2N OfficeRoute & Alcatel OXO connected via SIP trunk

2N OfficeRoute has these parameters: IP address 192.168.92.200 Listening port: 5060/5065 firmware version 2.3.11 Alcatel OXO parameters: IP address 192.168.92.246 Incoming port: 5060 version 7.7.1 Settings of 2N OfficeRoute Settings of SIP trunk for Listening port 5060 a. Switching SIP proxy off Default settings for SIP destination port in Alcatel OXO is 5060. 2N®OfficeRoute has port 5060 used by internal SIP proxy. That is why the SIP proxy has to be switched off for this case. b. Creating a new SIP trunk At menu Telephony services → Devices → SIP lines add a new line. For a new SIP line: Fill in SIP server address as IP address of Alcatel OXO, listen port 5060. Highlight or change a priority of codecs which should be used for this trunk. Uncheck Register with proxy checkbox Write some description of this trunk. 2. Settings of SIP trunk for Listening port 5065 In this case it is not necessary to switch SIP proxy off. Creating a new SIP trunk At menu Telephony services → Devices → SIP lines add a new line. For a new SIP line - the recommendation as above. 3. Configuration of the LCR (Least Cost Routing) At Telephony services → LCR disable all rules which are not routed into GSM. Remove all rules from Telephony services – LCR – Normalization. 4. Incoming calls Incoming calls can be routed to some DDI in OXO. For our case it is number 100 which will be dialed into our SIP trunk for all incoming calls. Create a new Operator rule in Telephony services → GSM routing → Operator. Operator number is a number which will be dialed into chosen VoIP line for incoming call. Assign the Operator service with GSM modules which should use this incoming rule. You can do it at Telephony services → GSM routing → GSM. Create here a new rule for each GSM module and choose the service Remove all normalization rules in GSM routing menu. Alcatel OXO configuration Setup a count of VoIP-Trunk channels for VoIP trunk to OfficeRoute. 2. Choose Trunk group and check Public trunk checkbox. Change Link Category settings. 3. At Link Category menu setup all necessary parameters. 4. Assign Trunk Groups. 5. Setup Numbering plans table. As Base settings choose ARS. 6. Check Public numbering plan for incoming calls. 7. At Trunk Groups List assign List ID with Index at menu Numbering → Automatic Routing Selection → Trunk Groups List. 8. Setup IP address of 2N OfficeRoute, codecs, bandwidth and GW keep-alive timeout for VoIP trunk at menu Numbering → Automatic Routing Selection → Automatic Routing: Prefixes. 9. At menu Numbering → Automatic Routing Selection → Gateway Parameters setup listening port 5060. If you want to use 2N OfficeRoute with different listening port (for example 5065), just setup the number of the port here.

Can I send a FAX with my 2N OfficeRoute?

Can I send a FAX with my 2N® OfficeRoute? Yes, you can send a FAX via GSM or via T.38.

Default login credentials

The default user name is Admin with the password 2n. By default there is DHCP server enabled. Your PC will obtain the IP address automatically. The default IP of the 2N OfficeRoute is 192.168.1.1

Does the 2N OfficeRoute support HSPA technology?

Yes, it supports HSDPA. In case your bigger interest we can provide the solution with HSUPA as well

E-mail to SMS - How to set up

This explaining guide is created for 2N VoiceBlue Enterprise or 2N Office Route. Create user in VoiceBlue Enterprise/OfficeRoute. Enable the SMS rights for this user. If you want to create the user without SIP phone, do not enter Line number. Message setting – do not forget to set up „Email domain name“. To avoid the memory overload set the automatic storage action according to the picture below. In case you would like to allow sending SMSs from a part of the network or only from one computer, tick SMTP IP access control in „SMS delivery“ and then create a rule in „SMTP IP access“. In case you would like to make restriction only for the range 192.168.50.1 – 192.168.50.255, create the rule that you can see below. Now you need to set up POP3 and SMTP server in Outlook. First of all go to Tools -> Account settings and create new account. In account settings specify user name, IP address of OfficeRoute in incoming and outgoing mail server and login information. After this setting click "More Setting..." button and in section Outgoing Server activate parameter "My outgoing server (SMTP) requires authentication". Now you can create SMS and send it to a mobile phone. Do not forget to choose the correct account in Account setting.

FAX - How to configure

This HOW TO GUIDE is valid only with 2N OfficeRoute which has got a FXS port (analog interface). Open web configuration interface of the Office Route and log in. First of all you need to set up the FXS line. Configuration wizard has created this line by default but you have got a possibility to modify this line. For example to change its number. You can do it in Telephony - Services - Devices - FXS line. Then you need to create FAX line. This line IS NOT created by the configuration wizard so you have to do it manually. Go to the Telephony Services - Devices - FAX line and click + sign to add a new fax line. Choose the GSM module where is your SIM with fax number (you need to allow multi-numbering on your SIM). Then fill in FAX number (the same number like in your FXS line) and click save. If you want to use more then one GSM module for FAX, add another FAX line with different GSM module and the same FXS line number.(in GSM module there have to be also SIM with FAX number inserted). Now you have created all lines what you needed. Go to the LCR and create a new route with your FAX lines. Telephony services - LCR - routes. Fill in a name for your route and add your FAX lines that you want to use in this route. When you save your new route, click on LCR again because we need to create a rule for FAX route. It means that for example all your outgoing FAX calls will start with the digit 9. You need to define that a call with 9 at the beginning of called number will use the route that includes FAX lines. Click add sign + to create a new rule. Fill in Destination name (must be different then route name), fill prefix ^9 (it means that all called numbers start with 9), choose route and priority that will be used to choose lines in your route (when you have got more then one line, this priority will be used). Priority must be chosen every time. Click save. When you dial FAX call, you enter digit 9 at the beginning (in our case). This number is used for routing purpose but you can not dial this number to GSM network. It means that we need to cut off this digit. Go to the LCR - Normalization. Add a new normalization rule. Fill in prefix 9 and as a type enter "called outgoing". Prefix will be removed automatically so you do not have to fill in something else. Remove count keep 0. Save it. Everything for outgoing FAX is set. You can check your settings in LCR TEST. Fill caller number 199 (in our case) and call according to your wish but it must start with 9. Then click save and you will see the result of normalization and used outgoing SIM card for this FAX call. You do not need to set almost anything for incoming direction. GSM module recognizes the incoming FAX call (because it comes to your FAX number on your SIM) and transfers this FAX to your FXS line directly. (number 199 in our case). Please make sure that in LCR settings there is number 199 routed to FXS port. You have to create a rule for that with appropriate FXS line. There is also a possibility to send fax via TCP/IP. When you have got an account in VoIP provider which supports T.38 protocol, you can send FAX from the FXS port on your Office Route directly to your VoIP provider via TCP/IP connection. You need to do following: Create a new SIP Line. Telephony Services - Devices SIP lines. Fill VoIP provider SIP server IP address. SIP name (number or account name from VoIP provider), user name and password for authentication. The rest of settings are the same like in the case before. You just need to change lines in your FAX route. Instead of FAX line you will use this new SIP line.

GSM to VoiceMail - How to route all incoming calls from GSM to VoiceMail

If you want route all incoming calls from GSM to VoiceMail, you need to create one user to whom all messages will be recorded by VoiceMail. Afterwards you will be able to listen to all stored messages by calling to VoiceMail from the user’s station. Step by step guide First of all please check if you have a license for VoiceMail in your OfficeRoute. Then move to User management -> Users and create a new user. Set the parameters according to the picture below. When you have set all user´s parameters, move to Telephony services -> GSM routing -> Operator, create a new operator and set a number, which will be dialed to SIP proxy. Number is composed from prefix (888) and user number (101). When you have created the operator, assign it to GSM module in GSM routing -> GSM. Now all incoming calls to the given GSM module will be routed to SIP proxy. In SIP proxy we make “loopback” and reroute all calls with prefix 888 to LCR. Before we start with LCR configuration, we have to create a Voicemail. We can do this in Devices -> DISA lines, where we create a new DISA line and configure it according to the picture below. In LCR normalization create a new rule with parameters from the picture below. In this case LCR will be detecting prefix 888 and automatically replacing it with prefix vm-. In LCR -> Routes add new route to previously created DISA line. Finally go to LCR and add new LCR record in which you will be detecting prefixes 888 and vm-. Calls with those prefixes will be routed to voicemail. Now is everything set properly and all incoming calls to the set GSM module will be routed to voicemail.If you want to listen to recorded messages, register to OfficeRoute as user 101 and dial number 888.

H.323 trunk - How to configure

IMPORTANT!!! READ IT FIRST!!! If you got a permanent license for OFR, you can use SIP and H.323 but not at the same time. Go to the Administration – Main configuration menu. Set the VoIP Protocol parameter to H.323. If there is no VoIP parameter, that means that you can use both SIP and H.323 at the same time. Example Call Method: Choose mode for H.323 trunk Gateway address: The IP address of office route which will be used for call method Gateway Gateway prefixes: H.323 message Alias dialed digits field. Only for Call method Gateway Display name: H.323 Caller ID field Numbers: If there exists aclip (Caller ID) when calling to this H.323 line, this identificator will be overwritten. Otherwise this number will be sent. Listen port: OfficeRouteH.323 listening port Codecs: Choose one or more codecs which will be used for the trunk Reject call code: Call clear reason for rejected calls. No route code: Call clear reason for calls which can not be routed to the network. If you have troubles with connection, try setup Disable Fast Start and H.245 tunneling on both devices.

How many ME users can I use?

It depends on SIP proxy users. License key allows how many SIP Proxy users, then you can have the same number of ME users.

How many UMTS channels has got the 2N OfficeRoute?

There is only one UMTS channel . You can also choose up to three GSM channels but for the version with FXS port only up to two GSM channels.

How to deny malicious calls from internet that make enormous bills on OfficeRoute

Your OfficeRoute/Omega Lite could be exposed to attacks of malicious applications (robots) from Internet. They search for "victims" to make outgoing calls for some paid services. That way owner of the device could get an enormous bill for calls over his account (usually, at the VoIP operator). A robot looks for a device that should respond to its INVITE and tries various prefixes in front of the called party number. If it finds the right one to root the call to the PSTN, it repeats the call untill someone finds the enormous bill on his account. You could protect your device by setting acceptable source addresses of messages which are sent to listening ports of SIP lines. It could be done in Network --- Filtering. It could be a little more complicated if IP addresses of you VoIP providers are being changed from time to time. From the firmware version 2.3.13 you can protect your device easily, just with smart LCR settings. From the version 2.3.13, all calls from outside to the SIP internal line are denied by default (see settings in Network --- Filtering). Define the unique prefix that will be forwarded to the PSTN (over SIP line to the VoIP provider). By default it is 0. You need to reject any call coming from the network using the SIP line intended for the VoIP provider, i.e. addressing its listening port. That means you have to reject calls starting with prefix 0. So you can change 0 to e.g. 9 in Normalization, if it comes from the provider's SIP line. There would be no rule for 9 in LCR and this call would be rejected. IMPORTANT: Do not put the rule for "any prefix" (.*) in the LCR. At the other side, choose hard-to-match passwords for user accounts in your OFR/OL. This will protect it from robots trying to register to your SIP proxy.

How to deny malicious calls from internet which make enormous bills on Office Route

Your OfficeRoute/Omega Lite could be exposed to attacks of malicious applications (robots) from Internet. They search for "victims" to make outgoing calls to some paid services. That way owner of the device could get an enormous bill for calls through his account (usually, at the VoIP operator). Robot looks for a device that should respond to its INVITE and tries various prefixes in front of the called party number. If it finds the right one to root the call to the PSTN, it repeats the call till someone finds the enormous bill on his account. You could protect your device by setting acceptable source addresses of messages which are sent to listen ports of SIP lines. It could be done in Network – Filtering. It could be a little more complicated if IP addresses of you VoIP providers are being changed from time to time. From the firmware version 2.3.13 you can protect your device easily, just with smart LCR settings. From the version 2.3.13, all calls from outside to the SIP internal line are denied by default (see settings in Network – Filtering). Define the unique prefix that will be forwarded to the PSTN (over SIP line to the VoIP provider). By default it is 0. You need to reject any call coming from the network using the SIP line intended for the VoIP provider, i.e. addressing its listen port. It means that you have to reject calls starting with prefix 0. So you can change 0 to e.g. 9 in Normalisation, if it comes from the provider's SIP line. There would be no rule for 9 in LCR and this call would be rejected. IMPORTANT: Do not put the rule for "any prefix" (.*) in the LCR. At the other side choose hard–to–match passwords for user accounts in your OFR/OL. This will protect it from robots trying to register to your SIP proxy.

How to make a factory reset of 2N Office Route?

Switch off the device Press button FRES and keep it pressed. Button is in the hole of the case, so you will need appropriately long and thin "tool" e.g. pen. Switch on the device After 30 s realese the button (depress it) Wait for OFR to boot. When you finish it, you will have the default IP address. It has been 10.0.0.1 till version 2.3.1. Since 2.3.1 default IP address is 192.168.1.1.

How to report technical issue on 2N OfficeRoute?

Please send us: Exact problem description Firmware version Serial Number Configuration file Download logs from the device Ethereal (Wireshark) trace in case of a VoIP (RTP) problem (specially if there is a problem in communication between SIP phones and SIP proxy inside OFR) Called party and calling party numbers (if relevant) Any information which you think might help us to solve the issu

Iam not able to connect to the configuration interface of OfficeRoute

I have changed the IP address of the OfficeRoute and I am not able to connect to the new IP from my web browser. If you use DHCP server of OfficeRoute, you have to change the settings of the server. Connect OfficeRoute directly to the ethernet port of your computer Change your network settings not using DHCP - set up the IP address of your computer from the same subnet of IP address of OfficeRoute. For example: You have set up the new IP of OfficeRoute to 192.168.200.1. Set up the IP address of computer to 192.168.200.2. Open web browser and enter 192.168.200.1 Login to the OfficeRoute as Admin Go to the Network --- DHCP server and modify the settings of DHCP server (Start address, End address, Router option and DNS option) in accordance to the new IP address of Office Route. Change your computer´s network settings to DHCP If you do not use DHCP server of Office route, set up the IP address of you computer from the same subnet of IP address of OfficeRoute. For example: You have set up the new IP of OfficeRoute to 192.168.200.1. Set up the IP address of computer to 192.168.200.2.

I forgot the Admin password

Connect the OfficeRoute via USB cable and change the Admin password After connecting the USB cable to your PC, a new COM port will be installed. You can be asked for a driver (you can download it from http://www.2n.cz/en/downloads/p=officeroute%7Ccategory=software). Connect to the OfficeRoute by some terminal software (hyperterminal or putty) via a new virtual COM port which will appear after OfficeRoute connecting. The communication speed is 921600. Press Enter after connecting - the configuration will appear. You can change the Administrator´s password by pressing 2

if-you-need-to-change-an-ip-of-the-2n-officeroute

Go to Network – and modify DHCP server settings At main configuration modify settings of network Change The IP address of internal SIP line (Telephony services – Devices – SIP lines – line 16) in accordance with new IP address. Restart the OfficeRoute manually.

IVR - How to create (without 2N IVR Editor)

IMPORTANT!!! READ IT FIRST!!! This FAQ explains how to prepare an IVR script and voice messages for an OFR IVR. This document became almost obsolete when 2N IVR Editor has been issued (http://www.2n.cz/en/products/umts/officeroute/downloads/ – Software). Anyway, it is still useful because you can find the detailed description how to prepare correct voice message files, as well as to understand internal structure of IVR. Creating voice files The format for IVR messages into 2N OfficeRoute is 8bit, 8kHz, mono, Alaw. You can record it (or edit recorded samples) in GoldWave audio editor (http://www.goldwave.com/). Record a new message Highlight the part of the recorded voice which you can save. Choose File – Save Selection as... Choose Attributes as A-Law, 8000Hz, 64kbps, mono Rename type of file from *.wav to *.Alaw Creating an IVR script Record all messages in audio files. Messages must be in ISDN A-law raw format. Create IVR script named script.disa, according to the following example. Put all messages and script.disa to one zipped file using tar.gz. Files must be zipped without folders. This file package should be stored in the internal FLASH memory. After upgrade to this version it is necessary to upload package of voicemail messages (in Telephony services > Devices > DISA lines > Upload DISA/IVR package to internal flash). IVR Commands WAIT - waits without starting audio message- parameters: TIMEOUT – waiting interval - example:WAIT TIMEOUT=1s CALL_ANSWER - picks the incoming ringing call up- parameters: none CALL_RINGING - changes the incoming call’s status to RINGING, e.g. if the call is received from SIP (VoIP) side, calling party receives SIP message 180 Ringing and hears a ringing tone- parameters: none CALL_PROGRESS - changes the incoming call’s status to RINGING with the voice channel connected, e.g. if the call is received from SIP (VoIP) side, calling party receives SIP message 183 Session progress and any audio file could be played to it- parameters: none CALL_HANGUP - hangups a current call- parameters: none AUDIO - plays an audio file (message)- parameters:SRC – file name (file must be coded in A-law format); be careful to write a correct pathBARGEIN - {TRUE|FALSE} – if TRUE and a DTMF code is received, playing of the audio file is immediately cancelled and script jumps to the next line- example:AUDIO SRC=Welcome.Alaw BARGEIN=TRUE WAIT_DTMF - waits to receive a DTMF code and stores it in the variable DTMF- parameters:TIMEOUT – maximal time waiting for DTMFMAXCNT – maximal count of digits (characters) waiting forENDCHAR – the end-of-receive characterSTRIPENDCHAR - {TRUE|FALSE} – if TRUE, the end character will not be stored to the DTMF variable- example:WAIT_DTMF TIMEOUT=10s MAXCNT=1 STRIPENDCHAR=FALSE- waits for only one character TRANSFER - transfers call to the certain number- parameters:DEST – the number a calling party should be transferred to- example:TRANSFER DEST=100- transfers call to line 100TRANSFER DEST=$DTMF- transfers call to a line number, stored in DTMF variable GOTO - jumps to the certain raw of the script- parameters:DESTID – a raw ID- example:GOTO DESTID=begin- jumps to the raw that has parameter ID=begin GOTO_IF - conditional jump – jumps if variable gets a certain value- parameters:DESTID – a raw IDVAR – a variable to be compared toVALUE – a value to be compared to- example:GOTO_IF VAR=DTMF VALUE=1 DESTID=sales- if the DTMF variable has value 1, jump to the raw with ID=sales An Example of the IVR Script (script.disa): WAIT TIMEOUT=1sCALL_ANSWERWAIT ID=begin TIMEOUT=1sAUDIO SRC=1-Welcome.Alaw BARGEIN=TRUEWAIT_DTMF TIMEOUT=10s MAXCNT=1 STRIPENDCHAR=FALSEGOTO_IF VAR=DTMF VALUE=1 DESTID=czechGOTO_IF VAR=DTMF VALUE=2 DESTID=englishGOTO_IF VAR=DTMF VALUE=3 DESTID=deutschGOTO_IF VAR=DTMF VALUE=4 DESTID=espagnolGOTO DESTID=beginWAIT ID=czech TIMEOUT=1sAUDIO SRC=menu-czech.Alaw BARGEIN=FALSEGOTO DESTID=beginWAIT ID=english TIMEOUT=1sAUDIO SRC=menu-english.Alaw BARGEIN=TRUEWAIT_DTMF TIMEOUT=10s MAXCNT=1 STRIPENDCHAR=FALSEGOTO_IF VAR=DTMF VALUE=1 DESTID=salesGOTO_IF VAR=DTMF VALUE=2 DESTID=nobodyGOTO_IF VAR=DTMF VALUE=3 DESTID=nobodyGOTO_IF VAR=DTMF VALUE=4 DESTID=nobodyGOTO_IF VAR=DTMF VALUE=5 DESTID=operatorGOTO DESTID=beginWAIT ID=sales TIMEOUT=1sAUDIO SRC=sales.Alaw BARGEIN=FALSETRANSFER DEST=202GOTO DESTID=beginWAIT ID=nobody TIMEOUT=1sAUDIO SRC=nobody.Alaw BARGEIN=FALSEGOTO DESTID=beginWAIT ID=operator TIMEOUT=1sAUDIO SRC=operator.Alaw BARGEIN=FALSEGOTO DESTID=beginWAIT ID=deutsch TIMEOUT=1sAUDIO SRC=deutsch.Alaw BARGEIN=FALSEGOTO DESTID=beginWAIT ID=espagnol TIMEOUT=1sAUDIO SRC=espagnol.Alaw BARGEIN=FALSEGOTO DESTID=begin Creating IVR package *.tar.gz Open Power Archiver 2007 (http://powerarchiver.brothersoft.com/powerarchiver2007-10.2) and Create a new archive. Enter name for a new archive and choose Type of archive TAR Find folder with IVR files (script, audio files)

Lets say that we want to divide outgoing GSM numbers between 2 GSM providers in accordance to the dialed prefix and all another outgoing calls should be routed into VoIP provider. Prefixes 71, 72, 75 belong to GSM Provider1, prefixes 74 and 84 belong to GSM Provider2. Lets say, that we have SIM cards of Provider1 inserted in GSM module 1 and for Provider2 we have only one SIM in GSM module 2. How to create that rule? Add new Routes (Telephony services – LCR – Routes) for GSM Provider1 and Provider2 and assign GSM modules for them (it is possible to highlight more than one line. Just hold CTRL and click the mouse button). Add a new LCR rule (Telephony services - LCR) for the prefixes of both GSM providers. Fill destination name and the prefixes, choose Route name. Shift the rule up (to the top of list) – it has to be before .* (that means everything). Delete the destination „To GSM“ which has the prefix .* (this is chart for all prefixes). Make a new rule for all prefixes which will be routed into VoIP provider. Leave prefix field blank and choose the route to your VoIP provider.

Mobility Extension - How to configure

In case you have registered SIP extension (IP phone, softphone, ...) to your 2N OfficeRoute and you would like to assign Mobility Extension to this station, then please follow steps bellow. Go to Users management -> Users and choose user to whom you want to assign Mobility Extension. In settings of this user check parameters "Outgoing ME enabled" and "Incoming ME enabled" and specify the number of your Mobility Extension station. Now save all changes and move to Telephony services -> Mobility Extension. For proper function of Mobility Extension function you have to activate parameters "Outgoing ME enabled", "Incoming ME enabled" and set in "Mobility Extension SIP route" IP address of your 2N OfficeRoute. If you call from your Mobility Extension station to 2N OfficeRoute, you will be automatically recognized and routed not to standard destination but to DISA set in parameter "Mobility Extension DISA line". Now if there is an incoming call to SIP station 100 from example above, Mobility Extension station will start ringing. If you do not get any call on your ME station, then please check if your LCR is set properly. ME station prefix (above- in the example 7) must be routed to GSM network.

Normalization - How to use

If you are not familiar with Normalization in LCR, read the following explanation!!! Normalization means changing the prefix of the called or calling (caller’s) number. This figure shows how does it works. It could be done before the LCR (incoming point) or after the LCR (outgoing point). If it is done before LCR, number will be changed i.e. normalized and after that sent to the LCR table. If we do the normalization after the LCR, number is sent to the LCR table in original form, appropriate route is found and change is done at the exit (outgoing) point. You can select one of four options, according to the number which you want to change and which action you want to do: Called outgoing (the most often used) Called incoming Caller outgoing Caller incoming If you select Any in the Line field, defined normalization will be applied to any line in case when the specified prefix is matched. If you specify a certain line, normalization will be done only if the call is incoming/outgoing from/to the specified line. Attention!!! Digit(s) that you put in the Prefix field will be automatically stripped (removed). Number in the Remove count field means: how many digits AFTER THE PREFIX will be removed too. Digits in the Add number field will be added to the number, after removing prefix (and additional digits) from the LEFT i.e. at the head of the number. Example: You call from 111 to 0603555666 and you want to send it over the route ToPBX (contains only one line named PBX). You want to change the called number to 55603555666 in the case of prefix 06. This change would apply only if the call is routed to the line PBX. At the same time you need leading 0 to select ToPBX route in the LCR table. You have to set: Line: PBX valid for calls routed to line PBX only (see the Type setting too) Prefix: 06 digits 0 and 6 will be removed Remove count: 0 no digits after the leading 06 will be removed Add number: 556 add 55 and “returns” 6 Type: Called outgoing

Registration of 2N Lift8 to 2N OfficeRoute

Info This FAQ briefly describes how to register 2N Lift8 with 2N OfficeRoute. In a few steps you will learn how to configure both devices. Every step contains screen from web interface and brief description of configuration. Configuration of 2N Lift8 Open Lift8 tool and go to section Configuration->Parameters->Network Settings. Fill in IP address, mask and gateway for your VOIP module. SIP server is IP address of the OfficeRoute. Login and password will be created in the next steps. Now open web interface of the OfficeRoute and go to section Telephony services->SIP Proxy and click on “Switch SIP Proxy on“. Now go to section User Management->Users and press + button in the low right corner. Now create user, fill in username and password (these credentials will be used in the first step). Also fill in the line number, it is recommended to use the same for all the fields. E.g. 224 for password, 224 for username and so on. Once you will finish press save in the low right corner. Now you should be able to see the user you just created in the user list. Now you can use credentials of created user in the first step. Then you can check whether the registration was successful or not in the section Telephony services->SIP Proxy->Registrations.

Settings of 2N OfficeRoute as a GW for Siemens OpenScape PBX

By default there is DHCP server enabled. Connect your PC to the OfficeRoute, open standard web browser and enter the IP address 192.168.1.1. The configuration wizard will appear. Go through configuration wizard and apply each step by icon at the right bottom corner. Go through the steps and fill all necessary parameters¨ Create one Extension Enter the password for the Extension – the password is compulsory for the extensions. Do not fill in any fields for VoIP provider Make routing for incoming calls. Outgoing calls has to be dialed with prefix for outbound calls „0“. It is possible to change this settings.

SIP Proxy - How many users can be registered?

The basic license is for ten users. When you want to register into your SIP Proxy (part of the 2N OfficeRoute?) more then 10 users, you need to buy an additional license. Please contact sales department of 2N sales@2n.com The number of SIP proxy users is not limited but we recommend to use the devices for maximum 50 SIP proxy users registered at the same time.

SIP Proxy port - How to change it from default 5060

If you want to change OfficeRoute SIP proxy port from default 5060, please follow steps below. First of all go to section Administration -> Main configuration and in parameter SIP proxy port specify a port which you want to use (in example below 5062). Then go to Telephony services -> Devices -> SIP lines and edit the internal SIP line. Here fill ":port_number" to parameter SIP server address. Now is everything ready and you can register your IP station on port 5062 instead of 5060.

Voicemail - How to configure

Recorded messages may be stored in the internal FLASH memory (approximately 1 hour of stored messages) or on the SD card. After upgrade to this version it is necessary to upload package of voicemail messages (in Configuration > Devices > DISA lines > Upload voicemail language pack). It is possible to record your own pack of massages (all messages must be in ISDN A-law format; set of massages is packed by TAR and GNU Zip and it has fixed structure). How do internal structure and internal logic work? An IP phone is registered to the internal SIP proxy. SIP proxy is connected with LCR by an internal SIP line. VoiceMail is DISA service line. To be able to reach Voicemail we need to set up a LCR route and LCR rule. Call must reach VoiceMail in the format vm-### where ### is number of IP phone’s number. As most of IP phones does not support dialing alphabetical characters, it is possible to make a normalization in LCR. Step by step guide Let say we have IP phone number 200 and we want to set a voicemail box for this extension. Log into web interface as user ‘Admin’. Check status of licence (voicemail users´ number must not be equal to zero – see picture bellow). Create a new account for user 200 (User management > Users > Add user). Set Username and Password (it serves as registration name and password). Set line number (it is extension number – UID). Enable Voicemail for this user. Set PIN for authorization for calls from another extension/mobile phone. Set mobile phone number for notification SMSs. Add a new Voicemail DISA service line (Telephony services > Devices > DISA lines > Add DISA service line). Set route to Voicemail (Telephony services > LCR > Routes > Add route). Set LCR route to VoiceMail (Telephony services > LCR > Add). Set Normalization in LCR (Telephony services > LCR > Normalization > Add). Set rule in SIP proxy for Voicemail (Telephony services > SIP Proxy > Add rule). Set Forwarding into Voicemail and set Voicemail number to 888extension (in our case it is 888200) – you can see example from a web interface of a hardware IP phone.

What is the maximum number of VoIP calls via UMTS module?

With codec G.711 (best quality) you can do approximately 4 calls. With codec G.729 (medium quality) you can do approximately 5-6 calls.

What to do when OfficeRoute stopped working after a month.

Check your license Connect to the configuration menu via web interface (enter the IP address of OfficeRoute to your web browser) Log in into the device as Admin or by your user´s account On the first page you will see information about licenses – check line Expires (hours) If your license expired, ask Customer Care or Technical Support for an unlimited license

Wireshark trace - How to make a trace of a call?

The 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).In order to capture all the packets which are sent via particular LAN, the devices have to be part of the same 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). Installation Download the program and appropriate installer (e.g. Windows installer). Wireshark is then initiated Start capturing: „Capture->Start" or use icon #1 End capturing: „Capture->Stop" or use icon #2 Restart capturing with deletion of previously captured packets: „Capture->Restart" or use icon #3 Save results: „File->Save As->Wireshark/tcpdump/... -libpcap(*.pcap;*.cap) Open previously saved trace: „File->Open" Basics of filter application: I want to filter the packets according to source IP- eg: ip.src_host=="192.168.5.7" I want to filter the packets according to destination IP - eg: ip.dst_host=="192.168.5.7" Protocol type: tcp, udp, sip, ... The terms can be combined via logical expressions : and, or, not, ... Applying a filter - Apply, Deleting a filter - Clear Example of captured communication Example of captured communication after applying a filter 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. Save it via Save as>Wireshark/tcpdump/... -libpcap(*.pcap;*.cap) and send it to us.