Question 1

2N VoiceBlue Next GSM gateway - How to configure it with 2N IP intercoms?

Accepted Answer

The steps in this article require using the web interface of a 2N device. Please follow this manual in case this is your first time using the web interface. This guide describes basic steps to configure Peer to Peer communication between 2N IP intercoms and 2N VoiceBlue Next GSM/UMTS gateway to be able to make calls intercom  GSM network. All names and IP addresses are only for this example, please change this information according to your names and IP plan.  Wiring  Connect 2N IP intercom to an IP network, 2N VoiceBlue Next has to be connected to the same IP network.The values in our example:2N IP intercom - 192.168.50.1422N® VoiceBlue Next - 192.168.50.66Mobile phone number - 734123456    2N IP intercom settings  You need to specify IP address and subnet mask for your 2N IP intercom.    Select a contact which is mapped for one touch call (e.g. for button #1 on your 2N IP intercom). You need to modify Phone number for this contact according to the desired phone number you would like to call through the GSM gateway (sip:mobile_number@IP_address_of_VBN). In the following picture there is 192.168.50.66 an IP address of 2N® VoiceBlue Next and 734123456 is the mobile phone number, so you set: sip:734123456@192.168.50.66.    Set up SIP settings according to the next picture. IP addresses should match to the IP of 2N® VoiceBlue Next.    Allow the intercom to automaticaly accept all Incoming calls.     Settings of 2N® VoiceBlue Next gateway   You need to specify IP address and subnet mask for your 2N VoiceBlue Next in the menu Ethernet configuration.    In the VoIP parameters check option Send 200 OK on REGISTER request to allow "virtually" register 2N IP intercom to 2N® VoiceBlue Next.    The gateway is set to process all outgoing call requests to the GSM network in default, but you also need to define the IP address of the 2N IP intercom in the SIP proxy (GSM->IP): setting in order to process the incoming calls from GSM to the intercom.    In the GSM incoming groups set mode Accept incoming calls + dialtone and DTMF dialing timeout[s]:to 0.     And also set the phone number of the intercom (111) in the List of called numbers. Now you can easily call your mobile phone by pressing the dial button on the intercom and also reach the intercom directly from the mobile phone by calling the SIM card in the gateway. Please note that 2N® VoiceBlue Next gateway can handle voice calls only, there is no support for the video stream in this scenario even if your 2N IP intercom intercom is equipped with a camera.

Question 2

2N VoiceBlue Next GSM gateway - How to configure it with 2N IP intercoms?

Accepted Answer

The steps in this article require using the web interface of a 2N device. Please follow this manual in case this is your first time using the web interface. This guide describes basic steps to configure Peer to Peer communication between 2N IP intercoms and 2N VoiceBlue Next GSM/UMTS gateway to be able to make calls intercom  GSM network. All names and IP addresses are only for this example, please change this information according to your names and IP plan.  Wiring  Connect 2N IP intercom to an IP network, 2N VoiceBlue Next has to be connected to the same IP network.The values in our example:2N IP intercom - 192.168.50.1422N VoiceBlue Next - 192.168.50.66Mobile phone number - 734123456    2N IP intercom settings  You need to specify IP address and subnet mask for your 2N IP intercom.    Select a contact which is mapped for one touch call (e.g. for button #1 on your 2N IP intercom). You need to modify Phone number for this contact according to the desired phone number you would like to call through the GSM gateway (sip:mobile_number@IP_address_of_VBN). In the following picture there is 192.168.50.66 an IP address of 2N VoiceBlue Next and 734123456 is the mobile phone number, so you set: sip:734123456@192.168.50.66.    Set up SIP settings according to the next picture. IP addresses should match to the IP of 2N VoiceBlue Next.    Allow the intercom to automaticaly accept all Incoming calls.     Settings of 2N VoiceBlue Next gateway   You need to specify IP address and subnet mask for your 2N VoiceBlue Next in the menu Ethernet configuration.    In the VoIP parameters check option Send 200 OK on REGISTER request to allow "virtually" register 2N IP intercom to 2N VoiceBlue Next.    The gateway is set to process all outgoing call requests to the GSM network in default, but you also need to define the IP address of the 2N IP intercom in the SIP proxy (GSM->IP): setting in order to process the incoming calls from GSM to the intercom.    In the GSM incoming groups set mode Accept incoming calls + dialtone and DTMF dialing timeout[s]:to 0.     And also set the phone number of the intercom (111) in the List of called numbers. Now you can easily call your mobile phone by pressing the dial button on the intercom and also reach the intercom directly from the mobile phone by calling the SIM card in the gateway. Please note that 2N VoiceBlue Next gateway can handle voice calls only, there is no support for the video stream in this scenario even if your 2N IP intercom intercom is equipped with a camera.

Question 3

2N VoiceBlue Next - How to configure it with Auerswald PBX?

Accepted Answer

Following FAQ page describes how a 2N VoiceBlue Next can be connected to Auerswald PBX using SIP trunk.  Parameters of 2N VoiceBlue Next:  Example IP address 192.168.1.2  Incoming port: 5080  Firmware: 1.25.0.28.4   Auerswald equipment parameters:  COMpact VoIP 5010, COMpact VoIP 5020, COMmander Basic.2 and COMmander Basic  Example IP address: 192.168.1.3  version: 4.2F Build 000     How to set 2N VoiceBlue Next? After having logged on to the 2N VoiceBlue Next, navigate to “Gateway configuration” on the menu tree in the left column, then choose “VoIP parameters”. Set following parameters:  Use CLIP from INVITE field: to value "From"  Use DIAL from INVITE field: to value “To”  Send 200 OK on REGISTER request: to enabled  Replace CLIP from GSM with Caller ID: to enabled  Above is important because otherwise the Auerswald PBX won’t register to the 2N VoiceBlue Next or incoming GSM calls won’t be displayed correctly on the phones’ displays. Please see image below:  Next, please scroll down in “VoIP parameters” until and set “Codec priority” . Please make sure here that G711a(8) is chosen as priority 1, then G711u(0) followed by G729(18). Further down under “IP addresses” fill in connection details to your Auerwald PBX. Enter the IP address of the PBX in the following fields (lets assume IP address of Auerswald PBX is for example 192.168.1.3):  SIP proxy (IP->GSM)  SIP proxy (GSM->IP)  Note: You can also set the SIP port. Remember that Auerswald already uses port 5060 for internal calls so you need to choose different one, for example 5080 w To make sure that no other equipment can access the 2N® VoiceBlue Next check the box “no reply on other IP addresses’ requests:”. Leave the rest unmodified. Click on the save button (button right, the icon with the paper sheet and the pen).      How to set Auerswald PBX?  Login as administrator to your Auerswald PBX (regular username is “admin“).  Then please find the sub-menu “Voice over IP (VoIP)” in the menu “COMset” -> “External numbers”.  Click on “Provider”. In an empty field of the column named “Provider name” on the right half of the screen enter a name for your 2N GSM gateway, e.g. “2N”. Then click on the disk symbol on the left side to save that entry.  After this click on the “Configure” button in the right column. Your screen should look like this:    Once you clicked on “Configure” you will see following page:    Please fill out the fields “Domain”, “Outbound proxy”, “SIP port (1...65535)”, and “Codec settings” as illustrated in the screenshot above.  In the field “Domain” you enter the IP address of your 2N VoiceBlue Next GSM gateway. The factory settings are “192.168.1.2” but if you changed it in the configuration page of the 2N VoiceBlue Next, make sure to enter the correct IP address here accordingly. In the field “Outbound proxy” you choose “manually” and enter the same IP address as in the “Domain” field, it’s the IP address of the 2N VoiceBlue Next in your network. In the field “Port” you enter the SIP port number that you have entered / will enter in the configuration pages of the 2N VoiceBlue Next. Usually that is 5060, but please keep in mind that the Auerswald PBXs use this SIP port exclusively for their extension lines (the internal parties) and that there was no option in former firmware versions to change it. Therefore, in this example, we changed the SIP port to 5080. You can take whatever you like, but make sure to have the same port open on your 2N VoiceBlue Next and that it is different than the port used for the internal calls (PBX’s extensions). “Codec settings”: Make sure only to choose “G.711” as priority number 1 and to leave the other entries blank. Otherwise you will receive a SIP response 415 (Unsupported Media Type) when you make outgoing GSM calls from your PBX via your 2N VoiceBlue Next.  Go to the field “Format of called telephone number” and set it to “send as dialled”. “Format of own telephone number” should be set to “Only MSN or main number and DDI”.“Type of number presentation” should be set to “In the display text”. Choose “Anonymous” for “Method of number presentation suppression”.Check the box named “Trust transferred telephone number” and uncheck the one named “Internationalise unknown telephone numbers”.  After having set everything click on “Action Accept” to save your settings.   Click on “Accounts” on the menu tree on the left side. It’s right below “Provider”. You will see similar page to following picture:     Make sure that you choose “PTMP connection” (point to multi-point) for the provider “2N”, that means for your 2N VoiceBlue Next and give the account a name by entering this in the field “Account name”, e.g. “GSM Gateway”. Save this setting by clicking on the disc symbol in the left column.  Click on “Configure”. You will see a page similar to picture below:      Enter MSNs (multiple subscriber numbers) that will be used by the 2N VoiceBlue Next to assign calls to certain extensions lines in your PBX.You can enter whatever you like here. In this case we chose “1111” to “4444” for the 4 SIM card modules of the 2N VoiceBlue Next with 1111 being the number called by 2N VoiceBlue Next if there is an incoming GSM call on the first SIM module etc.  Give a name to the MSNs to identify them easily, e.g. “GSM GW (SIM 1)” to “GSM GW (SIM 4)”. Leave “Ringer rhythm” on “1 x long”.It is very important to set the fields above with country code and area code to make sure that outgoing calls are dialled correctly and incoming calls are displayed correctly.So enter your country code the way it is dialled in your country in the field “Country prefix” (in this case we have Germany’s country code “0049”) and in the field “Area code” enter your area code numberthe way you would usually dial it (in this example the number is “05306”, the area code of the city of Cremlingen in Germany).You will also need to enter a number to identify this VoIP line within your PBX. This is an internal number used to choose this VoIP line specifically and make external calls with it. Enter whatever you like here that fits the internal numbering scheme of your PBX.In our example the number is 1501.  The fields “User name” and “Password” are mandatory. You have to enter something here. It is not important what you enter as the 2N VoiceBlue Next will accept any user name and password and acknowledge it with a “200 OK” on registration.  Make sure you click on “Action -> Accept -> Execute” to save your entries!  Next you click on “Call distribution” in the menu tree on the left. It’s below “Accounts” and “Telephone numbers”Here you will find your MSNs that you entered before to identify the SIM modules 1 to 4 of 2N VoiceBlue Next (in our case that was 1111, 2222, 3333, and 4444). HHere you can assign extension numbers, internal numbers, team numbers etc. to these MSNs so that certain internal phones will ring once there is an incoming GSM call from 2N VoiceBlue Next.After having assigned your internal numbers to the MSNs make sure you click on “Action -> Accept –> Execute” to save your setting.      Choose “Routing” and then “VoIP and GSM Routing” from the menu tree in the left column.This is where you can set the PBX’s least cost routing functionality.Please enter the code numbers of the cell phone networks in your country (in this case we see the first three digits of the code numbers of the German cellular network operator, 015, 016, and 017) and choose the VoIP trunk line that we set up earlier named “GSM Gateway”.In case the 2N VoiceBlue Next should not be reachable due to any reason you can set fallback lines here, too. In this case ISDN lines have been chosen as fallbacks under the columns named “2nd dialling” and “3rd dialling”.      Finally make sure that the IP address of 2N VoiceBlue Next is listed in the white list of the PBX.Otherwise, if the registration fail, the PBX might add 2N VoiceBlue Next to the blacklist and then there will be no more connection to 2N VoiceBlue Next possible.      Please also make sure that the internal parties are allowed to access the VoIP trunk to the 2N VoiceBlue Next.You can set this feature up in the menu tree on the left: COMset -> Internal numbers -> Subscriber (scr) -> Properties -> Exchange line settings + subscriber selection in the list field at the top.

Question 4

2N VoiceBlue Next - How to set? (version 1.19 and lower)

Accepted Answer

2N VoiceBlue Next is designed to reduce the costs of outgoing calls to GSM networks. The main advantage is the ability to expand the IP PBX to include a GSM gateway and make use of a LCR which selects the best GSM operator depending on the called number.   How to connect 2N VoiceBlue Next  Connect the 2N VoiceBlue Next via Ethernet cable directly to the computer. Change the IP address setting in the configuration of your PC (e.g. 192.168.1.33 and Subnet Mask 255.255.255.0)  Open your web browser and enter the default IP address of GSM gateway 192.168.1.2 with the default credentials (Admin/2n). If web page does not appear, make a factory reset of gateway with RESET button on front panel. Press it for 10 seconds. After this time gateway restarts and you can reload the page.  In the gateway configuration and Ethernet configuration you are able to change the IP address.    How to set up connection to the opposite device (PBX)  For the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for secure communication just with traffic from your PBX. You can specify the IP address and port which will accept SIP packets from.  In case you leave there 0.0.0.0 in SIP proxy(IP->GSM) gateway will be open for traffic from all IPs. !!! This can lead to misuse of the equipment by an unauthorized person !!!!    How to configure LCR table Situation...  Let's say we have SIM cards of two GSM operators:  T-Mobile: Using the following prefixes (602, 606, 607, 723, 724) and it requires that you dial the number from your mobile phone with the international prefix (+420). All numbers have a nine digit length with the operator prefix but without the international prefix.  Vodafone: Using the following prefixes (901, 902, 907, 909) and requires you to dial the number with the prefix (+420). All numbers have a nine digit length with the operator prefix and without "+420".   First we have to assign modules to two GSM outgoing groups (for each operator).The first step which we have to do is to place the SIM cards into the SIM holders. We will start with modules 0 using the T-mobile operator SIM. Module 2 will follow the same procedure but using the Vodafone SIM.   Configuration of GSM outgoing groupsYou are able to set up different setting for each GSM group (CLIR, free minutes, Virtual ring tone, roaming and others).  Prefix listsWe have to create two network lists, the first one for T-mobile and the second one for Vodafone.  T-mobile network list:We configure the normalization of Called party number in the Table of replaced prefixes (the number in front of the slash mark is replaced by number behind the slash mark. If there is not any number in front of the slash mark, it equals to "everything").We also have to fill in the Table of prefixes with all prefixes of the T-mobile operator. Because all numbers are 9 digits length, it is not necessary to specify it for each prefix, we can use the parameter Default count of digits and fill there the value 9.For the match in prefix list, the Table of replaced prefixes, Table of accepted prefixes and Digits count must correspond with the called number.  Vodafone network list:We configure the normalization of Called party number in the Table of replaced prefixes (the number in front of the slash mark is replaced by number behind the slash mark. If there is not any number in front of the slash mark, it equals to "everything"). We also have to fill in the Table of prefixes with all prefixes of the Vodafone operator. Because all numbers are 9 digits length, it is not necessary to specify it for each prefix, we can use the parameter Default count of digits and fill there the value 9.   LCR TableWe have to configure rows in the LCR table where we bind together GSM Outgoing groups with Prefix lists. Click on ADD button (or edit the first line of LCR table) to add the first line and configure it the following way for the T-mobile: Click on ADD button to add the second line and configure it for the Vodafone.Now you can see two LCR lines in the LCR table:    Incoming calls  For incoming calls you can define 2 groups with the different behavior and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.  In GSM incoming groups you can define the behavior for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.  With mode Accept..., you can define the list of called numbers which will be automatically dialed after DTMF dialing timeout in case the customer does not press any button within the specified time. From the configuration, you can see 5 seconds for DTMF dialing and after that the call will be routed to the extension 100 in your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters). If you want immediate connection set DTMF dialing timeout to 0.    SIM cards  In the Gateway control and module control you can see the status of the SIMs. The recommendation for quality of the signal is between -65 and -85dBm.  The logged SIM card is indicated by switched off the led diode on the 2N VoiceBlue Next

Question 5

3CX - How to interconnect with 3CX?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.50.51  Incoming port: 5060  Firmware version: 01.00.03rc3   3CX PBX parameters:  IP address 192.168.50.115  Incoming port: 5060  Software version: 9.0    2N Voice Blue Next settings  SIP trunk interconnection For the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed to secure communication with traffic from your CCM only. You can specify the IP address and port from which SIP packets will be accepted. In case you leave there 0.0.0.0 it will be open for all traffic.   Configuration of the LCR (Least Cost Routing) The GSM operator has e.g. in our country prefix 6 and 7 with nine digits in a number. The setting is below.     You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.   3. Configuration of GSM outgoing groups You are able to set up different settings for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others)  4. Incoming calls For incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.  In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.  You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button within the specified time. From the configuration you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 100 to your 3CX PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).     3CX PBX SETTING  You can download the 3CX PBX for free. Then you can activate a free version by registering the product and using a demo license key (Settings → Activate License) and the 3CX system will be changed to Commercial Edition (3CXPSDEMO) where all features are available. Demo license is not limited by time.    You can check differences between Free version and Commercial Edition in this document on page 11.   Create an Extension Use the button Add Extension and fill up Extension Number and ID and Password.    Create a new SIP trunk Add → VOIP Provider in the menu of the 3CX PBX. Use the name of provider and choose Generic SIP Trunk. Then use the button NEXT.   Fill up the IP address and the listening port of the 2N® VoiceBlue Next.  Fill up the External number (this number will be identification of call - FROM and CONTACT field)  Set up the the extension where incoming call will be routed from the 2N VoiceBlue Next.     Set up Outgoing rules  Prefix called from the extension  Set up the range of extension which can use this trunk (e.g. 111-120)  SIP trunk and Strip Digits (0 will be striped in this case)  Set up codecs and turn off registration of the SIP trunk. This setting is in VOIP Providers → Advanced  Create inbound rules specify numbers or range numbers that could be dialled from 2N® VoiceBlue Next. You can use "*" for all numbers. specify the SIP trunk from which the number will come specify the extension, ring group or voice mail where the call will be connected    Make a call You can register your SIP phone or download the 3CXPhone from 3CX web page for free: http://www.3cx.com/VOIP/voip-phone.html Register your SIP phone to the 3CX PBX and make an outgoing call with specified prefix to GSM via 2N VoiceBlue Next.

Question 6

Alcatel-Lucent OXE (OmniPCX Enterprise) R12: How to interconnect with 2N VOICEBLUE NEXT

Accepted Answer

2N VoiceBlue Next firmware Version 1.37.3.40.3    Alcatel-Lucent OXE (OmniPCX Enterprise) R12    Functionality  Incoming and outgoing call, opening the door  2N VoiceBlue Next Configuration  Please set all usual Network settings according to usual practice. Do not use Authorization ID nor fill out any password. For both options described further is the setting of 2N VoiceBlu Next the same.     2N VoiceBlue Next has these parameters:  IP address 192.168.92.200  Incoming port: 5060  Firmware version: 1.37.3.40.3   Alcatel OXE parameters:  IP address 192.168.92.246  Incoming port: 5060  Version 12.2   Alcatel OXE configuration - Option - SIP  Create Trunk group, Type T2 - MINI SIP or SIP      Go to SIP Gateway settings     SIP Gateway - fill out number of SIP Trunk Group    After this please restart the Alcatel OXE.  Make sure that trunk channels are free.     SIP Proxy - set minimal authentication method to SIP     2N VoiceBlue Next configuration  1. SIP trunk interconnection For the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed to secure communication with traffic from your PBX only. You can specify the IP address and port from which SIP packets will be accepted. In case you leave there 0.0.0.0 it will be open for all traffic.     2. Configuration of the LCR (Least Cost Routing) The GSM operator has e.g. in our country prefix 6 and 7 with nine digits in a number. The setting is below.   You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.       3. Configuration of GSM outgoing groups You are able to set up different settings for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others). In case you don´t have a Ring back tone, set up Delay for ALERTING to option 4.      4. Incoming calls For incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.   In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.   You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button till the specified time. From the configuration, you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 101 to your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).

Question 7

Alcatel OXO - How to interconnect with Alcatel OXO 7?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.92.200  Incoming port: 5060  Firmware version: 01.00.03i10   Alcatel OXO parameters:  IP address 192.168.92.246  Incoming port: 5060  version 7.7.1    2N VoiceBlue Next configuration  SIP trunk interconnection For the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed to secure communication with traffic from your PBX only. You can specify the IP address and port from which SIP packets will be accepted. In case you leave there 0.0.0.0 it will be open for all traffic.   Configuration of the LCR (Least Cost Routing) The GSM operator has e.g. in our country prefix 6 and 7 with nine digits in a number. The setting is below. You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.   Configuration of GSM outgoing groups You are able to set up different settings for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others). In case you don´t have a Ring back tone, set up Delay for ALERTING to option 4.   Incoming calls For incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls. In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback. You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button till the specified time. From the configuration, you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 101 to your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).     Alcatel OXO configuration  Setup a count of VoIP-Trunk channels for VoIP trunk to 2N® VoiceBlue Next    Choose Trunk group and check Public trunk check box. Change Link Category settings    At Link Category menu set up all necessary parameters    Assign Trunk Groups    Setup Numbering plans table. As Base settings choose ARS    Check Public numbering plan for incoming calls.  At Trunk Groups List assign List ID with Index at menu Numbering → Automatic Routing Selection → Trunk Groups List   Setup IP address of 2N VoiceBlue Next, codecs, bandwidth and GW keep alive timeout for VoIP trunk at menu Numbering → Automatic Routing Selection → Automatic Routing: Prefixes    At menu  Numbering → Automatic Routing Selection → Gateway   Parameters set up listening   port 5060. If you want to use 2N VoiceBlue Next with any different listening port (for example   5065), just set up the number of the port here.

Question 8

Asterisk - How to interconnect with Asterisk?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 10.0.0.20  Incoming port: 5060   Asterisk parameters:  IP address 10.0.0.10  Incoming port: 5060    Scenario  If we have an IP network in which Asterisk PBX, several SIP phones and 2N VoiceBlue Next are connected, the configuration would be as shown in the figure below. Furthermore, suppose that the network is addressed as shown in the figure and GSM numbers are all numbers starting with 6, 7 and containing 9 digits.     2N VoiceBlue Next settings  SIP trunk interconnection For the setting of the trunk between the 2N VoiceBlue Next and your Asterisk PBX, you need to configure "SIP proxy (GSM→IP)" for GSM incoming calls. "SIP proxy (IP→GSM)" is designed only for secure communication with the traffic from your Asterisk. You can specify the IP address and port where the IP packets will be accepted. If you leave there "0.0.0.0", the traffic will be unsecured. To enable incoming calls to Asterisk, you can register the 2N VoiceBlue Next directly into the Asterisk system. You can register it as "Friend" types in case you require registration based on username and password or "peer" type (on based of IP address and port). SIP registrar...an Asterisk IP address which registers the gateway Registration domain - IP address where the gateway is going to be registered Username...username under which the gateway shall be registered Password...registration password    Configuration of the LCR (Least Cost Routing) You have to specify prefixes for the operators in the country you are currently located. An example of this would be that in Czech Republic prefix 6 and 7 have a 9 digits number. The setting is displayed below.  You need to create specific guidelines connecting prefixes with the GSM group. In the "GSM group" you will specify settings for SIM cards assigned to this specific group. In the "GSM group assignment" you can assign the module for the appropriate GSM outgoing group.    Configuration of GSM outgoing groups You are able to have different setting for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others)   Incoming calls For incoming calls you can define 2 groups with the different behaviours and assign them to the GSM modules. The settings are similar with "GSM groups assignment" for outgoing calls.  In GSM incoming groups you can specify the traits for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.  You can define the list of numbers called. The number will be automatically dialled after the DTMF dialling has timed out. This happens when the customer does not press any button until the specific time. At this point, the number will be routed to the extension 100 to your Asterisk (if you set up SIP proxy (GSM->IP) in VoIP parameters).     Asterisk settings  Now add a few lines into the Asterisk configuration for proper routing of outgoing calls to the 2N VoiceBlue Next gateway and receiving calls coming from the GSM gateway to Asterisk.   Outgoing calls The core of Asterisk connection is saved in the /etc/asterisk/extensions.conf file. Open this file in your favorite editor and add the following lines: exten=>_6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r) exten=>_7XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,,r) Once you have saved and closed the file, restart Asterisk. From this point forward, all calls starting with 6 and 7 should be routed to the 2N VoiceBlue Next gateway.  Incoming calls It is highly recommended to make few restrictions for incoming calls in order to prevent unauthorised people from calling over your system. Since the 2N VoiceBlue Next system works with the SIP, modify the /etc/asterisk/sip.conf file where the 2N VoiceBlue Next section could look as follows: [general] port = 5060 bindaddr = 0.0.0.0 allowgues=no context = sip disallow=all allow=ulaw [VoiceBlueNext] type=peer host=10.0.0.20 username=voiceblue secret=password fromdomain=10.0.0.20 Furthermore restart the Asterisk after saving the file. Then the Asterisk will be ready to receive calls coming from the 2N VoiceBlue Next gateway.  What to do in case of trouble:  First of all check our web page faq.2n.cz and try to see if there is any solution to your problem. In case you cannot find the proper answer, use the link: How to report an issue on the 2N  VoiceBlue Next.    Here is the the direct link: https://jira.2n.cz/confluence/pages/viewpage.action?pageId=22513331

Question 9

Audio - What should I do if I can't hear ringback tone?

Accepted Answer

Please go to GSM Outgoing groups and set "Delay for CONNECT" = 0   and "Delay for ALERTING"=4   in all groups.   If you want to hear the tone and voice messages (like "We are sorry, the called party is not available.") from GSM network then verify that "Send 180 Ringing instead of 183 Session Progress" is disabled in VoIP parameters and that "Generate virtual tone" is disabled in GSM outgoing groups.   If you want to overwrite tones and voice messages coming from GSM network with generated tone by gateway then set appropriate "Ring tone" in GSM Basic Parameters and select "Generate virtual tone" in GSM Outgoing groups. Verify that "Send 180 Ringing instead of 183 Session Progress" is disabled in VoIP parameters.   If you want to overwrite tones and voice messages from GSM network by ring-back tone from IP PBX/IP Phones then enable "Send 180 Ringing instead of 183 Session Progress" in VoIP parameters.

Question 10

Audio - What to check if I have voice quality issue?

Accepted Answer

You need to check following points:  Signal quality – The quality of the signal must be in the range -65dBm and -85dBm in the module without call activity. You can check the quality of the signal in the configuration of the 2N VoiceBlue Next (Gateway control –> Module control) or by using the command at&qall from the hyperterminal or putty.  A Switch setting – Check, if the port is in the mode of FULL DUPLEX. Half duplex doesn't make sense for UDP RTP packet. In case there is just one call via this port, the quality can be satisfactory, but with more call traffic the issue with voice quality is present.  The best tool how to check if it´s caused by defective 2N VoiceBlue Next or other things is to connect the 2N VoiceBlue Next to the PC and make a direct SIP call from software phone as SJ phone. Use the called number in this format: sip:phonenumber@ipaddressofVoiceBlueNext

Question 11

Audio - What to check if You can't hear the ring-back tone?

Accepted Answer

Please check that you have latest firmware version (download section) and following settings:  Delay for ALERTING in GSM outgoing groups is set on 1 and higher  If you enable Virtual Ring Tone in GSM outgoing groups you must set Ring Tone in GSM basic parameters so gateway knows the tone to generate. Be aware that this setting overrides Ring tone (which can also contain e.g. the information that the calling party is not reachable, notice about transferred number, ringing tone etc.) and replaces it by the virtually generated ringing tone.

Question 12

Audio - What to check if you have voice quality issue?

Accepted Answer

You need to check following points:  Signal quality – The quality of the signal must be in the range -65dBm and -85dBm in the module without call activity.You can check the quality of the signal in the configuration of the 2N VoiceBlue Next (Gateway control –> Module control) or by using the command at&qall from the hyperterminal or putty.  Switch setting – Check if the port is in the mode of FULL DUPLEX. Half duplex does not make sense for UDP RTP packet. In case there is just one call via this port, the quality can be satisfying, but with more call traffic the issue with voice quality appears.  The best tool how to check if it is caused by defective 2N VoiceBlue Next or another things is to connect the 2N VoiceBlue Next to the PC and make a direct SIP call from software phone as SJ phone. Use the called number in this format: sip:phonenumber@ipaddressofVoiceBlueNext.

Question 13

AutoCLIP routing - How to set AutoCLIP routing?

Accepted Answer

What is AutoCLIP?  The GSM gateway automatically stores information about outgoing calls to the AutoCLIP routing table. When a person calls back the call is routed directly to the original number. If original number is not available the call is routed to the next number set up in the configuration (e.g. receptionist).    How to set in 2N VoiceBlue Next?  Parameters need to be set are in field Others   Time to keep CLIP in table - set the time duration for which records are to be kept in the AutoCLIP routing table.  Add record only for unconnected call - only unconnected outgoing calls (missed calls on a mobile phone) are stored in the AutoCLIP table.  Delete record for connected answer - if the call back is answered, the record in the AutoCLIP table will be deleted.  In the AutoCLIP routing table you can see the record of unconnected call.    When the user (737276015) has a missed call and returns that call, he will be directly forwarded to destination 11 as shown in the AutoCLIP routing table and also in the trace bellow.

Question 14

Automation, Calling two numbers from 2N IP Intercom over GSM gateway with a single SIM card

Accepted Answer

The steps in this article require using the web interface of a 2N device. Please follow this manual in case this is your first time using the web interface.  Target scenario This manual describes how to call two GSM numbers from 2N IP intercom over GSM gateway with one SIM card only. The scenario is that we would like, after pushing the button on the intercom, make a call to a GSM number (number 1) and if this call is not answered, intercom starts ringing the next number. If this were configured in a standard way, i.e. assigning numbers to the user and using Group calling, intercom would get busy tone when trying to reach the second number. This is caused by limitations of GSM technology — any GSM call takes way longer (few seconds) to hang up. On the other hand, SIP call is hung practically immediately. Therefore there must be added a delay between hanging up the first call and making the second call. This scenario requires to be configured in Automation:    Variables  KeyPressed - defines which key must be pressed to begin the sequence. In this example, Key 1 (first intercom button) is used.  BeginCall 1 - here you can define which GSM number or user must be first called. 123456 is used as an example number in this case.  BeginCall 2 - here you can define which GSM number or user must be called after nobody answered on the first number. 456789 is used as an example number in this case.  Delay 1 (D1) - defines how long will it ring on the first GSM number.  Delay 2 (D2) - must be always higher than Delay 1. D2-D1=delay between hanging up on the first number and making a call to the second number. We recommend 5 seconds difference to make sure the first GSM call is properly hung.  Delay 3 (D3) - must be always higher than Delay 2. D3-D2=ring time when ringing on the second GSM number.  Rest of the variables must be set exactly as you see on the picture above.   Firmware & License info  2N OS v2.37 with GOLD license  Last updated on 9th of January 2023  Note: All commands used in the Automation section are thoroughly described in our Automation manual altogether with corresponding examples.  The manual is available on our web page: Automation Manual

Question 15

Automation, Calling two numbers from 2N IP Intercom over GSM gateway with a single SIM card

Accepted Answer

The steps in this article require using the web interface of a 2N device. Please follow this manual in case this is your first time using the web interface.  Target scenario This manual describes how to call two GSM numbers from 2N IP intercom over GSM gateway with one SIM card only. The scenario is that we would like, after pushing the button on the intercom, make a call to a GSM number (number 1) and if this call is not answered, intercom starts ringing the next number. If this were configured in a standard way, i.e. assigning numbers to the user and using Group calling, intercom would get busy tone when trying to reach the second number. This is caused by limitations of GSM technology — any GSM call takes way longer (few seconds) to hang up. On the other hand, SIP call is hung practically immediately. Therefore there must be added a delay between hanging up the first call and making the second call. This scenario requires to be configured in Automation:      Variables  KeyPressed - defines which key must be pressed to begin the sequence. In this example, Key 1 (first intercom button) is used.  BeginCall 1 - here you can define which GSM number or user must be first called. 123456 is used as an example number in this case.  BeginCall 2 - here you can define which GSM number or user must be called after nobody answered on the first number. 456789 is used as an example number in this case.  Delay 1 (D1) - defines how long will it ring on the first GSM number.  Delay 2 (D2) - must be always higher than Delay 1. D2-D1=delay between hanging up on the first number and making a call to the second number. We recommend 5 seconds difference to make sure the first GSM call is properly hung.  Delay 3 (D3) - must be always higher than Delay 2. D3-D2=ring time when ringing on the second GSM number.  Rest of the variables must be set exactly as you see on the picture above.   Firmware & License info  2N OS v2.44 with GOLD license  Last updated on 11th of July 2024

Question 16

Avaya - How to interconnect with Avaya Aura Communication Manager using SIP

Accepted Answer

Avaya - How to interconnect with Avaya Aura Communication Manager using SIP  Following FAQ page describes how a 2N gateway capable of VoIP to GSM calls can be connected to Avaya Aura Communication Manager using SIP. If you are currently using firmware that is older than version 1.20 (1.19 and lower) please upgrade your firmware version.   This FAQ applies for following gateways:  2N® BRI Lite, 2N® BRI Enterprise (2 channels)  2N® VoiceBlue Next (2 or 4 channels)  2N® VoiceBlue MAX (4 channels)  2N® BlueTower (2 to 8 channels), 2N® StarGate (2 to 32 channels).   Parameters of 2N gateway:  IP address 10.10.16.2  Incoming port: 5060  Firmware: 1.17.0.20.2   Avaya equipment parameters:  Avaya Aura® Communication Manager  version: R6.3 Build R016x.03.0.124.0  update: 03.0.124.0-20850   Avaya Aura® System Manager  version: R6.3 Build 6.3.0.8.5682-6.3.8.1814  update 6.3.3.5.1719   Avaya Aura® Session Manager  version: R6.3 Build 6.3.3.0.633004  IP address: 10.10.16.214     Scenario   We have an IP network in which an Avaya solution (that consists of Communication Manager, System Manager, Session Manager and an Avaya G430 Gateway), several SIP phones and 2N® VoiceBlue Next are connected.    The Communication Manager is configured to communicate to the 2N VoiceBlue via a SIP Trunk configured on the Session Manager.    The configuration would be as shown in the figure below.            2N® VoiceBlue Next settings  SIP trunk interconnection¨  For the setting of the trunk between the 2N® VoiceBlue Next and Avaya Aura Communication Manager you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for secure communication with traffic from your Avaya Aura Communication Manager only. You can specify the IP address and port from which SIP packets will be accepted.In case you leave there 0.0.0.0 it will be open for all traffic.    In the IP addresses section, enter the following:     SIP proxy (IP->GSM): Enter the IP address of the Session Manager and Port number  SIP proxy (GSM->IP): Enter the IP address of the Session Manager and Port number  Ring tone to VoIP: Select From GSM from the drop box     Configuration of the LCR (Least Cost Routing) The GSM operator has e.g. in our country prefix 6 and 7 with a nine digits in the number. The setting is below.  You need to create LCR rule for defined prefixes. In firmware 1.20 the LCR process has been changed, you can find more information about the change here:Call routing - Changes in call routing in firmware 1.20+ compared to earlier versions on 2N VoiceBlue NextIn default LCR table is already configured to send calls from VoIP to GSM over the GSM outgoing group 1. If you are using only outgoing GSM group 1, you don't have to change anything. If you need to setup call routing over different GSM outgoing groups, please look at following FAQ that will explain what you need to configure:Call routing - SIM selection based on called number (version 1.19 and lower) on VoiceBlue Next       Configuration of GSM outgoing groups You are able to set up different settings for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others).    Incoming calls For incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.  In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.  You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button within the specified time. From the configuration you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 100 (if you set up SIP proxy (GSM->IP) in VoIP parameters).    Avaya Aura® Communication Manager settings Configurations of the Communication Manager illustrated in this section were all performed using Communication Manager’s SAT administration interface. It is implied a working system is already in place including a Node for the Session manager and procr.  Configure signalling-group A signalling group is required before a trunk-group can be configured. Use the add signalling-group command followed by next available signalling-group number to configure the following:  Group Type: Enter sip  Transport Method: Enter tcp  Near-end Node Name: Enter procr  Far-end Node Name: Enter SM63RPSIG (this is the Node of the Session Manager)  Near-end Listen Port: Enter 5060  Far-end Listen Port: Enter 5060  Far-end Domain: Enter devconnect.local (This is the domain configured on the Session Manager)   Configure Trunk Group  This section describes the Trunk Group configuration used during compliance. Use the add trunk-group command followed by next available Group number and configure the following:  Group Type: Enter sip  Group Name: Enter an informative name for the trunk i.e. To SM6.3 SIP  TAC: Enter a TAC number i.e. 701  Service Type: Enter public-ntwrk  Signalling Group: Enter 1 (This is the Signalling Group configured in Section 5.1)     Configuring Avaya Aura ® Session Manager A number of configurations are required to enable the Session Manager to route calls between the Communication Manager and 2N VoiceBlue Next. All configurations of the Session Manager are performed using System Manager.  Create 2N® VoiceBlue Next as a SIP Entity Log into web interface of your Avaya Aura® System Manager and under the Elements column select Routing.  To add a SIP Entity, select SIP Entities on the left panel menu and then click on the New button Enter the following parameters for 2N VoiceBlue SIP Entity under General:  Name: Enter an informative name (e.g., 2N VoiceBlue Next)  FQDN or IP Address: IP address of the signalling interface of the 2N VoiceBlue Next  Type: Select SIP Trunk  Location: Select location from the drop-down box, we have selected DevConnectRP  Time Zone: Select Time Zone for this location  SIP Timer: Enter 4-  Once the correct information is entered click the Commit Button.   Create an Entity Link  Click Commit to save changes. The following screen shows the Entity Links used.  The SIP trunk between the Session Manager and the 2N VoiceBlue requires an Entity Link. To add an Entity Link, select Entity Links on the left panel menu and click on the New buttonEnter the following:  Name:Name An informative name, (e.g. VoiceBlue Next).  SIP Entity 1: Select SM63 from the SIP Entity 1 drop-down box.  Protocol Select: TCP from the Protocol drop down box  Port: Enter 5060  SIP Entity 2: Select 2N VoiceBlue Next from the SIP Entity 2 drop-down box (configured in previous section)  Port: Enter 5060  Connection Policy: Check the Trusted check box  Click Commit to save changes. The following screen shows the Entity Links used:    Create a Routing Policy To direct calls to the 2N VoiceBlue via the Session Manager you need to create routing policies. To add routing policy select Routing Policies on the left panel menu and then click on the New button.Enter the following:  Name: Enter an informative name (example, To GSM)  Retries field: Enter 0  SIP Entity: Select Destination  Click the Select button  Once the SIP Entity List screen opens, check the 2N VoiceBlue radio button. Click on the Select button to confirm the chosen options and then return to the Routing Policies Details screen and select the Commit button to save.    Create a Dial Pattern To be able to route calls to 2N VoiceBlue Next you need to create a Dial Pattern on Session Manager. On the left panel select Dial Pattern and click on New.In example below a dial pattern has been created where all calls starting with 086 will be sent to 2N VoiceBlue Next. Under General carry out the following for each number   Pattern: Enter 086  Min: Enter 10 as the minimum length of dialled number  Max: Enter 10 as the maximum length of dialled number  SIP Domain: Select devconnect.local from the drop down box  Click the Add button in Originating Locations and Routing Policies  As a last step go to Originating Location and check the DevConnectRP check box. Under Routing Policies check the To GSM check box and click on Select button to confirm new setting.  Return to Dial Pattern and finally click on Commit to save the change.

Question 17

Callback - How to set Callback on 2N VoiceBlue MAX?

Accepted Answer

Callback feature allows you to save money on mobile calls made from abroad to the office. The mobile phone which is in a foreign country calls the SIM number of 2N VoiceBlue MAX and stops calling after hearing one ringing. 2N VoiceBlue MAX will callback the mobile phone and once the call is accepted, all the SIP extensions can be accessed saving roaming costs of a conventional international call. This feature allows calling on expenses of SIM in the gateway. It can be used for national calls as well.   Create incoming group for callbackGo to menu Gateway Configuration ==> GSM Incoming GroupsIf you want to dedicate GSM group just for the callback, choose either mode Callback after ring/Reject (other callers will be rejected) o Callback after ring/Ignore (other callers will be ignored). If you want to share the GSM group for callback users and other callers from GSM then select Accept incomingcalls + dialtone or Accept incoming calls + voice message. Add + in CLIP(- removes one digit). Set the configuration of the digits that is necessary when calling an extension (Minimum and Maximum digits in DTMF). Set enough timeout to dial the numbers in DTMF dialling timeout (10 for example).Save settings.  Specify numbers in CLIP Routing TableNext step is to specify the phone numbers that are allowed to use Callback. Go to Gateway Configuration and select CLIP routing table. Add a new number to the table (for example: +34662314159). This number will be recognised by the system. The field "Used service" must be disabled (None) and make sure that Auto Callback service check box is on.   Save settings   Prefix configurationIn order to callback we need to set up the international prefix in menu Gateway Configuration->Prefixes. In this case all numbers will be processed including its international prefix (in the example: +34). Fill in the field "Prefix" and "Replace with" with +34 and then press Add.   LCR tableGo to LCR table menu and check how the calls are processed. In this case GSM group 1 and Prefixlist 1 must be selected. You can use functions "Add" and "Edit record". The first row will be the first to be processed in LCR router.   Further checking Although you think 2N VoiceBlue MAX is properly configured, the Callback function is still not working. Please make a trace using Hyperterminal, Putty or other telnet program and search in the text if the call establishment is indicated with the CBACK message.  In case if this message is not there, please check the configuration once again, otherwise check if LCR isrouting correctly.

Question 18

Callback - How to set Callback on ViceBlue Next?

Accepted Answer

Callback feature allows you to save money in mobile calls made from abroad to the office. The mobile phone which is in a foreign country calls the SIM number of VBN and stop calling after hearing one ring. VBN will callback the mobile phone and once the call is accepted, all the SIP extensions can be accessed saving roaming costs of conventional international call. This feature allows calling on expense of SIM in gateway. It can be used for national calls as well.   Create incoming group for callbackGo to menu Gateway Configuration ->GSM Incoming GroupsIf you want to dedicate GSM group just for callback choose either mode Callback after ring/Reject (other callers will be rejected) or Callback after ring/Ignore (other callers will be ignored). If you want to share GSM group for callback users and other callers from GSM then select Accept incoming calls + dialtone or Accept incoming calls + voice message. Add + in CLIP(- removes one digit) . Set the configuration of the digits which is necessary to call an extension (Minimum and Maximum digits in DTMF). Set enough timeout to dial the numbers in DTMF dialling timeout (10 for example).Save settings.  Specify numbers in CLIP Routing TableNext step is to specify the phone numbers that are allowed to use Callback. Go to Gateway Configuration and select CLIP routing table. Add a new number to the table (for example: +34662314159). This number will be recognised by the system. Used service must be disabled (None) and make it sure that Auto Callback service check box is on.     Save settings  Prefix configurationIn order to callback we need to set up the international prefix in menu Gateway Configuration->Prefixes. In this case all numbers will be processed including its international prefix (in the example: +34). Fill prefix and Replace with with +34 and the press Add.    LCR tableGo to LCR table menu and check how the calls are processed. In this case GSM group 1 and Prefixlist 1 must be selected. Otherwise you can edit or add , just notice that the first row will be the first to be processed in LCR router.    Further checking It can happen that 2N VoiceBlueNext is properly configured but Callback function still does not work. Please make a trace using Hyperterminal,Putty or other telnet program and search in the text if the call establishment is indicated with the CBACK message   In case it is not present, please check the configuration once more, otherwise check if LCR isrouting correctly.

Question 19

Callback - How to set SMS Callback on VoiceBlue Next

Accepted Answer

Basic description of this feature  How does SMS CallBack work with 2N VoiceBlue Lite device? Basically it is possible to send SMS to the telephone number of SIM card inserted in 2N VoiceBlue Lite device and receive a callback connecting to a specific SIP extension. After sending the SMS in correct form, the gateway will call to the telephone number from which the SMS has been received. Once the call is picked up, you will be automatically connected with the required extension. 120 in the example.  Call back from GSM to IP: GSMIP (VoIP)   Necessary software:  2N XAPI Server v. 1.20A  SMS CallBack centre v. 1.0 → license needed     Installation of 2N XAPI server  On the link bellow you are able to find detailed manual how to install and set up 2N XAPI Server.http://www.2n.cz/en/products/extras/sms-server/downloads/  Download sms-sever-manual-1en.pdf and follow the instructions from page 4 to page 8.  For a SMS Callback purpose please create a specific user with these features (check SMS and SMR). Select Firm: D-Callback-2N. Set desired Name and password (in this example: jarda/jarda).      Basic settings of 2N VoiceBlue Lite  On following pictures there is explained how it is recommended to set up the device in order to provide you SMS CallBack feature.   VoIP parameters    In SIP proxy (GSM->IP) set the IP address of PBX (192.168.50.5 in this case) and specify the listening port (5061 in the example). Remember that at least one extension must be connected to the PBX in order to receive the calls (we assigned 120).    GSM group assignment    GSM modules can be assigned up to 4 groups that can be independently assigned for incoming and outgoing calls. In this case all modules are assigned to Group 1.    GSM incoming groups    In option „Mode" in the top part of window choose Report to PC +dial tone. Then fill + to the clip field.    Prefixes    Make sure the prefix configuration is properly set for your operator. On the example below the /+420 prefix is added (international prefix for Czech Rep.)      LCR Table    Click to "Add" and fill required information in the first row (marked with red frame).    Form of activation SMS  Because of structure of 2N VoiceBlue Lite device you have send the SMS in this form*sg*Phone number of destination  SMS example: *sg*120     Settings of SMS CallBack centre  Logging to XAPI    It is necessary to login SMS Callback center to XAPI server. Username and password corresponds with XAPI server users settings. In this example it is jarda/jarda.      Users    Create a new user - fill the CLIP and in the field Tariffs of groups activate the interval 1.      Tariff templates      Create a new Tariff template by click to the button „New"       Groups    In the Group which we added a prefix is required. In the example the final destination is 120 so it is necessary to fill the prefix "1". Do not forget to set prefixes regarding the extension number which the call will be connected to.  Other settings: The rest of configuration regarding Prefixes to GSM, Prefixes to ISDN Settings and Routing is recommended to remain as default.

Question 20

Call disconnection - What to do if a call is not ended immediately?

Accepted Answer

If the call is not ended immediately by VoiceBlue Next then check Busy Tone settings in GSM Basics parameter. This parameter influence call which is ended from VoIP side. If you set None, then the call will end immediately. If you set the tone then the gateway will keep the connection in GSM to generate the busy tone for up to 30s.

Question 21

Call disconnection - What to do if a call still continues up to 30s after call ended?

Accepted Answer

Issue description: The call is disconnected from VoIP side and it takes up to 30 seconds before the call is disconnected in GSM side. This typically happens if a voice mail is reached. If you make a test call to your cell phone, you can hear a busy tone. This behaviour can be caused by PBX settings or by 2N VoiceBlue Next settings. Please check following page:  Generate busy tone to GSM/UMTS is used to simulate an analogue line where you get a busy tone after disconnection. If you want the immediate disconnection, set this parameter to Disabled. Similar settings could be necessary on PBX side.

Question 22

Call routing - Changes in call routing in firmware 1.20+ compared to earlier versions on 2N VoiceBlue Next

Accepted Answer

In firmware 1.20 the way call routing was configured has been changed. In this FAQ you are going to find out what is different from previous firmware versions.   On 2N VoiceBlue Next gateway the only visible change has been done to LCR table. Every other element of call routing has been left as it was before (GSM groups assignment, setting of Outgoing/Incoming GSM group, prefix lists,..) New LCR table allows you to have a complete control over where an incoming call (from VoIP or GSM) should be routed to. On pictures bellow you can see examples of differences between the new LCR table and the old one.    1) Explanation of new LCR table Records in LCR table are read from top to bottom. When a record that matches all conditions is found than the call is going to be sent to outgoing destination specified in that record. Therefore it is important to pay attention to the order in which you have your LCR table records organised!    From (channels/groups) - this parameter specifies where outgoing call originates (in case of VoiceBlue Next it can be GSM incoming group or VoIP) Prefix list - you must specify either one prefixlist, that the outgoing call number must match, or you can select option All prefixlists, where any outgoing call number that matches at least one prefixlist, will be routed by channel specified under Groups/Tariffs Groups/Tariffs - this parameter specifies if outgoing call should go through a specific GSM outgoing group or through VoIP   This way it is possible to route incoming calls from GSM back to GSM which was not possible in older firmware version.   2) Default LCR table settings   Firmware 1.20   Firmware 1.19 and older         You have two records:  One record that routes calls from GSM to VoIP  Second record that routes calls from VoIP to GSM (via GSM outgoing group 1)   You have only one record that routes all incoming calls from VoIP to GSM (via GSM outgoing group 1)     3) Routing outgoing calls based on prefix lists (2 outgoing GSM groups), incoming calls are routed to VoIP    Firmware 1.20    Firmware 1.19 and older          The only difference from previous version is a line specifying that incoming calls from GSM should go to VoIP  Two records, one for each GSM outgoing group     4) Incoming call from GSM group 4 (matching prefix list 2) will be routed back to GSM through GSM outgoing group 2. Every other call will be sent to VoIP. All outgoing calls from VoIP are routed to GSM through GSM outgoing group 1    Firmware 1.20   Firmware 1.19 and older     This function was not possible  On picture above you can see how to set a scenario where all incoming calls from GSM incoming group 4should be sent back to GSM through GSM outgoing group 1 (if dialled number will match Prefix list 2)  This function was not possible     This function was not possible  On the picture above you can see how to set described scenario. All calls from VoIP are going through GSM outgoing group  This function was not possible     To find out more about how to setup call routing please look at FAQ: Call routing - SIM selection based on called number (version 1.19 and lower) on VoiceBlue Next

Question 23

Call routing - Forwarding call to one extension number - VoiceBlue Next

Accepted Answer

Following guide explains how to route the call directly to the defined extension number. The call on GSM side is answered after it was answered on VoIP side. Example: All calls routed to extension 100 in VoIP. Gateway configuration/GSM Group Assignment - put all modules in one Incoming group  Gateway configuration/GSM Incoming Groups - change following settings in Incoming group 1  Mode: Set Accept incoming calls + dial tone or Accept incoming calls + voice message (it works the same way as if DTMF dialling time-out is set on 0)  Minimum and Maximum digits in DTMF: Set correct length of digits (for example if you want to dial 100, put 3, if you want to dial 10, put 2)  DTMF dialling time-out: Set 0 which means that the gateway does not wait for any dialling from a mobile user and the call is immediately routed to numbers in List of called numbers     Scroll down to List of called numbers and insert number of extension that you want to dial automatically. You can specify more numbers. If the first number is busy then the next number will be dialled.     Gateway configuration/VoIP parameters: Verify that correct IP of your PBX is set so that the gateway knows where to send the call     The gateway is ready. Make a test call. Note: If you want to dial a different number in VoIP for incoming call on each SIM then put GSM modules in more incoming groups and set other groups in the same way, only with a different number in List of called numbers. Note 2: Add a new route in LCR if there is firmware 1.22 and higher

Question 24

Call routing - How can I route all incoming calls to receptionist or operator?

Accepted Answer

Go to the  GSM incoming group  menu and set appropriate  GSM group  . Then set timeout for entering  DTMF  digits to  0  . On the right side fill your operator/receptionist number into l  ist of called number table  . This number should match max and min digits settings above timeout.

Question 25

Call routing - How many simultaneous calls can I do?

Accepted Answer

It depends on version of your 2N VoiceBlue Lite gateway. When you have got version with two GSM modules, you can make two calls only. When you have got version with four GSM modules, it means four simultaneous calls then.

Question 26

Call routing - SIM selection based on called number

Accepted Answer

In firmware version 1.20 we have changed the way how call routing is set up. This FAQ is going to show you how to select SIM card for outgoing call in firmware  1.20+  . If you want to be able to select the SIM for outgoing call, then it can be done with "technical" prefix which is removed by the gateway before dialling the number to GSM/UMTS networks. Example: You want to dial number 789456123, so instead you will send to the gateway 01789456123 to select SIM1. The gateway will cut 01 and dial 789456123 through SIM1. The guide is following this example for 2 channel 2N VoiceBlue Next. Settings on PBX side You will have to set your PBX to add following prefixes (or other) based on your requirements  01 to dial through SIM1  02 to dial through SIM2  Settings on VoiceBlue Next Gateway configuration/GSM Group Assignment - put each module inside different Outgoing group   Gateway configuration/Prefixes - set different technical prefix in each prefix list. I am going to configure 2 channel gateway. I will set just 2 prefix lists. Prefix list 1   Prefix list 2   Avoid settings bellow. "/" is a indeterminate character which allows to skip Table of replaced prefixes. The beginning of the called number is then compared only with the right table. When you enter 0 in the right table, the gateway would dial 02789456123 instead of 789456123 as we wish.   Gateway configuration/LCR Table - select where incoming/outgoing call should be routed based on prefix list - we are going to interconnect prefixes with outgoing groups (which represent specific module/SIM card) First of all we need to add a new rule to LCR table   Click Add and fill Prefixlist 1 and Outgoing GSM group 1 (you don't need to do this step since this is one of the predefined rules in default):   Click Add and fill Prefixlist 2 and Outgoing GSM group 2:   Save the result which must look like this:   Now everything is set. Make a call to verify settings.

Question 27

Call routing - SIM selection based on called number (version 1.19 and lower) on VoiceBlue Next

Accepted Answer

If you want to be able to select the SIM for outgoing call, then it can be done with "technical" prefix which is removed by the gateway before dialling the number to GSM/UMTS networks. Example: You want to dial number 789456123, so instead you will send to the gateway 01789456123 to select SIM1. The gateway will cut 01 and dial 789456123 through SIM1. The guide is following this example for 2 channel 2N VoiceBlue Next. Settings on PBX side You will have to set your PBX to add following prefixes (or other) based on your requirements  01 to dial through SIM1  02 to dial through SIM2  Settings on VoiceBlue Next Gateway configuration/GSM Group Assignment - put each module inside different Outgoing group  Gateway configuration/Prefixes - set different technical prefix in each prefix list. I am going to configure 2 channel gateway. I will set just 2 prefix lists. Prefix list 1   Prefix list 2   Avoid settings bellow. "/" is a indeterminate character which allows to skip Table of replaced prefixes. The beginning of the called number is then compared only with the right table. When you enter 0 in the right table, the gateway would dial 02789456123 instead of 789456123 as we wish.   Gateway configuration/LCR Table - interconnect prefixes with outgoing groups (which represent specific module/SIM card) Click Add and fill Prefixlist 1 and Outgoing GSM group 1:   Click Add and fill Prefixlist 2 and Outgoing GSM group 2:    Save the result which must look like this:   Now everything is set. Make a call to verify settings.

Question 28

CCM Express version 4.1 - How to interconnect with CCM Express version 4.1?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.22.42  Incoming port: 5060   Cisco Unified Communication Manager Express parameters:  IP address 192.168.22.35  Incoming port: 5060    2N VoiceBlue Next Settings  SIP Trunk InterconnectionFor the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for secure communication with traffic from your CME only. You can specify the IP address and port from which SIP packets will be accepted.In case you leave there 0.0.0.0 it will be open for all traffic.  Configuration of the LCR (Least Cost Routing)The GSM operator has e.g. in our country prefix 6 and 7 with a nine digits in the number. The setting is below.  You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.     Configuration of GSM outgoing groups:You are able to set up different settings for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others).     Incoming callsFor incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.  In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.  You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button within the specified time. From the configuration you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 100 to your CME (if you set up SIP proxy (GSM->IP) in VoIP parameters).    Cisco Unified Communication Manager Express setting  Freeware program was used for the configuration- Cisco Configuration Professional - version 2.3.  For configuration enter the menu: Configure → Voice → Dial Plan → VoIP and set up the prefix and IP address to the 2N VoiceBlue Next gateway.  The prefix 6.T in the example means that prefix is 6 plus other digits after 6 without limit.    In the picture below you can see the configuration program with the saved routing to the 2N VoiceBlue Next.    Incoming calls are automatically enabled by a new trunk. All incoming calls from 2N VoiceBlue Next will be routed to stations in CME or you can create your own dial plan.  In the CME version 4.1 you are not able to register SIP phones.    !!! Download the PDF by clicking on the arrow on the left side!!!

Question 29

CDR - How can I delete CDR?

Accepted Answer

The command to delete call records is  at&cr  . When you want to delete any line from CDR table you need to confirm this at command by ASCII sign #. Then it is impossible to delete something inadvertently.

Question 30

CLIP - Incoming CALL ID transfer to phone or PBX

Accepted Answer

The CLI transmission is via FSK which must be supported by connected PBX or phone. (CLI transmission with DTMF protocol is not supported)      Go to Gateway Configuration/FXS Interface:  Set Ring profile, Tone profile, Call ID profile to your country or to the closest country from the list. Example:

Question 31

Configuration - Can I use 2N VoiceBlue Next with my PBX?

Accepted Answer

2N VoiceBlue Next uses the protocol SIP for calling. Check if your PBX supports this VoIP protocol. Below you can find a link to our pages where you can check a document with examples of IP PBX systems. If you can not find your IP PBX, then the importing thing is whether your IP PBX uses protocol SIP. This protocol is required by 2N VoiceBlue Next Please, go to the following link Supported PBXs and click on the tab Other.

Question 32

Configuration - How do I configure 2N VoiceBlue Lite?

Accepted Answer

You need a special configuration tool, which you can find on our web page. Then you need to connect this configuration tool with your gateway via TCP/IP, USB or serial cable. Please be aware that via TCP/IP it is impossible to upload a new firmware, new license key or change IP address.

Question 33

Configuration - How to set 2N VoiceBlue MAX

Accepted Answer

In firmware version 1.20 we have changed the way how call routing is set up. This FAQ is going to show you how to set 2N VoiceBlue MAX in firmware  1.20+  . If you are currently using firmware that is older than version 1.20 (1.19 and lower) please upgrade your firmware version or go here: older FW version, where you will find FAQ on how to set 2N VoiceBlue MAX in older FW version.    2N VoiceBlue MAX is a device that helps directly interconnect the SIP-supporting VoIP network with GSM networks and can be used for direct connection with a SIP telephone set. Besides, 2N VoiceBlue MAX contains 4x FXS ports for analogue telecommunication equipment. The voice mode, i. e. outgoing and incoming calls, is the basic function of the gateway. 2N VoiceBlue MAX is equipped with all voice mode functions and provides the highest user comfort. In addition to voice transmission, 2N VoiceBlue MAX enables you to send and receive text messages (SMS). You can use the web interface or AT commands for all the gateway settings. The programmable parameters are set in such a way that you can make calls the moment you connect the system to the Ethernet, connect an antenna and insert the SIM card. 2N VoiceBlue MAX can be combined with the 2N Mobility Extension service (remote GSM extension) for up to 8 users.  SIM Card Placement Insert the SIM cards into one of the four SIM holders on the 2N VoiceBlue MAX front and make sure that the chip contacts are at the front bottom. The SIM holders are equipped with the push&pull technology for facilitation.     How to connect to the 2N VoiceBlue MAX  The configuration of the 2N VoiceBlue MAX is done via the web interface  In the following chart there are the factory settings for the key configuration parameters of the 2N VoiceBlue MAX   Connect the 2N VoiceBlue Next via Ethernet cable directly to the computer. Change the IP address setting in the configuration of your PC (e.g. 192.168.1.5 and Subnet Mask 255.255.255.0)   Open your web browser and enter the default IP address of GSM gateway 192.168.1.2 with the default credentials (Admin/2n). If web page does not appear, then do a factory reset of the gateway with RESET button on the front panel. Press it for 10 seconds. Afterwards the gateway restarts and you can reload the page.  In the Gateway configuration / Ethernet configuration menu you are able to change the IP address.     How to set up the Call Routing In the VoiceBlue MAX calls can come from the FXS Ports or from the VoIP port and calls are routed to any GSM/UMTS port according to the LCR (Least Cost Routing) table. If the GSM port is busy, other ports are checked automatically for availability (depending on the configuration) and in case if an allowed outgoing port is not available, the outgoing call will be rejected. The LCR algorithm identifies the outgoing call type, current time tariff rate, day in a week, eventually free minutes of GSM providers and routes outgoing calls accordingly. Incoming calls from GSM networks are routed directly either to the FXS Ports or to the defined SIP address, or the DISA function is activated. Furthermore, calls can be routed according to the CLIP (caller's telephone number). And the CallBack service is also available. If you want to check current settings of call routes, go to the Menu Gateway Configuration / LCR table    How to set up the LCR Table The LCR (Least Cost Routing) helps you select outgoing call routes according to the port the outgoing calls is originating from (FXS, GSM or VoIP), called subscriber's number (or some prefix), time of the day and weekdays. Whenever a call is being done, the LCR table is searched sequentially from top to bottom. If the called destination prefix matches the network prefix included in the List of networks table (and designated as Network number in the LCR table), the call is routed according to the routing group parameters (Groups in the LCR table) as set in the GSM outgoing groups table. The call will be connected via the GSM module which was assigned to the GSM outgoing group in the Assignment to GSM groups table. Therefore, you need to set and configure:  GSM Group Assignment  GSM Outgoing Groups  Prefixes  Example   Let’s say we have SIM cards of two GSM operators:  Operator 1: Using the following prefixes (602, 606, 607, 723, 724) and it requires you to dial the number from your mobile phone with the international prefix (+420). All numbers have nine digit length with the operator prefix but without the international prefix.  Operator 2: Using the following prefixes (901, 902, 907, 909) and requires you to dial the number with the prefix (+420). All numbers have nine digit length with the operator prefix and without “+420”.  GSM Group Assignment  First we have to go to the menu Gateway Configuration / GSM Group Assignment and assign modules to two GSM outgoing groups (for each operator). Let's assume that   Module 0 and 1 will have SIM cards from Operator 1 (Outgoing Group 1)  Module 2 and 3 will have SIM cards from Operator 2 (Outgoing Group 2)     Configuration of GSM outgoing groups You are able to set up different settings for each GSM group (CLIR, free minutes, Virtual ring tone, roaming and others).   Prefix lists We have to create two network lists, the first one for both of the Operators.  Operator 1 network list:We configure the normalisation of Called party number in the Table of replaced prefixes (the number in front of the slash mark is replaced by number behind the slash mark, if there is not any number in front of the slash mark it equals to “everything”).We also have to fill in the Table of prefixes with all prefixes of the Operator 1. Because all numbers are 9 digits length, it is not necessary to specify each prefix, we can use the parameter Default count of digits and fill there the value 9.The Table of replaced prefixes, Table of accepted prefixes and Digits count must correspond with the called number and it leads to the match in prefix list   Operator 2 network list:We configure the normalisation of Called party number in the Table of replaced prefixes (the number in front of the slash mark is replaced by number behind the slash mark, if there is not any number in front of the slash mark it equals to “everything”). We also have to fill in the Table of prefixes with all prefixes of the operator 2. Because all numbers are 9 digits length, it is not necessary to specify each prefix, we can use the parameter Default count of digits and fill there the value 9.    LCR TableWe have to configure rows in the LCR table where we bind together GSM Outgoing groups with Prefix lists and ports where call is originating. Please note that LCR table is searched sequentially from top to bottom. So position of each rule in LCR table is important. If there is a match found in a LCR rule than the call is routed to the destination specified in that rule. All other rules that are after the successful one do not matter. In default LCR table looks like this:  Where each line means following:  1st line - all calls from GSM are going to be routed to FXS interfaces  2nd line - calls from FXS ports, where called number matches prefix list 1, will be routed to GSM through GSM outgoing group 1  3rd line - calls from FXS ports, where called number matches any prefix list, will be sent to VoIP  4th line - calls from VoIP, where called number matches prefix list 1, will be routed to GSM through GSM outgoing group 1  5th line - calls from VoIP, where called number matches any prefix list, will be sent to FXS interfaces  To complete setting of our scenario we need to add 2 more lines (or only 1 based on if you are calling only from FXS or only from VoIP). We already have 2 lines that are routing outgong calls from FXS ports to GSM outgoing group 1 and calls from VoIP to GSM outgoing group 1. We need to add only lines for outgoing GSM groups 2.  Add rule for calls from FXS to GSM outgoing group 2 (called number must match Prefixlist 2)Click on ADD button to add new line and configure it for Operator 2 Click on OK and move the LCR rule into position under 2nd line.  Add rule for calls from VoIP to GSM outgoing group 2 (called number must match Prefixlist 2)Click on ADD button and add new line and configure it for Operator 2 Click on OK and move the LCR rule into position before the last line  Now your LCR table should look like this:  If you want to find out more information about LCR table please go to following FAQ: LCR table in FW 1.20 where you can find more information on how to set LCR table   How to set up connection to the opposite device (PBX)  For the settings of the trunk between the 2N VoiceBlue MAX and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for security and limitation of communication only on your PBX. You can specify the IP address and port from which SIP packets will be accepted.  In case you leave there 0.0.0.0 in SIP proxy (IP->GSM), the gateway will be open for traffic from all IPs. !!! This can lead to misuse of equipment by an unauthorised person !!!     Incoming calls  For incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.   In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.   In the mode Accept you can define the list of called numbers which will be automatically dialled after DTMF dialling timeout in case that the customer does not press any button within the given time. From the configuration you can see 5 seconds for DTMF dialling and after that the call will be routed to the extension 100 in your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters). If you want immediate connection then set DTMF dialling timeout to 0.     SIM cards  In the Gateway control and module control you can see the status of the SIMs. The recommendation for quality of the signal is between -65 and -85dBm.   The logged SIM card is indicated by the led diode switched off on the 2N VoiceBlue MAX

Question 34

Configuration - How to set 2N VoiceBlue Next?

Accepted Answer

In firmware version 1.20 we have changed the way how call routing is set up. This FAQ is going to show you how to configure 2N VoiceBlue Next in firmware  1.20+  . If you are currently using firmware that is older than version 1.20 (1.19 and lower) please upgrade your firmware version     2N VoiceBlue Next is designed to reduce the costs of outgoing calls to GSM networks. The main advantage is the ability to expand the IP PBX to include a GSM gateway and make use of a LCR which selects the best GSM operator depending on the called number.   How to connect 2N VoiceBlue Next  Connect the 2N VoiceBlue Next via Ethernet cable directly to the computer. Change the IP address setting in the configuration of your PC (e.g. 192.168.1.33 and Subnet Mask 255.255.255.0)  Open your web browser and enter the default IP address of GSM gateway 192.168.1.2 with the default credentials (Admin/2n). If web page does not appear, make a factory reset of gateway with RESET button on front panel. Press it for 10 seconds. After this time gateway restarts and you can reload the page.  In the gateway configuration and Ethernet configuration you are able to change the IP address.    How to set up connection to the opposite device (PBX)  For the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for secure communication just with traffic from your PBX. You can specify the IP address and port which will accept SIP packets from.  In case you leave there 0.0.0.0 in SIP proxy(IP->GSM) gateway will be open for traffic from all IPs. !!! This can lead to misuse of the equipment by an unauthorised person !!!!    How to configure LCR table Situation…  Let’s say we have SIM cards of two GSM operators:  T-Mobile: Using the following prefixes (602, 606, 607, 723, 724) and it requires that you dial the number from your mobile phone with the international prefix (+420). All numbers have a nine digit length with the operator prefix but without the international prefix.  Vodafone: Using the following prefixes (901, 902, 907, 909) and requires you to dial the number with the prefix (+420). All numbers have a nine digit length with the operator prefix and without “+420”.   First we have to assign modules to two GSM outgoing groups (for each operator).The first step which we have to do is to place the SIM cards into the SIM holders. We will start with modules 0 using the T-mobile operator SIM. Module 2 will follow the same procedure but using the Vodafone SIM.   Configuration of GSM outgoing groupsYou are able to set up different setting for each GSM group (CLIR, free minutes, Virtual ring tone, roaming and others).  Prefix listsWe have to create two network lists, the first one for T-mobile and the second one for Vodafone.  T-mobile network list:We configure the normalisation of Called party number in the Table of replaced prefixes (the number in front of the slash mark is replaced by number behind the slash mark. If there is not any number in front of the slash mark, it equals to “everything”).We also have to fill in the Table of prefixes with all prefixes of the T-mobile operator. Because all numbers are 9 digits length, it is not necessary to specify it for each prefix, we can use the parameter Default count of digits and fill there the value 9.For the match in prefix list, the Table of replaced prefixes, Table of accepted prefixes and Digits count must correspond with the called number.  Vodafone network list:We configure the normalisation of Called party number in the Table of replaced prefixes (the number in front of the slash mark is replaced by number behind the slash mark. If there is not any number in front of the slash mark, it equals to “everything”). We also have to fill in the Table of prefixes with all prefixes of the Vodafone operator. Because all numbers are 9 digits length, it is not necessary to specify it for each prefix, we can use the parameter Default count of digits and fill there the value 9.   LCR TableWe have to configure rows in the LCR table where we bind together the incoming way (GSM group or VoIP) with the outgoing way (specific GSM outgoing group or VoIP) by Prefix lists.  Click on ADD button (or edit the second line of LCR table) to add another line and configure it the following way for the T-mobile: Click on ADD button to add new line and configure it for the Vodafone.  Now you can see following lines in the LCR table:   Incoming calls  For incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.  In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.  With mode Accept…, you can define the list of called numbers which will be automatically dialled after DTMF dialling timeout in case the customer does not press any button within the specified time. From the configuration, you can see 5 seconds for DTMF dialling and after that the call will be routed to the extension 100 in your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters). If you want immediate connection set DTMF dialling timeout to 0.    SIM cards  In the Gateway control and module control you can see the status of the SIMs. The recommendation for quality of the signal is between -65 and -85dBm.  The logged SIM card is indicated by switched off the led diode on the 2N VoiceBlue Next.

Question 35

Configuration Tool Download

Accepted Answer

Compatibility   The tool might give false reports about compatibility. Try to load and save parameters and you will see if some parameter can not be load or save. It might happen that 1 or 2 parameters are not saved but these parameters are usually not important.      Downloads: 2N VoiceBlue Next Configuration Tool 2.0y USB Drivers Manual Connect via USB port, COM port or IP, see Communication settings:

Question 36

Connection via HyperTerminal - How can I connect to VoiceBlue Lite using HyperTerminal?

Accepted Answer

The HyperTerminal program is included in Windows XP. You can find it in menu Start > All programs > Accessories > Communication. For connection to the gateway using COM port set following parameters:  Bitrate - 57600  Flow control - none  Rest of values will stay default  Second option is to use the TCP/IP connection. When you use TCP/IP you have to choose port 23 for communication with gateway.

Question 37

Credit check - How can I recognize an amount of credit on my SIM card?

Accepted Answer

Credit check - How can I recognize an amount of credit on my SIM card?   Go to the terminal and use this AT command to get an amount  AT&gxx=xtd*101#  and then after # you should use ; or without ; it depends on module. xx=is number of GSM module 00 means the first module.

Question 38

Credit - How can I check balance on my SIM card? (USSD commands usage)

Accepted Answer

Go to the Putty, HyperTerminal or other terminal supporting telnet protocol. Connect to 2N VoiceBlue Next IP on port 23 and fill login data. Finally use these commands: AT&gXX=xtdUSSD or AT&gXX=xtdUSSD; (semicolomn at the end is required by some GSM/UMTS modules) XX is number of GSM module counting from 0USSD is the operator code for getting of credit Example: USSD sequence *122# sent over second module: at&g01=xtd*122#g01 OK-->g01 +CUSD: 2,"YOUR CREDIT IS 22 EUR VALID UNTIL 18-12-2010.",15 Other GSM/UMTS module type: at&g01=xtd*122#;g01 OK-->g01 +CUSD: 2,"YOUR CREDIT IS 22 EUR VALID UNTIL 18-12-2010.",15 If you receive CUSD 4 replz instead of a text reply, it usually means  wrong USSD code is used  service is not allowed for used SIM card  service is temporary out of order in mobile network or in part of the network

Question 39

CUCM - How to interconnect with CISCO Call Manager 6, 7, 8?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.22.42  Incoming port: 5060  Firmware: 01.00.04   Cisco Call Manager parameters:  IP address 192.168.22.35  Incoming port: 5063  Firmware: CUCM 8.0    Scenario  If we have an IP network in which an Cisco Call Manager, several SIP phones and 2N VoiceBlue Next are connected, the configuration would be as shown in the figure below. Furthermore, suppose that the network is addressed as shown in the figure and GSM numbers are all numbers starting with 6, 7 and containing 9 digits.     2N VoiceBlue Next settings  SIP trunk interconnectionFor the setting of the trunk between the 2N VoiceBlue Next and your CUCM, you need to configure "SIP proxy (GSM→IP)" for GSM incoming calls. "SIP proxy (IP→GSM)" is designed only for secure communication with the traffic from your CUCM. You can specify the IP address and port where the IP packets will be accepted.  Configuration of the LCR (Least Cost Routing)You have to specify prefixes for the operators in the country you are currently located. An example of this would be that in Czech Republic prefixes 6 and 7 have a 9-digit number.The setting is displayed below.You need to create specific guidelines connecting prefixes with the GSM group. In the "GSM group" you will specify settings for SIM cards assigned to this specific group. In the "GSM group assignment" you can assign the module for the appropriate GSM outgoing group.     Configuration of GSM outgoing groups:You are able to have different setting for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others)  Incoming callsFor incoming calls you can define 2 groups with the different behaviours and assign them to the GSM modules. The settings are similar with "GSM groups assignment" for outgoing calls. In GSM incoming groups you can specify the traits for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback. You can define the list of numbers called. The number will be automatically dialled after the DTMF dialling has timed out. This happens when the customer doesn't press any button until the specific time. At this point, the number will be routed to the extension 100 to your Cisco Call Manager (if you set up SIP proxy (GSM->IP) in VoIP parameters).    Cisco Call Manager settings  Create a new trunkAdd new trunk in the menu Device → Trunk → Add newYou need to set up Trunk Type: SIP Trunk and Device Protocol: SIP Set up the trunk as in the picture below Calling Search Space is setting for incoming call to CCM. In the Test Team is set up which numbers are allowed to be called.AAR (Alternative Call Routing) is not necessary to set up. It depends on the customer´s requirements. Destination address is the IP address of the 2N VoiceBlue Next.SIP Trunk Security Profile have to be set as: Non Secure SIP Trunk Profile and Outgoing Transport type must be set up for UDP communication     Route pattern settings Enter the menu Call routing → Route/Hunt → Route Pattern → Select gatewayYou can use a route list and groups but if you intend to use only one gateway connection, then you do not have to do it.You can simply set up rules in Route pattern.Examples of Route pattern:724! means prefix 724 and all other digits after724XXXXXX means prefix 724 and 6 more digits0.6! means that dialled 0 will be separated and 6 with other digits can be dialled. In the Called Party Transformation must be set up PreDot for the stripping of the 0. If you make any change, you need to save it and then apply changes!!

Question 40

Data transfer - Does 2N VoiceBlue Next gateway support data transfer?

Accepted Answer

2N VoiceBlue Next gateway supports voice calls, sending/receiving SMS messages but it does not support data transfer.You can use our product 2N OfficeRoute.

Question 41

Do 2N VoiceBlue Lite gateways support data transfer?

Accepted Answer

No, they don't.

Question 42

Elastix PBX - How to interconnect with Elastix PBX?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.50.45  Incoming port: 5060  Firmware: 01.00.04   Elastix PBX:  IP address 192.168.50.115  Incoming port: 5060  Firmware Elastix: 2.0.0  Firmware Asterisk: 1.6.2.13    Scenario  If we have an IP network in which an Elastix PBX, several SIP phones and 2N VoiceBlue Next are connected, the configuration would be as shown in the figure below. Furthermore, suppose that the network is addressed as shown in the figure and GSM numbers are all numbers starting with 6, 7 and containing 9 digits.     2N VoiceBlue Next settings  SIP trunk interconnectionFor the setting of the trunk between the 2N VoiceBlue Next and your Elastix PBX, you need to configure "SIP proxy (GSM→IP)" for GSM incoming calls. "SIP proxy (IP→GSM)" is designed only for secure communication with the traffic from your Elastix PBX. You can specify the IP address and port where the IP packets will be accepted.  Configuration of the LCR (Least Cost Routing)You have to specify prefixes for the operators in the country you are currently located. An example of this would be that in Czech Republic prefixes 6 and 7 have 9-digit numbers.The setting is displayed below.You need to create specific guidelines connecting prefixes with the GSM group. In the "GSM group" you will specify settings for SIM cards assigned to this specific group. In the "GSM group assignment" you can assign the module for the appropriate GSM outgoing group.     Configuration of GSM outgoing groupsYou are able to have different setting for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others).  Incoming callsFor incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with "GSM groups assignment" for outgoing calls. In GSM incoming groups you can specify the traits for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback. You can define the list of numbers called. The number will be automatically dialled after the DTMF dialling has timed out. This happens when the customer does not press button within the specific time. At this point the number will be routed to the extension 100 to your Elastix PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).    Elastix PBX settings    Create an extensionIn the "PBX Configuration" and "Extensions" you create an extension as in the print screens below.   You have to define type as "friend" and listening port, e.g. 5060 as in the example.  Set up the routeAdd a new route in the section "Outbound Routes". In the example the route is called VoiceBlue. If you set up Dial Patterns "0|." it means that you have to make an outbound call via prefix 0.  Set up the trunkFill up the „Trunk description", „Outbound Caller ID" for the outbound identification.You can limit maximum VoIP channels via dedicated trunk in the menu. Also if you want to send SIP OPTION packets command regularly to check whether the device is still online, turn on the parameter „qualify" as yes.  Incoming callsYou can route Incoming calls to the IVR. You find the setting in the section „IVR".   In the „General Settings", you can allow an anonymous incoming calls as in the picture below.

Question 43

Factory reset - How to make a factory reset VBM EN

Accepted Answer

Factory reset - How to make a factory reset VBM EN  If you need to setup your device to the factory values, please proceed following instructions. This action is irreversible. The factory values and credentials are listed at the end of this document.  Factory reset via web interface:  Go to the section Management  In the right column, click on the Configuration reset  Select parameters which will be set to default values        Save settings       Reset to factory settings using a button:  To reset the factory values, press the Reset button until the LED indicators start flashing red.    Note: If you forget the password or set the IP interface incorrectly, you can press the RESET button to the right of the power supply connector for a long time. Doing this, you restore the factory default configuration values for all parameters, including the Ethernet interface parameters and access data.       Default Settings: IP address: 192.168.1.2 IP mask: 255.255.255.0 IP gateway: 192.168.1.1 Username: Admin Password: 2n

Question 44

Factory reset - How to make a factory reset VBN

Accepted Answer

If you need to setup your device to the factory values, please proceed following instructions. This action is irreversible. The factory values and credentials are listed at the end of this document.  Factory reset via web interface:  Go to the section Management  In the right column, click on the Configuration reset  Select parameters which will be set to default values        Save settings        Reset to factory settings using a button:  To reset the factory values, press the Reset button until the LED indicators start flashing red.    Note: If you forget the password or set the IP interface incorrectly, you can press the RESET button to the right of the power supply connector for a long time. Doing this, you restore the factory default configuration values for all parameters, including the Ethernet interface parameters and access data.       Default Settings: IP address: 192.168.1.2 IP mask: 255.255.255.0 IP gateway: 192.168.1.1 Username: Admin Password: 2n

Question 45

Firmware 2N VoiceBlue Lite - How can I upgrade firmware?

Accepted Answer

You need to use serial or USB cable. Then connect your gateway with configuration tool, go to the firmware/license menu. There choose your firmware file and click „Upgrade Firmware"

Question 46

Firmware - Can I use firmware 03.00.xx?

Accepted Answer

Yes, you can. This firmware is used for the SimStar server communication but there is no limitation to use this firmware in the stand alone VoiceBlue Lite gateway.

Question 47

Firmware Upgrade - How to make a firmware upgrade on 2N VoiceBlue Next?

Accepted Answer

Visit our website www.2n.com and download FW for 2N VoiceBlue Next  Open web interface of your GW and go to the section Management->Firmware update. Select the FW file you downloaded in the previous step and upload it to the GW.   GW will reboot, after reboot you can check in the same section whether the update was successful as you can see the FW version there.

Question 48

HiPath 3000 - How to interconnect with Siemens HiPath 3000?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.1.120  Incoming port: 5060   Siemens HiPath 3000 parameters:  IP address 192.168.1.50  Incoming port: 5060    2N VoiceBlue Next settings  SIP trunk interconnectionFor the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for secure communication only with traffic from your PBX. You can specify the IP address and port from which SIP packets will be accepted.In case you leave there 0.0.0.0 it will be open for all traffic.  Configuration of the LCR (Least Cost Routing)The GSM operator has e.g. in our country prefix 6 and 7 with a 9-digit number. The setting is below. You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.     Configuration of GSM outgoing groupsYou are able to set up different settings for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others). In case you do not have a Ring back tone, set up Delay for ALERTING to option 4.  Incoming callsFor incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls. In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback. You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button within the specified time. From the configuration, you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 101 to your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).    SIEMENS HiPath 3000 version 8.0  Create a new IP trunkAdd a new trunk - Trunks → IP TrunksNumber of trunks = number VoIP channelsYou need to assign IP trunk to Trunk group 16 (always for IP trunk). This trunk is called interwork in the picture below. The setting of Trunk group 16 is in the picture below.Important is to use en-block setting for sending of dialled number.  LCR setting in PBXEnter the menu "Least cost routing" → "Dial plan"Example setting of Dialled digits:0C6Z means: 0... prefix for outgoing calls from PBXC... user get dial tone (morse A)6... prefix to GSM networkZ... unlimited number of digitsYou have to send the prefix to Route table (in our example Route table 3) Choose your Route table and press "Dial rule wizard". Now you are able to set up Dial rule format A. It means that it will repeat all digits after C (0 will be cut off from called number).  Setting of VoIP card - via web interface (HG 1500 V.8.0)First of all you need to have licenses for VoIP channels (2 channels should be open as a standard)Enter the menu: Explorers → Voice Gateway → PBX → NodesNode 1 needs to be configured for incoming traffic from 2N VoiceBlue gateway. This setting is for routing to your own system.LAN trunking protocol needs to be "Native SIP" and IP address is the IP of the Siemens HiPath 300.  Node 2 needs to be configured for outgoing traffic to 2N VoiceBlue gateway.LAN trunking protocol needs to be "Native SIP" and IP address is the IP of 2N VoiceBlue gateway (192.168.1.120) Siemens HiPath can check the connection with the gateway by setting of "Alive monitoring". You are able to set it up in Node setting and you can choose PING or TCP IP monitoring. Both methods are supported by 2N VoiceBlue Next gateway.  RoutingNow you have to set up routing digits to your predefined Nodes.Example of setting:Number 6 and 7 are routed to the Node 2. There is a gateway 2N VoiceBlue NextNumber 1 is routed to the Node 1. This node is for own Siemens HiPath PBX.  Codec settingEnter the menu Voice Gateway → Codec ParametersYou can set up priorities for codecs.The Siemens HiPath support DTMF via RFC 2833.  Incoming call from 2N VoiceBlue NextIn the VoIP card we have already set up routing for prefix "1" to own Siemens HiPath PBX. Currently the number coming from 2N VoiceBlue Next is compared with DID number and routed to the particular phone.

Question 49

How to configure sending of SMS messages via SMPP feature on 2N gateways

Accepted Answer

In this FAQ you will see how to configure sending of SMS messages via SMPP feature on 2N gateways. First of all you can read a brief introduction how do these features work and after that there are several important steps which have to be met for proper settings of this feature:  Set the GSM gateway to send/receive SMS via SMPP  Define the routing rules for incoming and outgoing SMS messages  Check the GSM gateway SMS database´s filling capacity  This FAQ applies for following gateways:  2N BRI Lite, 2N BRI Enterprise (2 channels)  2N VoiceBlue Next (2 or 4 channels)  2N VoiceBlue MAX (4 channels)  2N BlueTower (2 to 8 channels), 2N StarGate (2 to 32 channels).   Requirements Note: As a very first step please make sure that the license for SMPP feature is uploaded into your 2N gateway. This license is sold as a package for 10 users (you can buy licence for 10, 20, 30, etc. users) and you can have up to 200 licensed users in the gateway. If you want to verify that you have uploaded the correct license in your gateway, go section "Management -> License key". You should see a licenses called SMSS and SMSUx in the "Active" row. SMSS indicates that SMPP feature is enabled and SMSUx indicates number of licensed users where x stands for this number (see the picture below). If there will not be a license for SMPP (SMSS) then you will not see tab "Messaging" on the top bar!    Gateway configuration  Function enabling and modules grouping In the first step go to section "Messaging –> Modules grouping" and enable SMS server – check the appropriate check-box. Next add GSM modules to SMS groups in order to facilitate incoming/outgoing SMS routing. It can be done based on your requirements e.g. according to the GSM provider, etc. The last step in this section is to assign SMS groups created in the previous step to the User groups if you want to use messaging with different routing rules (e.g. to different company departments). The GSM gateway can contain up to 32 User groups and up to 32 SMS groups (4 in case of smaller gateways - BRI, VoiceBlue Next, VoiceBlue Max) Note: in this section you can also enable AutoCLIP routing for SMS. Set the maximum time for which the automatic SMS routing record (response) will be saved in the AutoCLIP table. It means that once you send an email to some specific GSM number and after some time the mobile user responds to this message and sends a SMS message to SIM card inserted in the GSM module from which the email was previously sent, then this message will be automatically resent directly to the user who sent an email. Note: newly you can also specify the message expire time. In case, that SMS message is not sent, in case it is received but not read or in case that DLR message has been received and not read you can specify after how long these SMS messages should be erased from gateway database.    Add user As a next step you need to skip to the section "Messaging –> Users", click on the "Add" button and create new SMS user(s) who will send and receive SMS messages. There are several parameters in this section which have to be set.  User group: assign the user to User group  Access type: Select the communication protocol for SMS processing as SMPP (SMSC)  Login and password: Define the Login name and Password under which the SMPP server will log into the gateway  Tcp type: Set parameter as Server  Message direction of Esme: Set this parameter as Transceiver  You can see all parameters from this section in the picture below where was created user "Logica" with the login "logica" and password "logica". Once you fill in all parameters, click on the "OK" button. The user will be added and after that you have to make one more click to the "Save settings" button in the bottom right corner of the page (blue circle label in the picture).    Incoming SMS routing Now specify the routing rules for incoming SMS messages – this is done in the "Messaging –> User Group –> Incoming SMS routing" menu where you click on the "Add" button. There are several parameters which can be set or modified (see the picture below). Three most important ones are:  Message from SMS group – select the SMS group that receives SMS messages  Action type – specify how incoming messages will be processed. You can set AutoCLIP routing (the incoming SMS will be routed according to the AutoCLIP table records), Deliver to user (the SMS will be delivered to the selected user) or Drop message (the SMS will be dropped).  Deliver to user – this field is used only if you selected “Deliver to user” in the previous step. Here is defined the user who receives a SMS.  Note: The rules are applied for incoming SMS sequentially, starting from the first rule. If all the set rules are met in a row, the SMS is routed according to the row.  SMS routed directly to one user accountIn the picture below you can see that each message which is received on GSM modules assigned to SMS group 0 (in our case these are all the GSM modules) are routed directly to user "Logica". It means that once you send SMS to SIM cards inserted in the gateway, this SMS will be converted to an SMPP message and sent directly to the SMPP server that is saved under this account. You can also specify the time and day when this routing rule is active. Once you finish the configuration click on the "OK" button and after that on the "Save settings" button.  SMS routed based on the AutoCLIP routing tableSometimes it is not desirable to send all received SMS messages to one user but you can require to send SMS messages based on the AutoCLIP routing table records. Only if there is no record in this table for the incoming message then you can forward the message to one specific user. The setting is shown below – firstly 2N gateway tries to deliver SMS message based on the AutoCLIP routing and if there is no record for incoming GSM number then the SMS message is sent to Receptionist´s email account.     Outgoing SMS routing As a last step you need to specify routing rules for outgoing SMS – it is done in the "Messaging –> User Group –> Outgoing SMS routing" menu where you click on the "Add" button. In this section you will specify mainly these parameters:  Message to SMS group – the SMS group via which the outgoing SMS will be routed  Action type – define how to process outgoing SMS. You can choose from two options: Submit from user (SMS can be sent only from selected users) or Submit from all users (enable sending SMS from all the SMS group users).  Submit from user – choose a user who can send SMS (this is used only if "Submit from user" was selected)  Note: do not forget to enable the SMS server as shown in the first step (there is a check-box)! In the picture below there is shown settings for User Group 0. Messages from all users in this User group are routed via SMS group 0. It means that these messages will be transferred to SMS and sent only via GSM modules assigned to SMS group 0. It is all GSM modules in our example (as you can see in the picture from step 1).     Settings of Logica SMPP client Download Logica "Java test application" from the website Logica OpenSMPPOpen the Windows Terminal and get into the file folder where is saved smpp.jarWrite to the terminal this command for executing smpp.jary java -cp smpp.jar;smpptest.jar com.logica.smpp.test.SMPPTest if you get the message: " 'java' is not recognised as an internal or external command, operable program or batch file", it means that you need to put java to your path in the window´s commands. Follow this online guide how to do it:       4. You will get to the settings menu. Press "1" to bind client with the gateway and follow the settings on the picture          java -cp smpp.jar;smpptest.jar com.logica.smpp.test.SMPPTest          5. Press "2" for submit and follow settings on the picture        6. After that press Enter and you will send a SMS over the gateway

Question 50

How to provide all information to effectively resolve a technical issue on the 2N VoiceBlue Next?

Accepted Answer

Dear customers, In case you have found some issues with the device, please report it to https://support.2n.cz/. If you do not have access information, contact your account manager. Reported issue should contain following information:  Detailed description of the issue -  Please describe to us your problem in detail, so that we will be able to replicate the behaviour based on that description.    Backup of your configuration -  Send us the current configuration of the device – you can download device's current configuration from device's web interface  (menu Management - Configuration download - Save to disk)   The current firmware version you use and serial number of device.  Gateway's internal trace of described issue according to following FAQ Article.  Network trace (Wireshark) of described issue (for VOIP communication, SMPP communication, EMAIL2SMS communication, etc....) according to following FAQ Article.  Optional: For more complex scenarios, please also attach network topology diagram of LAN (for VOIP communication, SMPP communication, EMAIL2SMS communication, etc....).  All those information will help us to resolve your issue and provide you our feedback much faster. Thank you for your understanding.

Question 51

Innovaphone PBX (IP302) - How to interconnect with Innovaphone PBX (IP302)?

Accepted Answer

2N VoiceBlue Next has these parameters:  IP address 192.168.22.42  Incoming port: 5060   Innovaphone PBX parameters:  IP address 192.168.22.227  Incoming port: 5060    2N VoiceBlue Next settings  SIP trunk interconnectionFor the settings of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed for secure communication just with traffic from your PBX. You can specify the IP address and from which SIP packets will be accepted.In case you leave there 0.0.0.0 it will be open for all traffic.  Configuration of the LCR (Least Cost Routing)The GSM operator has e.g. in our country prefix 7 and 8 with a 9-digit number. The setting is below. You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.     Configuration of GSM outgoing groupsYou are able to set up different setting for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others)  Incoming callsFor incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar to GSM groups assignment for outgoing calls. In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback. You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button within the specified time. From the configuration you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 100 to your PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).    Innovaphone PBX settings    You need to set up Domain and Address for the IP address of the 2N VoiceBlue Next. Communication protocol is SIP with the Mode: "Gateway without registration).  Then you need to set up prefixes (routing table) which will be routed to the 2N VoiceBlue Next. In the example there is defined prefix 7 only.  In the picture below there is the setting for incoming calls to Innovaphone PBX. The call is directed to the phone 1.  In the picture below there is the complete setting for the Innovaphone PBX routing.

Question 52

IP setting - How can I change the IP address of 2N VoiceBlue Lite?

Accepted Answer

You need to use serial or USB cable. It is not possible to change it remotely via TCP/IP.

Question 53

IP setting - How can I use NAT keep alive?

Accepted Answer

You can set lower time for registration expire timeout. Then you will keep your NAT tables actual with Registration requests.

Question 54

IP setting - How to find out the IP address?

Accepted Answer

The default IP is 192.168.1.2 and DHCP client is off. There are two ways how to find the modified IP. You can use the 2N software IP Scanner in the network with your 2N VoiceBlue Next and scan the network. You can download this software from our web page www.2n.cz. IP Scanner searches the network with higher priority in your computer. If you are connected to more networks with your computer, stay connected just to the network where the 2N VoiceBlue Next is located.  In case you cannot find 2N VoiceBlue Next with IP Scanner then the gateway is probably situated in different network than your computer. In such case please use the Wireshark software. Start the capture on network interface with VoiceBlue Next and filter ARP packets. You can wait for the packet but it is usually quicker to disconnect and connect Ethernet cable on 2N VoiceBlue Next side or switch side. The packet Source MAC address is marked with our ID 2nTeleko.

Question 55

License - How can I use license key?

Accepted Answer

When you obtain license key for your gateway go to the menu  Firmware/License  . There you can see also status of actual license but when you want to upload a new, please fill the license key into "license key for the gateway" field.

Question 56

Licenses on VoiceBlue Next - Detailed information

Accepted Answer

This FAQ document describes principles of particular licenses and what are they determined for. Information about licenses in your devices you can find in Management section, picture bellow.   ISDN Line SMS monitoring (MONI) This license allows you to get information about ISDN lines immediately via SMS, when they changed their status. You can configure 2 types of information SMS. You can specified the body of the message in both of them. First one is, when the ISDN line change the state, e.g. disconnect. You can set parameter how long needs to take this change before SMS is sent. Second type is kind of periodical check status message. The gateway will send SMS after some time period without changing the ISDN line status.   Mobility Extension users (MEx) This feature allows any user to add GSM phone number next to SIP or ISDN extension. The gateway will routed the call normally to users´ fixed station and also to his GSM extension. You are able to pick up incoming call on any of this stations. Because of this function you don´t miss any important call. The letter 'x' in the shortcuts name refers to the number of users. The maximum amount of Mobility extension users is 56.   VoIP (SIP) This licenses allow you to provide calls via Voice over IP technology. You can register the gateway to the SIP Proxy server or you can easily connect single VoIP phone. 2N GSM/UMTS gateways support large amount of codecs and can be configured by wide variety of parameters. Both call directions supported, GSM –> VoIP and VoIP --> GSM. More information you can find in manual.   Where may I find it?       SMPP users (SMSS) This license allows you to send SMS via 2N GSM/UMTS gateways directly from your computer or some server. SMPP (Short Message Peer to Peer) is the open telecommunication protocol enables sending SMS between SMS center and the external applications called ESME (External Short Messaging Entity). This ESME application allow you to send and receive SMS similar way as on your mobile phone. You can have up to 16 SMPP users.   Email2SMS users (SMSE) Second license related to SMS is Email2SMS license. This license allow you to use any E-mail client on your computer to sending SMS via 2N GSM/UMTS gateways. It is very easy to use because our gateways provide their own SMTP (Simple Mail Transfer Protocol). Our devices can also send SMS to e-mail. You can download it via POP3 (Post Office Protocol version 3). After all settings is done, you need to create new account in your e-mail client and now you are able to send SMS as easy as any classic e-mail message. The maximum number of users is 200, included SMPP users. For more information please visit following FAQ pages - faq.2n.com where can you find for example: SMS messages - How to set SMS2Email and Email2SMS on 2N®StarGate with CPU M202 version (2013+) with Microsoft Outlook?   SNMP monitoring (SNMP) SNMP (Simple Network Management Protocol) is a protocol determined for managing devices connected to IP network. Main task of this protocol is monitoring and managing devices over network via predefined parameters. Particular parameters are usually specified by a developer into MIB table. Whole communication take place between to side. One side called manager, this is master devices that collect data from particular devices and is able to set parameter at the monitored side. There is SNMP agent, running directly in the device that provide communication with the manager side.

Question 57

License status: Locked

Accepted Answer

If you can see status "Locked" in device´s website - Management - Licence key ...  ... than please follow this steps to make it working again   1) Please upgrade the firmware to the latest version. 2) Please set the right date and time in the device. 3) Please upload the license file again. Try to download it via another email client, do not open it, just upload. 4) Eventually please make a factory reset.

Question 58

Mobility Extension - How can I configure Mobility Extension?

Accepted Answer

In case you want to use this feature you need to  cooperate with another SIP proxy  . There you have got registered users. When you want to create ME for one of these users go to the ME menu in  2N VoiceBlue Lite  . There create the same user with the same authentication parameters like in SIP proxy. Of course you have to fill GSM number as well. For proper registration this user from  2N VoiceBlue Lite  into SIP proxy you need to register whole  2N VoiceBlue Lite  in the same SIP proxy. You need to fill name, user name and password in SIP parameters menu and also SIP registration field in the same menu (with the address of SIP proxy).

Question 59

Mobility Extension - How to set Mobility Extension on VoiceBlue Next?

Accepted Answer

If you would like to use Mobility Extension feature you have to cooperate with another SIP proxy where are the SIP users registered. When you want to create ME station for one of these users, follow steps below.  You need a proper license for the Mobility Extension feature.Go to the menu: Gateway control - Licence - Active and take a look if there is ME8 (16, 24, 32 etc.) license for 8 (16, 24, 32 etc.) users available. If you have not received any license, please contact our account manager.  The 2N VoiceBlue Next gateway has to be registered to the same existing SIP proxy (domain).Go to: Gateway configuration - VoIP parameters - SIP registration and fill the registration details (see example below).     Now add the Mobility Extension users (which are already created in your SIP proxy) and assign them the mobile phone numbers.Go to: Gateway configuration - Mobility Extension - Add user and fill the user details (see picture below).   "Follow me" function enables the forwarding of the call to the mobile number (for ex. 777112233) when someone calls the SIP user (100 in this example).  Finally make sure the phone number (prefix) of the SIP proxy user (100) matches with the prefixes in the prefix list.Go to: Gateway configuration - Prefixes - Table of accepted prefixes

Question 60

Network Trace - How to get Wireshark network trace and network trace from 2N VoiceBlue Next?

Accepted Answer

Network trace contains whole communication between IP devices over LAN. In order to analyse what this communication between devices contains, a network trace can be requested. If network trace is requested, please note there are two possible network traces available. Gateway can provide network tracing with limited size on its side, for unlimited network trace of IP communication, external device (usually laptop or PC) equipped with network capture software needs to be used. This article explains how to perform both network trace in such software called Wireshark as well as network trace directly from gateway.  How to get Wireshark network trace?  Program Wireshark is used for tracing the communication between devices interconnected by LAN. The devices (e.g. PBX connected with a GSM gateway via LAN) send to each other packets which are captured by the above mentioned program. Wireshark is distributed under the Open source licence (http://www.wireshark.org/download.html).  Before starting network capture  Before Wireshark (or in general, any packet capture tool) is used, careful consideration should be given to where in the network packets are to be captured. In order to capture all the packets which are sent via particular LAN, the devices have to be part of the same network segment (they have to be connected via HUB). If you do not have a hub you can use a switch which supports so called port mirroring (http://en.wikipedia.org/wiki/Port_mirroring).   Download Wireshark  Obtain a Wireshark package or installer for the operating system running on the system which is to be used for packet capture.Download a binary or installer from http://www.wireshark.org. With installers, ensure all product components are selected for installation.   Start Wireshark On a Linux or Unix environment, select the Wireshark or Ethereal entry in the desktop environment's menu, or run "wireshark" (or "ethereal") from a root shell in a terminal emulator. In a Microsoft Windows environment, launch wireshark.exe from program installation folder. Note that on Unix systems, a non-GUI version of Wireshark called "tshark" (or "tethereal") may be available as well, but its use is beyond the scope of this article.   Use Wireshark (capture network communication) After starting Wireshark, do the following:  Select Capture | Interfaces  Select the interface on which packets need to be captured.  Now click the Start button to start the capture.  Recreate the issue. The capture dialog should show the number of packets increasing. If packets are not being captured, try removing any filters that have been defined.   Important: In order to provide us with detailed information about the possible problem with the device, please do not apply any filter and simply capture all the communication during the test (call/smpp/email2sms,etc...).  Once the issue which is to be analysed has been reproduced, click on Stop. It might take a few seconds for Wireshark to display the packets captured.  Save the packet trace in following format: „File->Save As->Wireshark/tcpdump/… -libpcap(*.pcap;*.cap) .Just click on the File menu option and select Save As.By default Wireshark will save the packet trace in libpcap format. This is a filename with a.pcap extension.   Please send saved file to us for analysis.     How to get network trace from 2N® VoiceBlue Next?  Network trace from gateway can be run via menu Utils -> Network capture. The web interface can store up to 4096kB long trace.  Network trace can be started by clicking on button START. You will be notified by message "Running..." in capturing status. The Buffer utilisation and Packets captured should show the number of packets increasing. If not, then restart the capture. If Buffer utilisation is on value 4096kB (same as Buffer size), please restart the trace and tests from beginning.  Recreate the issue.  Once the issue which is to be analysed has been reproduced, click on Stop. Once trace is stopped, you can save it to your computer by clicking on the diskette icon located in bottom right corner of web interface  Please send saved file to us for analysis.

Question 61

No tones - I can't hear any tones on 2N VoiceBlue Lite when setting up a call. What should I do?

Accepted Answer

Go to GSM outgoing group menu and set "Delay for CONNECT"=0 and "Delay for ALERTING"=2. And also select Dial and Ring tones in the GSM basic parameters menu

Question 62

panasonic-ncp-how-to-interconnect-with-panasonic-ncp-pbx

Accepted Answer

2N VoiceBlue Next Parameters  IP address: 192.168.88.2  Port: 5060  Firmware: 03.00.03rc3   PBX Panasonic KX-NCP500VNE  IP address: 192.168.88.101  IP DSP: 192.168.88.102  Port: 5060  Port Invite: 5060  Firmware: 003-000    2N VoiceBlue Next settings  SIP trunk interconnectionFor the setting of the trunk between the 2N VoiceBlue Next and your PBX you need to configure SIP proxy (GSM→IP) for GSM incoming calls. SIP proxy (IP→GSM) is designed to secure communication just with traffic from your Panasonic PBX. You can specify the IP address and port from which SIP packets will be accepted.In case you leave there 0.0.0.0 it will be open for all traffic.  Configuration of the LCR (Least Cost Routing)The GSM operator has e.g. in our country prefix 7 and 8 with a 9-digit number. The setting is below.You need to create LCR rule for defined prefixes. The GSM group defines a way for the outgoing call routing. An appropriate SIM card is selected based on the GSM groups assignment.   Configuration of GSM outgoing groupsYou are able to set up different setting for each GSM group (CLIR, free minutes, virtual ring tone, roaming and others)  Incoming callsFor incoming calls you can define 2 groups with the different behaviour and assign them to the GSM modules. The settings are similar with GSM groups assignment for outgoing calls.In GSM incoming groups you can define the behaviour for each GSM incoming group. Choose the mode to Reject, Ignore, Accept incoming calls or Callback.You can define the list of called numbers which will be automatically dialled after DTMF dialling timeout if the customer does not press any button within the specified time. From the configuration you can see 10 seconds for DTMF dialling and after that the call will be routed to the extension 100 to your Panasonic PBX (if you set up SIP proxy (GSM->IP) in VoIP parameters).    Panasonic NCP settings (by Masscomm)  Connection settingsFirst of all we need to create a new slot as SIP Trunk (SIP Gateway Virtual type).By default Panasonic NCP has the port 35060 opened to receive INVITE messages from the other side. We have to switch Virtual rack and V-SIPGW16 card to mode "OUS". Then we have to change the parameter in the virtual rack and card Properties of the rack->Virtual SIP Gateway (Trunk) Number of port of SIP Value: 5060. (Default value 35060) After this change, the PBX must be restarted!  SIP Trunk settingsThe port which is going to be used has to be set as OUS.In the main tab fill the following parameters:Channel Attribute=Basic ChannelName=VoiceBlueIP address=192.168.88.2 (in the example)In tab Account the fields Name, Authentication ID, password, should be filled. In our case we set all parameters as VoiceBlueWe go to tab REGISTRAR and we have to set REGISTRAR CAPACITY as DISABLED. After that we will configure the server to Registrar Name: VoiceBlue; IP= 192.168.88.2. After having followed all previous steps, the port must be set in mode INS (It should remain marked). Now we are able to do the testing of the interconnection between both devices.

Question 63

Setting up a call between 2N LiftIP / Lift8 with VoIP module and 2N VoiceBlue Next gateway

Accepted Answer

This article describes how to set up a call between 2N LiftIP / Lift8 with VoIP module and 2N VoiceBlue Next gateway.    Setting of 2N LiftIP / Lift8 with VoIP module   Connect the device 2N LiftIP - more here: Calling between 2N LiftIP and IP Phone  or 2N Lift8 with VoIP module - more here: Calling between 2N Lift8 and IP Phone via VoIP module  Parameter 011, enter the number (address) of the station to be dialed after pressing the alarm button on the elevator communicator - see e.g. local IP phoneat 192.168.50.156 in the picture below:Parameters 012-016 allow you to configure other destinations to be dialed in case of unavailability of the previous one - see e.g. public telephone station and mobile number for parameters 012 and 013 in the picture below:     Parameters 1105, 1113 a 1114 fill in the IP address of the 2N VoiceBlue Next gateway (see IP setting - How to find out the IP address?).  Parameters 1106, 1111 a 1112 enter e.g. 101 and don't use the parameter 1107 (password) - as shown in the picture below.     Be sure to click the "Save to Device" icon below to save changes.    Setting up the 2N VoiceBlue Next    Connect the device 2N VoiceBlue Next - more here: Configuration - How to set 2N VoiceBlue Next?  Adjust the parameter SIP proxy (GSM->IP) in the section VoIP parameters (see the picture below), fill in the IP address of the 2N LiftIP / VoIP module of the 2N Lift8,and then click on the pencil icon below to save the changes.      By dialing the number of the active SIM card in the 2N VoiceBlue Next gateway, you automatically call the 2N LiftIP / Lift8 device with a VoIP module.

Question 64

SIM card logging - What to check if SIM cards do not log into network?

Accepted Answer

Please check that  Antenna is correctly connected  PIN is deactivated or right PIN code is provided in Gateway configuration/System parameters  Wavecom has right frequencies set in Gateway configuration/GSM Basic parameters/Multi-band selection. You can check frequencies used in your country on Wikipedia (This applies only for gateways with order numbers starting 505105.. You can find order number on label on the back side of gateway)  Gateway configuration/GSM Outgoing groups/Roaming enabled for network code is set in case of SIM from foreign country or in case of SIM of virtual operator who uses infrastructure of the different operator. By default the gateway allows SIM to log only to home network. Code of SIM and code of network must be the same. This parameter allows usage of other network than the home network. The network is specified by 5-digit code. You can find these codes on Wikipedia. X is used as wild character. For example 23001 allows SIM to log to T-mobile Czech Republic and 230XX allows SIM to log to any network in Czech Republic. This parameter do not show which network should be used. The offer to accept SIM comes from the network and the gateway can allow logging or reject the offer.  There is no time restriction set in Gateway configuration/GSM Outgoing groups. By default SIM is allowed for the whole week from 00:00 till 24:00.  SIMs work in cell phone.  Cell phone shows good signal level.

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