as stat_log.txt.s3db is the identification of the script that copies the DB to the work directory (/disk) of RAMDisk. Therefore, you can both store a DB on a permanent carrier and copy it from a permanent carrier into the work directory (remember, a billing plan may be operated only with the DB located in /disk directory on RAMDisk).
This option is used for describing holidays and is necessary for correct calculation of the costs of calls (these days are charged as non-business time).
The window for description of a holiday looks like this:
Here, filed Name (Название) (any text), Day (День) (1 to 31) and Month (Месяц) (1 to 12) are mandatory. When clicking Refresh Record (Обновить запись) button, this record data are stored in the DB. When clicking Delete Record (Удалить запись) button, the record is removed from the DB.
Thus, the billing system described, as you can see, has limited capabilities, however, in most cases, it is quite sufficient for getting detailed data of communication costs provided the traffic is not too big (not more than 150,000 calls per month).
This page is used for on-line watching the recording of various device operation protocols (the same as tail -f command in Linux system).
The following protocols may be watched:
"ring" – call at the mobile subscriber end,
"sms" – SMS arrival from a mobile subscriber,
These events will be used for /vrem/event.sh script that can send a message for example to a third party http server using curl utility. Such message may contain the following parameters:
The content of /vrem/event.sh script may be adapted for specific Customer's requirements, such as via web-interface, System menu option, event tab.
This page is used for viewing some statistical parameters that indirectly characterize the connection quality with outgoing calls from PSTN to GSM such as:
The table shows the ACD and ASR parameters for each radio channel in all the directions for the current month.
Two types of diagrams are possible:
The type of the diagram is selected using Diagram control element or ACD&ASR or Calls parameters. A diagram may be plotted for all the channels (by X axis – channel numbers 1 to 24) or just for a single channel (by X axis – days of month); you can specify a direction for plotting or get a diagram for all the directions (without selecting any specific direction) or specify a month when plotting.
All the data for plotting are stored in sqlite3 format data base (one month – on file) on electronic disk /disk. With correct operation finishing, these files are stored on USB disk and are restored on the electronic disk from the USB disk after the device is switched on. There is also a possibility to store a diagram in a file or print it.
The unit supports SMS sending and receiving mode using SMPP protocol (version 3.4) over several connection at a time using the same user name and password (login/password – this pair is one and the same for all the established connections, with the maximum simultaneously served connections being equal to the number of existing GSM channels in the device) – only for v. 13.2.8 and higher software.
The following is provided:
A single client servicing is supported per a channel.
SMPP clients were tested:
The parameters for this mode of operation are described in /vrem/smpp.ini configuration file.
The fine content is:
SMPP_PORT:9008
SMPP_SERV:0.0.0.0
SMPP_USER:k16
SMPP_PASS:elgato
@
where
SMPP_PORT:9008 is the number of the port, at which the unit "listens" to SMPP client
SMPP_SERV:0.0.0.0 is the client's IP address (0.0.0.0 inquiry from any address is received,
if any specific address is indicated, such as 192.168.2.36, request for connection will be received only from this address).
The parameters values in the configuration file may be changed (but for the parameter names), however, to save the changes, the working SW shall be restarted.
SMPP_USER:k16 is user login (may be changed).
SMPP_PASS:elgato is user password (may be changed).
The last line of the configuration files shall consist of just one symbol: @.
Connection protocol for SMPP client(s) and the unit is recorded in /vrem/smpp_log.txt file and SMPP messages exchange protocols for each connection are recorded in separate files: /vrem/smpp_nconXX.txt where
XX is the ordinal number of the connection – 01 to 24.
SW version 14.7.6 and higher allows group operation, i. e. the radio channels are joined in groups (up to 24 groups) each having its own attributes. The group attributes are described in /vrem/.sms_group.ini configuration file which fields may be edited from the web-interface main page using SMPP – SMPP Groups menu option:
This mode allows for the device to choose a radio channel for SMS sending only from those assigned to the given group.
A specific radio channel for SMS sending inside the group is selected:
If Wait DLR (Ждать DLR) checkbox is not ticked for the radio channel, the channel is set to "ready for SMS sending" state as soon as Channel Busy Time (Время занятости канала) timer is over (this timer starts the countdown by SUMBIT_SM command from SMPP client).
If Wait DLR (Ждать DLR) checkbox is ticked for the radio channel, the channel is set to "ready for SMS sending" state when two conditions are met:
P.S.
The expected DLR codes are specified as decimal notation separated with commas (not more than 32 codes).
If any of the timers is set to zero, the timer is ignored and does not influence the radio channel transition to "ready for SMS sending" state.
Group mode of operation is set when launching the program with additional key words (if necessary these key words may be requested from the device developer).
When the devices is operated in SIP/DSS (PSTN) mode, CLIP function is supported.
What is the CLIP function?
When a GSM subscriber is called (successfully or not) from PSTN end, the following call parameters are stored in the database (a separate database for each channel):
When the call is received from GSM subscriber end, the database is searched for the entry with the calling GSM subscriber number. If such number is found, the call is directed to the PSTN subscriber number according to the entry then the entry is removed from the database. If no GSM subscriber number is found, routing will be applied as described on Routing GSM->PSTN page.
P.S.
INFORMATION MENU OF GSM OPERATORS' BASE STATIONS
The unit SW allows for displaying "the GSM operators' base stations layout at the installation site" and to "control" this layout manually or automatically.
Usually, any mobile phone is registered in the mobile communication operator network at one of the base stations with the best signal level, at the same time, there is a possibility to view six more base stations of the same operator in the vicinity, that is base stations available for registration.
INFORMATION MENU OF GSM OPERATORS
The unit SW allows for displaying the GSM operators layout at the site of installation, and, specifically, to display the information about the GSM network the radio channel is registered in (according to SIM card data), the available networks of other GSM operators and unavailable GSM operator networks. Available networks usually allow for registering roaming subscribers.
ADMINISTRATOR MENU
This page is used for modifying some system scripts and for rebooting both work SW of the device (elgato) and the whole system.
On this page, you can edit some important scripts (scenario and configuration files) of the system, therefore, be careful, if in doubt, make no changes.
On this page, the access to the following scripts (files) is possible:
MODE=sip
or
MODE=dss
If you changed one of the files in the edit field, tick the checkbox under the edit window and click Set. If you also tick With Restart System checkbox, the system will be restarted after the file is stored. If you tick With Restart Software checkbox, the operational SW will be restarted after the file is stored.
Please, remember, any changes you make on this page are made at your own risk, if you are not skilled in UNIX system communication, consult the equipment supplier.
The gateway service software consist of:
1. thttpd – web-server with PHP-5.3.2 support, used for the gateway working parameters setting.
2. sshd – server for authorized users access to the gateway using SSH2, SFTP protocol.
3. mc – file manager.
4. iptables – firewall closing the access to the gateway by a number of criteria.
5. tcpdump – utility for the IP packages writing to the file for their further analysis.
6. asterisk – softswitch version 1.6.2.24.
7. ethtool – utility for LAN interface parameters editing.
8. sqlite3 – utility for creating and servicing the databases.
9. ntpd – time synchronization server via Internet.
10. usb_modeswitch – utility for USB GSM modems utilization (such as huawei ec122).
11. autossh and sshpass – network utilities of ssh protocol.
12. minicom – terminal software for operation with rs232 ports.
13. pppd – daemon of PPP (Point-to-Point Protocol) used for controlling network communication between two units in UNIX like operational systems.
14. pptp – tunnel point-to-point type protocol allowing for safe connection with the server by creating special tunnel in standard unprotected network.
15. openvpn – software product utilizing Virtual Private Network (VPN) technology to create encoded point-to-point type or server-client type links between the computers. Allows for establishing the communication between the computers behind the NAT-firewall without the necessity to change the settings.
16. ipsec – the software product utilizing a set of protocols to protect the data transferred over the IP internetwork protocol allows for the authenticity and/or encoding confirmation of IP packages; it also includes protocols for safe communication of keys and VPN connection establishment.
17. reiserfsck – utility for servicing disk sections (integrity check) of logged reiserFS file system. Remember, this utility may be used only for uninstalled section with reiserFS file system.
18. kannel – this is a compact and very powerful WAP and SMS open code software complex (http://www.kannel.org). It is widely used in the world for organization of short messages (SMS) sending/ receiving in the mobile communication networks. Kannel complex is not included into the base package of G20 file system and is installed as an option from RPM package.
19. curl – service utility for the command line allowing for interaction between multiple different servers over various URL syntax protocols. The utility version is 7.33.0, protocols: dict file ftp ftps http https pop3 pop3s smtp smtps telnet, functions: Largefile NTLM NTLM_WB SSL libz.
20. net-snmp package is an SW set for deployment and utilization of SNMP protocol (v1, v2c and v3). The package version is 5.7.2.
21. smsd – network client (32 channels) for sending/receiving SMSs using RFC2217 protocol.
Go to the main page in your web interface, tick checkbox B (Б) of the required channel and click the RC button. (the channel switches to phase 1 – LOCKED)
Go to the main page in your web interface, remove a tick in checkbox B (Б) of the required channel and click the RC button. (the channel switches to phase 8 – FREE)
Go to Routing - GSM->PSTN page in your web interface and assign the numbers of the called subscribers opposite to each channel (SIP/PRI network subscribers who will be used for routing the incoming calls). Click Send.
Go to Routing - PSTN->GSM page in your web interface, for each direction prefix, tick the checkboxes of those channels that will be available for calls to the given direction. Click Send. XXX direction prefix is used for routing any calls to the channels with respectively ticked checkboxes.
The signal from the mobile network operator's base station is weak. More likely, there is no aerial on this channel or the aerial cable connection to the connector is loose. You need to block the channel, check the aerial connection and then unblock the channel.
Go to the main page in your web interface and tick P (П) checkbox of the required channel and click this channel button; thus you select the number of the channel the echo canceler will the enabled for. Then, in command menu on the same page:
The changes will be effected when making a new connection via this channel.
In /vrem/vin_log.txt file, the respective messages of these events will be displayed.
The unit has the possibility of changing the voice gain factors both on the radio module itself and on the voice processor channel (DSP) that is assigned to this radio module.
Go to the main page in your web interface and view the current voice gain factors for the channel (GAIN PRI/GSM column); they may be like these: 75.5/60.60 where 75 is the voice gain factor (in dimensionless unit, 0 to 99 for SIM300, SIM900, M10 modules and 0 to 4 for SIM5215 modules) on the radio module to SIP/PRI subscribers end, 5 is the voice gain factor (0 to 15 for SIM300, SIM900, M10 modules and 0 to 7 for SIM5215 modules) on the radio module to the GSM subscribers end; 60 is the voice gain factor on the DSP processor channel (place a mouse cursor on this figure to view the actual value in decibels) to SIP/PRI subscribers end, the next 60 is the voice gain factor on the DSP processor channel (place a mouse cursor on this figure to view the actual value in decibels) to the GSM subscribers end.
You can change these values on F8 page in your web interface.
To change the gain factors on the radio module, enter required values to RC: GAIN to PSTN and RC: GAIN to GSM fields of the respective channel and tick Read GAIN and Write GAIN checkboxes for this channel; click Send. Refresh the main page in your web interface and you will be able to view the changes you have made. You can change the gain factors only provided the channel has FREE (СВОБОДЕН) status, if the channel is busy, the changes will be stored and applied to the channel after it becomes free. For this, channel LOCKING or UNLOCKING is not required.
To change the gain factors on a DSP channel, select the required values in DSP: GAIN to PSTN and DSP: GAIN to GSM fields of the respective channel and click Send. Refresh the main page in your web interface and you will be able to view the changes you have made. New factors are stored immediately and used with the first call over this channel.
On Channel Parameters page in your web interface, tick Read RSSI checkbox of the required channel and click Send. On the main page in your web interface you will be able to view the changed signal level value in RSSI dBm column. You can make changes only provided the channel has FREE (СВОБОДЕН) status, if the channel is busy, the command will be stored and applied to the channel after it becomes free!!! For this, channel LOCKING or UNLOCKING is not required. The gateway software automatically checks the signal level from the mobile operator base station once every 120 seconds (approximately); newly measured value is immediately displayed on the main page of the web-interface in RSSI column.
On Channel Parameters page in your web interface, tick CallBack checkbox of the required channel and click Send. This operation sets the channel for receiving the incoming calls from the mobile subscribers in CallBack mode.
For the rest of the channels, tick checkboxes Conf+CBack and Enable DTMF and click Send. The channel won't be used for outgoing (to GSM) calls if this is forbidden on the main page of the web-interface. For additional information about CallBack function, see Data Base page. CallBack function is an option and require authorization.
using the console (only for "root" user), execute: /vrem/gsmctrl restart
using the administrator web page (http://192.168.2.77/lan), tick With Restart Software checkbox and click Set. (Restarting takes about 16 seconds.)