View source for FAQ
From berofix - professional SIP gateway
== beroFIX FAQ == === After upgrading i still have the previous installed Firmware, why ? === beroFix hast a seperate upload and install mechanism, you might have forgotten to press the install link after the firmware has been uploaded. beroFix has 2 Slots for installed Firmware Versions, so you can upload 2 Versions and activate one of them, this makes it possible to go back to a running state if the new firmware does not work as expected. You can check the installed firmware Version either in the update-tool or on the Management->Info Page. === The Webgui has no red Border at the right site, is that a bug ? === This is not a bug, it is intended to look like this. === When i press "Add" nothing happens, why? === You have most probably blocked pop-ups. The beroFix GUI has several configuration options in pop-ups. === bfdetect is not detecting my berofix card or berofix Box === To detect the berofix device the bfdetect tool sends a Broadcast packet to the network. If you have installed a berofix card in your system you have at least more then one network device. In this case you have to start bfdetect with an argument (the IP-Address of the network device) where bfdetect should send out the broadcast message. For instance if berofix is eth4 with ip-address 10.0.0.1 you have to start bfdetect like this "bfdetect 10.0.0.1". === berofix with Asterisk 1.6 hangup after connect === If you use berofix together with a asterisk 1.6. please set ignoresdpversion=yes in the sip.conf. Otherwise you will get an Hangup after the connect on Outbound ISDN Calls. === my Callerid is not presented correctly on outbound calls (PTP) === Depending on the Type of Number settings of the Telco on point to point lines, it is possible that the callerid will not be displayed correctly on outbound calls. Please read the [[Type of Number Note]] for further infos. === How many channels can my berofix card handle === It seams to be not easy to explain how many channels a berofix card can handle. Please have a look to the [[berofix Channel densities ]]Note for more informations. === I lost the berofix IP-Address. How can I detect the device === We provide a tool called bfdetect which can be downloaded in the "Support/Download" section. bfdetect is able to detect berofix devices in the network also if you lost or forgot your password. See the corresponding section in the documentation howto use bfdetect. === how to use calldeflection/partial rerouting with asterisk === First of all you need to enable Calldeflection in beroFix by using the ISDN Group Settings under "more". In Asterisk you create a Calldeflection Request by using the "Transfer" application. If you want to transfer Extension 999 to an external Mobilephone with extension 0179999999 than you can do something like this in your extensions.conf: exten => 999,1,Noop(Calldeflection) exten => 999,n,Ringing exten => 999,n,Transfer(0179999999) exten => 999,n,Wait(3) exten => 999,n,Hangup === Which Framing Setting (regarding CRC-4) does the beroFix E1 Lineinterfaces have ? === Currently CRC-4 Framing is on by default, it is not yet possible to modify this setting in the 1.x Firmware Releases. === Why is my Callerid "berofix"/ or the SIP Authentication Name instead of the real ISDN Callerid ? === ISDN and SIP have different Callerid Fields. In ISDN there is: '''oad''' - normal callerid '''oad2''' - second callerid (real callerid if oad is overwritten with clip-no-screening) '''qsig-name''' - Name of the device in qsig enabled PBX Systems In SIP we have the "From" Tag with: '''from_user''' - normally used as authentication username, but also as callerid by most PBXes '''displayname''' - the screened name of the caller berofix needs to put it's possible 3 callerids somehow into the SIP Fields and vice versa. Obviously the best way is to put the oad into the from_user to make sure that most PBXes including Asterisk get the callerid in the right SIP Field. But there are certain circumstances when berofix cannot put the oad in the from_user field. E.g. when beroFix needs to register and authenticate itself at the PBX, then beroFix needs to use the authentication username as from_user. beroFix puts it's oad then into the displayname. The Solution to this problem is either tell the PBX to use the displayname as callerid or to not register beroFix with the PBX and instead use the "proxy" setting in the berofix settings. In this case berofix can safely put it's oad into the from_user and nothing needs to be changed at the pbx site. beroFix has some additional configuration variables that can modify it's behavior regarding the callerid coding under "more": '''OAD Setting''' - modifies which SIP information should be used as Callerid in direction to ISDN '''From ID Setting''' - modifies which ISDN Info should be used as SIP from_user '''Displayname Setting''' - modifies which ISDN Info should be used as SIP displayname Please read [[Config String or "more" options]] for more details. === How do I downgrade from a Firmware > 1.13 to 1.12 or below ? === Since the 1.13 release Candidates we have a security Mechanism in the Firmware Update Process to avoid installing of potentially corrupt Firmware images. Since 1.12 and Firmware below did not have this mechanism you can not simply downgrade. If you still need to downgrade you must rename the Firmware Image to contain the word "downgrade". So for example if you want to install appfs-1.12.tar.gz you must rename it to appfs-1.12-downgrade.tar.gz. This overrides the security mechanism. === How do i use CLIR with beroFix? === There is a simple way to use CLIR with specific calls. Just set the Value "CLIR on SIP" in the berofix GUI under "SIP->SIP" in the SIP Entries under "more". If you set "CLIR on SIP" = "anon" then you can simply send a Call with the Callerid "anon" to berofix and beroFix will set CLIR on the ISDN Lines. === What do the clock sources and master/slave settings mean? === Berofix has 2 Line Interface slots. The maximum number of ISDN BRI Ports is 8 (2x 4BRI) and the maximum number of ISDN PRI Ports is 4 (2x 2PRI). It is very important to define the ISDN timing source properly to prevent bit slipping and jitter between berofix and the ISDN Network, which can cause damaged Faxes or even CRC-Errors and therefore Link-faults. The timing source can be configured via the clock setting in the Webgui. It can be found in the "Hardware" menu. In most cases berofix should be configured to have just '''1 Master Lineinterface''' module and a second '''slave module'''. The master module should have a clock source which is actually connected to an ISDN Port of the ISDN Network. === 3CX: No Ringing during Redirected Calls === This FAQ handles the problem that there is no "Ringing", when redirecting incomming ISDN calls over the 3CX back to ISDN again. Please have a look to the [[No_Ringing_Redirect_3CX]] Note for more informations. === Why do I have phantom Ringing on an FXO Line ? === The FXO Module detects a Ringing on the Line depending on the Minimum and Maximum Peak Voltage during a Ring Burst. These 2 Values differ in different countries, so the Country Setting in the GUI in "Hardware" must be set correctly. The default value is "FCC", especially in germany it must be changed to "1TR110_DE" === What's the Content of the beroFix CDR? === The berofix CDR contains comma seperated fields which summarize every call. example: analog to sip call, terminated by sip side with a bye CDR,1,ANALOG:1,SIP,,10,,,-11/01/03-23:24:51,11/01/03-23:25:06,11/01/03-23:25:01,11/01/03-23:25:05,SIP,NUA_I_BYE:200 columns: 0 "CDR" 1 PID 2 FromTechnology(:Port) 3 ToTechnology(:Port) 4 Incoming OAD (Callerid or SIP FROM Header) 5 Incoming DAD (Destination Number or SIP TO Header) 6 Outgoing OAD 7 Outgoing DAD 8 Call Start Time 9 Call Stop Time 10 Connect Start Time 11 Connect Stop Time 12 Terminated by which side 13 Termination Cause === Howto use Clip-No-Screening? === To use Clip-No-Screening you need to do two things: * Use the proper Type Of Number (depends on the provider) * Use the correct '''Formatting''' of the Callerid '''according''' to the '''Type Of Number''' beroFix allows to change the Type of Number via the "onumplan" advanced Settings, either in the Dialplan in a SIP->ISDN direction or in the ISDN Group. The Type of Number (onumplan) defines if the numbering Format is: * unknown * national * international * subscriber The definition is made by berofix at an outbound call (to ISDN), which means that the Number itself must have the correct Format. A telephone Number consist of the International Part (eg. 49 for Germany), the National Part (e.g. 30 for Berlin), the base number itself (e.g. 259389 beronet) and the DID (e.g. 0 ). Based on these Parts you can now see the complete number according to the Type Of Number: * unknown 2593890 * national 302593890 * international 49302593890 * subscriber 2593890 '''Note''': that we ommit any prefixes like 0 or 00 or + in the national and international formats! === Gemeinschaft 4 (freeswitch) no ringing at inbound ISDN calls / Kein Klingelton bei eingehenden ISDN Rufen === The situation is the following/ Die Situation sieht etwa so aus: ISDN Call --> beroFix --> Gemeinschaft 4 / freeswitch --> SIP Phone The Call flow is then/ Der Callflow ist dann: <table border=1> <tr> <td> <b>beroFix </b> </td><td> <b>GS4/Freeswitch </b> </td><td> <b>Phone </b> </td> </tr><tr> <td><b>ISDN <-> SIP</b> </td><td><b> SIP</b> </td><td><b> SIP </b></td> </tr><tr> <td>SETUP -> INVITE -> </td><td> INVITE -> </td><td> -> INVITE </td> </tr><tr> <td>PROGRESS <- 183 </td><td> <- 183 </td><td> <- 180 </td> </tr> </table> Gemeinschaft 4/Freeswitch is changing the SIP Status 180 to 183. BeroFix maps a 180 to an Alerting ISDN message and an 183 to a Progress ISDN Message. Only the Alerting ISDN Message creates the ringback tone. / Gemeinschaft 4/Freeswitch ändert den SIP Status 180 auf ein 183. Berofix übersetzt ein 180 in eine Alerting ISDN Nachricht und ein 183 in eine Progress ISDN Nachricht. Nur die Alerting Nachricht erzeugt im Amt den Klingelton. The correct behavior should be:/Das korrekte Verhalten wäre: <table border=1> <tr> <td> <b>beroFix </b> </td><td> <b>GS4/Freeswitch </b> </td><td> <b>Phone </b> </td> </tr><tr> <td><b>ISDN <-> SIP</b> </td><td><b> SIP</b> </td><td><b> SIP </b></td> </tr><tr> <td>SETUP -> INVITE -> </td><td> INVITE -> </td><td> -> INVITE </td> </tr><tr> <td>ALERTING <- 180 </td><td> <- 180 </td><td> <- 180 </td> </tr> </table> To change this mapping you can create a custom call progress table. To do that go to Preferences->Call Progress and click on "Add Table". Call this table gs4 and choose the ISDN-SIP direction. Now just enable the option "Add 183 to Alerting Rules", then click "save". Now go to the SIP Account for your Gemeinschaft 4/freeswitch and modify it. Under "more" you find the option "Call progress table I2S" choose "gs4", "save" and "activate". Now beroFix translates the 183 to an ISDN Alerting: / Um die Übersetzung der Progress Nachrichten zu verändern, erstellt man eine eigene Call Progress Tabelle. Um dies zu tun geht man nach Einstellungen->Call Progress und klickt auf "Tabelle hinzufügen". Die Tabelle nennt man einfach "gs4", setzt die Richtung auf ISDN-SIP, aktiviert die Option "Add 183 to Alerting Rules" und klickt "save". Als nächstes bearbeitet man das SIP Konto für die Gemeinschaft 4. Unter "Erweitert" findet man die Einstellung "Call progress table I2S" hier wählt man "gs4" aus, klickt "save" und schließlich "aktivieren". Nun übersetzt beroFix das 183 in ein ISDN Alerting: <table border=1> <tr> <td> <b>beroFix </b> </td><td> <b>GS4/Freeswitch </b> </td><td> <b>Phone </b> </td> </tr><tr> <td><b>ISDN <-> SIP</b> </td><td><b> SIP</b> </td><td><b> SIP </b></td> </tr><tr> <td>SETUP -> INVITE -> </td><td> INVITE -> </td><td> -> INVITE </td> </tr><tr> <td>ALERTING <- 183 </td><td> <- 183 </td><td> <- 180 </td> </tr> </table> NOTE: I2S Call Progress Tables can only be created with the firmware 2.2rc5 and newer. More info on [[BeroFix Call Progress Tables]]. ACHTUNG: I2S Call Progress Tabellen können erst ab Firmware 2.2rc5 erstellt werden. Mehr Informationen zu [[BeroFix Call Progress Tables]]. === berofix runs fine in general, but i have sometimes issues === If you ever encounter issues like: random call drops, bad voice quality, Echo or such, you should check our [[Known Issues]] page. === I'm using the freePBX / trixbox image for the beroapplicance but my keyboard layout is not correctly detected=== If it is not setup as you wish, you can, modify the file located at "/etc/sysconfig/keyboard" This should contain (the beginning) of the following two lines: KEYBOARDTYPE="sun|pc", where sun means a Sun keyboard is attached on /dev/kbd, or pc means a PS/2 keyboard connected to a PS/2 port. KEYTABLE="<file>", where <file> is the name of a keytable file e.g. 'en_US','en_GB' or 'de-latin1-nodeadkeys'. === I need to change the T38 behaviour of berofix === There is an additional configuration option that can be used in the dialplan and in the sip account, which modifies the berofix t38 behaviour. More details can be found [[T38_options|here]]. === My beroNet Gateway doesn't respond to network requests anymore === If the beroNet Gateway doesn't respond to network requests (even not to Ping) anymore, but doesn't reboot automatically and the ISDN or Analog LEDs are still lit, then it might be that a broken Ethernet broadcast package has killed the Network Stack of the DSP. These packages are often created by MS Hyper-V Systems (not by all). There is a new DSP Firmware available that resolves this issue: http://www.beronet.com/wp-content/uploads/downloads//berofix/dsp_firmware/dsp-firmware-13.tar.gz To reproduce the problem and to check if the firmware works you can use the following pcap: [http://www.beronet.com/wp-content/uploads/downloads/berofix/dsp_firmware/msp-kill-trace.pcap ping-of-death.pcap] and use tcpreplay as linux command line too like this: #sudo tcpreplay --intf1=eth1 msp-kill.pcap === I want to send Modem, Unrestricted Digital or Fax (without T.38) calls to a VoIP Provider, but these calls fail === The beroNet Gateway can transmit data calls without modifying the audio (eg. with Echocancelation, Jitter compensation or codec translation) to a voip carrier with the clearmode or clearchannel feature. To use the clear mode a dialplan rule from ISDN (or Analog) to SIP must be created. In the additional configuration options of the dialplan rule under "more" the setting: clearmode_on_alaw=1 must be set. If the destination is another beroNet gateway, it needs a dialplan rule from SIP to ISDN (or Analog) with the same additional configuration option. It is important that the SIP codec is set to alaw/pcma in the SIP account. == beroNet Telephony Appliance FAQ == === Creating the Appliance Boot Stick === Before you can start to create the Appliance Bootstick you will need 2 files from the folder: http://www.beronet.com/wp-content/uploads/downloads/bootstick/ 1.) Download the tuxboot-0.5.exe 2.) Download newest *.zip file 3.) Start the tuxboot exe 4.) Choose the USB Stick to flash (!!!be aware to choose the right one!!!) 5.) choose preloaded ISO 6.) choose the BNTA-*.zip file 7.) wait till the files where written, then you're ready === Screen Problems with Linux Systems === If you have installed Debian (probably also others) on the beroNet Telephony Appliance and detecting screen problems or the device is not booting, you can try to to add these kernel boot parameters: '''video=1024×786@60 video=LVDS-1:d === Resize Partition === If the SSD is bigger than 30GB the partition has to be resized. In order to do that open Control Panel and navigate to System and Security. <br> <br> [[File:Contol Panel2.jpg]] <br> Below the Administrative Tools menu select "Create and format hard disk partitions". <br> <br> [[File:Administrative Tools2.jpg]] <br> After the SSD has been detected just right-click on it (Usually it is "C:") and select "Extend Volume" from the menu. A Extend Volume Wizard will show up and proceed with "Next". <br> <br> [[File:Extend2.jpg]] <br> <br> In the Extend dialog, you have to enter the size you want to extend. <br> The system does automatically recommend the maximum space to extend. <br> <br> [[File:volume2.jpg]] <br> <br> Proceed with "Next". <br> Done! === Jumperplan beroNet Telephony Appliance === *[[Media:DN2800MT.pdf]] == Starterpack FAQ == === Why can't I use My analog Phone or why does my Gateway reboots endlessly? === beroNet Gateways have a special Port-Pinout for each RJ45 Port. In case you connect 2 beroNet devices together - like in the Starterpack Scenario - it is necessary to use only a 4 Pin cable, like the orange one which is shipped with the startarpack. For more information about the Pinout please see here: http://wiki.beronet.com/index.php/BerofixDocuV2#beroNet_VoIP_gateway_baseboard_overview
Retrieved from "
What links here
About berofix - professional SIP gateway