NetBorder Express Gateway FAQ
What is the basic functionality of NBE?
Outbound (PBX -> Telco) - NBE listens on port 5066 for a sip invite from the PBX, once this invite is received NBE places the call out to the telco
Inbound (Telco -> PBX) - NBE sends a sip invite to port 5060 which is the port the PBX listens on, once the pbx receives this the call gets setup with the PBX.
Now below is a diagram showing how this all works. Also the ports and IP addresses can be changed using the "Quick Setup" in the NBE manager.
NBE 2.0 can't open Analog B-channel
Symptom:
NBE starts but on the "System Status" page of the Gateway Manager you see the following error message:
B-Channel[X] Can't open Analog B-channel wanpipe! This error could occur if: - WANPIPE is not a valid wanpipe ID. Edit the pstn-config.xml to specify a valid wanpipe for sangoma interface ID=<NBE ID value for A200>
Solution:
You can get this error message if you do not have your FXO modules plugged in the correct socket on the A200. The modules have to go sequentially into the sockets, starting at the lowest numbered socket, and without any gaps.
The lowest numbered socket is the socket furthest away from the metal bracket on the lowest numbered card.
The lowest numbered card (when using an A200 with atleast 1 Remora card connected via a Backplane) is the left-most card in the bundle when looking from the metal bracket to the backplane.
Why Is There No Ringing When I Call Into The PRI?
This is because NBE is configured for the switch to provide the ringing, so if it is not then you will not hear any ringing. To fix this issue simply tell NBE to provide the ring back tone.To do this follow the steps below.
1) First go into the "Gateway Manager" web interface and then click on "Configuration -> PSTN Config".
2) Next go to "Call Control -> ISDN configurations" and then select your T1 configuration.
3) Next in the configuration change the "Inband Progress Tones Generation:" as shown below to "ALWAYS".
4) Stop and start the gateway to apply the changes.
NBE 1.6
To do this please edit your routing rules which should be located at "\Program Files\Netborder\Express\Gateway\config\pstn-config.xml". Once in there simply find the lines as shown below in gray and then add the new lines below in orange.
<group ID="XYZspan1"
termination="TERMINAL"
switchVariant="DMS100"
BChannelNegotiation="EXCLUSIVE"
initiateRestartProcedure="YES"
inBandProgressTonesGeneration="ALWAYS"
inBandProgressTonesIndicator="BOTH">
Server With Multiple IP Addresses Will Not Start
Edit your "\Program Files\Netborder\Express\Gateway\config\gw.properties" and place the following line in there "netborder.net.primaryIPAddress=x", x being the ip address that you want NBE to listen on. After this is done start NBE again and it will start up.
Configuring NBE to work with DID's (DDI)
To pass the DID from the telco to the SIP pbx simply just modify your "default_sip_out" rule as shown below, this is found in your "\Program Files\Netborder\Express\Gateway\config\routing-rules.xml" configuration file. This will save the pstn.in.dnis into the %1 variable since the pstn.in.ani has the %0 variable because it was declared before the dnis. Now the (.*) is what saves it into the next available variable.
<rule name="default_sip_out" outbound_interface="sip" qvalue="0.001">
<condition param="transfer" expr="false"/>
<condition param="pstn.in.channelName" expr=".*"/>
<condition param="pstn.in.ani" expr="(.*)"/>
<condition param="pstn.in.dnis" expr="(.*)"/>
<out_leg name="" media_type="sendrecv">
<!-- To modify the target SIP destination, just change the value below -->
<param name="sip.out.requestUri" expr="sip:%1@localhost:5060"/>
<param name="sip.out.from.uri" expr="sip:%0@GW_HOST_IP:GW_SIP_PORT"/>
<param name="sip.out.from.displayName" expr="Netborder Express Gateway"/>
<param name="sip.out.transport" expr="udp"/>
</out_leg>
</rule>
3CX Is Sending CID In Quotes
If 3CX is sending CID in quotes then you can simply remove this by editing your routing-rules.xml which can be found in the file browser of the management interface. Once in there locate the rule "default_pstn_out_with_caller_name" as shown below.
<rule name="default_pstn_out_with_caller_name" outbound_interface="pstn" qvalue="0.01">
<condition param="transfer" expr="false"/>
<condition param="sip.in.requestUri.canonical" expr="sip:([\+]?[-\*\#\d]+)@"/>
<condition param="sip.in.from.displayName" expr="(.*)"/>
<condition param="sip.in.from.uri.canonical" expr="sip:([\+]?[-\*\#\d]+)@"/>
Then change the sip.in.from.displayName line so it looks like the rule below. Once this is done then stop and start the gateway and the issue will be resolved.
<rule name="default_pstn_out_with_caller_name" outbound_interface="pstn" qvalue="0.01">
<condition param="transfer" expr="false"/>
<condition param="sip.in.requestUri.canonical" expr="sip:([\+]?[-\*\#\d]+)@"/>
<condition param="sip.in.from.displayName" expr="([0-9]+)"/>
<condition param="sip.in.from.uri.canonical" expr="sip:([\+]?[-\*\#\d]+)@"/>
Sometimes 3CX is not responding to NBE SIP INVITES
If the CID has other characters other then letters and numbers then 3CX has been known not to respond. So to fix this you can modify the default routing rule and add a second routing rule to send CID with a comma in it.
1) Take your normal routing rule and change "(.*)" to "([0-9a-zA-Z]+)" as shown below.
<rule name="default_sip_out" outbound_interface="sip" qvalue="0.001">
<condition param="transfer" expr="false"/>
<condition param="pstn.in.channelName" expr=".*"/>
<condition param="pstn.in.dnis" expr="(.*)"/>
<condition param="pstn.in.ani" expr="(.*)"/>
<condition param="pstn.in.callerName" expr="([0-9a-zA-Z]+)"/>
<out_leg name="" media_type="sendrecv">
<!-- To modify the target SIP destination, just change the value below -->
<param name="sip.out.requestUri" expr="sip:%0@localhost:5060"/>
<param name="sip.out.from.uri" expr="sip:%1@GW_HOST_IP:GW_SIP_PORT"/>
<param name="sip.out.from.displayName" expr="%2"/>
<param name="sip.out.transport" expr="udp"/>
</out_leg>
</rule>
2) Then add a second routing rule which will get any calls with a comma in it, eg. "Doe, John". This new rule will have 2 variables as shown below and there will be a space between them instead of a comma, you can see this below where is says "%2 %3". So insure the space is kept and other then that just copy and paste this rule below the first rule. All required changes from the default rule above have been high lighted.
<rule name="sip_out_comma_in_caller_name" outbound_interface="sip"
qvalue="0.002">
<condition param="transfer" expr="false"/>
<condition param="pstn.in.channelName" expr=".*"/>
<condition param="pstn.in.dnis" expr="(.*)"/>
<condition param="pstn.in.ani" expr="(.*)"/>
<condition param="pstn.in.callerName"
expr="([0-9a-zA-Z]+),([0-9a-zA-Z]+)"/>
<out_leg name="" media_type="sendrecv">
<param name="sip.out.requestUri" expr="sip:%0@localhost:5060"/>
<param name="sip.out.from.uri" expr="sip:%1@GW_HOST_IP:GW_SIP_PORT"/>
<param name="sip.out.from.displayName" expr="%2 %3"/>
<param name="sip.out.transport" expr="udp"/>
</out_leg>
</rule>
3) Then simply restart the gateway and test to insure the issue has been resolved.
How To Change the PRI Numbering Plan.
On some rare switches they will require the numbering plan to be set. Below shows you how to set this for the calling and called number and then the possible values for both fields. Now it would be best to get the telco to tell you exactly how to configure this but if not then you can just simply try all combinations. Now all of this will be done in the \Program Files\Netborder\Express\Gateway\config\routing-rules.xml file in the default_pstn_out rule. Each time this is set you need to restart NBE gateway.
There is two parameters below to set the number plan type for the "called" number which are below.
pstn.out.phoneNumber.type
pstn.out.phoneNumber.numberingPlan
Now for the "calling" number there is two other parameters as shown below.
pstn.out.ani.type
pstn.out.ani.numberingPlan
The "type" field can be set to one of the following numbers shown in the list below.
Value Description
0 Unknown
1 International number
2 National number
3 Network specific number
4 Subscriber number
5 Abbreviated number
7 Reserved for extension
The "numberingPlan" field can be set to one of the following numbers shown in the list below.
0 Unknown
1 ISDN/telephony numbering plan (Recommendation E.164 [19])
3 Data numbering plan (Recommendation X.121 [21])
4 Telex numbering plan (Recommendation F.69 [22])
8 National standard numbering plan
9 Private numbering plan
15 Reserved for extension
Below is a sample of how the rule may look once modified. Note the changes from the default are in orange.
<rule name="default_pstn_out" outbound_interface="pstn" qvalue="0.001">
<condition param="transfer" expr="false"/>
<condition param="sip.in.requestUri.canonical" expr="sip:([\+]?[0-9]+)@"/>
<out_leg name="" media_type="sendrecv">
<param name="pstn.out.channelGroup" expr="default"/>
<param name="pstn.out.cpa.enable" expr="false"/>
<param name="pstn.out.phoneNumber" expr="%0"/>
<param name="pstn.out.phoneNumber.type" expr="0"/>
<param name="pstn.out.phoneNumber.numberingPlan" expr="1"/>
<param name="pstn.out.phoneNumber.type" expr="1"/>
<param name="pstn.out.phoneNumber.numberingPlan" expr="1"/>
<param name="pstn.out.ani.type" expr="1"/>
<param name="pstn.out.ani.numberingPlan" expr="1"/>
</out_leg>
</rule>
Caller name is not showing up.
The caller name arrives in a FACILITY message and the gateway is configured to get the caller name from a DISPLAY IE in the SETUP message. To get the caller name right, you will need to change the path configuration of the gateway as follows:
- In the Pstn config tab of the Gateway Manager, choose "Call Control" and
  then "ISDN Configurations" in the panel on the left.
- Double-click on the Call control config name in the ISDN Configuration Grid (usually FAS1-T1)
- A Fas Configuration display menu will then appear, in the "Caller Name Location Method" scrolling menu, choose "IN-FACILITY-MSG"
- In the "Wair Facility Delay (ms)" menu enter 100
- Save the new configuration by clicking on the Save button.
- If you've got more than one Call Control Configuration, you will need to perform operations 3 to 5 for all your Call Control configurstions.
- Restart the gateway
After install/re-install there are no Sangoma cards available
After performing a new install or a re-install of NBE you find that there are no Sangoma ports active, Windows says it found new hardware, and/or after configuring the cards in NBE the gateway fails to start.
The problem is that the Sangoma drivers did not install correctly and need to be re-installed. You can confirm this by looking at the "Device Manager". The Sangoma card(s) will show up as "Network Controller"
To solve the problem, uninstall the drivers, reboot and install the drivers again.
- Go to the Windows "Device Manager" and uninstall the "Network Controller" by right-clicking and selecting "Uninstall".
- Go to the Windows "Add or Remove Programs" and remove all "Windows Driver Package - Sangoma Technologies...", if there are any there
- Reboot the computer
- After reboot Windows will report that new hardware has been found and will try to install the drivers, click "cancel" to this window. You will see a window for each Sangoma card you have.
- Click on "Start -> All Programs -> Netborder Express Gateway -> Install Device Drivers". A Command Prompt window will open up and the drivers will be installed. If any "New Hardware Found" windows pop-up, click the "cancel" button.
- Once the drivers are installed check in the device manager that the cards are detected properly. In the example the system has an A200 and an A104 card installed.
- If you continue to have problems please contact Sangoma support at "techdesk@sangoma.com"
Caller ID on SIP REFER transfers is incorrect/incomplete
By default, the routing rule handling the SIP REFER requests will copy the From: header to the resulting INVITE message, so that the Caller Number presented to the transfer target is the one of the transferring SIP endpoint (IVR or person triggering the transfer).
Default Rule:
<rule name="sip_to_sip_transfer" outbound_interface="sip" qvalue="0.001">
<condition param="transfer" expr="true"/>
<condition param="sip.in.referTo" expr="(?U)(.*)(?:;|\?|$)"/>
<condition param="sip.in.from.uri" expr="(.*)"/>
<out_leg name="" media_type="sendrecv">
<param name="sip.out.requestUri" expr="%0"/>
<param name="sip.out.from.uri" expr="%1"/>
</out_leg>
</rule>
To present the Caller Number corresponding to the PSTN number associated to the call instead, the contents of the To: header must be used instead
Alternate rule:
<rule name="sip_to_sip_transfer" outbound_interface="sip" qvalue="0.001">
<condition param="transfer" expr="true"/>
<condition param="sip.in.referTo" expr="(?U)(.*)(?:;|\?|$)"/>
<condition param="sip.in.to.uri" expr="(.*)"/>
<out_leg name="" media_type="sendrecv">
<param name="sip.out.requestUri" expr="%0"/>
<param name="sip.out.from.uri" expr="%1"/>
</out_leg>
</rule>
Also, the default rule only copy over the Caller Number, and not the Caller Name if available. The following rule shows how to extract the Caller Number and Caller Name from the PSTN side of the call and package it in the resulting INVITE message.
PSTN Caller Number and Name rule:
<rule name="sip_to_sip_transfer" outbound_interface="sip" qvalue="0.001">
<condition param="transfer" expr="true"/>
<condition param="sip.in.referTo" expr="(?U)(.*)(?:;|\?|$)"/>
<condition param="sip.in.to.uri" expr="(.*)"/>
<condition param="sip.in.to.displayName" expr="(.*)"/>
<out_leg name="" media_type="sendrecv">
<param name="sip.out.requestUri" expr="%0"/>
<param name="sip.out.from.uri" expr="%1"/>
<param name="sip.out.from.displayName" expr="%2"/>
</out_leg>
</rule>
Setting Up NBE To Be The Network Side (telco side).
#1 - Clocking - You can change the clocking in NBE as shown in the picture below.
#2 - Signaling - You will need to change to network mode termination as shown in the picture below.
#3 - Cabling - You will need a cross over cable if you are connecting back to back with NBE or another pbx. For pinouts go to http://wiki.sangoma.com/Pinouts#A101/2/4%20Cross. Now note this is different pinouts then a ethernet cross over cable.
How to assign particular channels to particular group
One "Channel Group" (PSTN Config of GUI) per campaign, assign channels in groups, then specify a channel group to use in routing rules based on the
campaign to use (phone number prefix?)
Why Am I Seeing "B-Channel[0] failed to setup callerID detector" In My Application Log?
If you are seeing this error in your Application log then your ToneDecoder.dll file has not been registered correctly. To fix this go to "\Program Files\Netborder\Express\Gateway\drivers\sangoma\hw_abstraction_driver\" from the command prompt and run the command "regsvr32 ToneDecoder.dll". Once this is done then try to start NBE again to be sure the issue is corrected.
International deployments with analog boards
For deployments with analog boards outside of North America, there are important aspects of the gateway configuration to consider:
1. Analog line variant
Analog line physical characteristics (ex. electrical impedance) are different from a country to another. This is set in the Physical Configurations->Analog Configuration->Analog Variant value of the Gateway Manager PSTN Config. About 110 countries are supported. If your country isn't in the list, try neighboring countries or countries you know have a similar phone system.
2. Tone detection set
In-band tones used to report progress of a call are different from country to country. The tone definition set used for detection by the gateway are selected via Physical Configurations->Audio Configuration->Tone Set Attributes value of the Gateway Manager PSTN Config. Tone sets are available for all the countries for which the line variant is supported.
If your country isn't in the list, an existing tone detection set (.xml file) will need to be modified to match tone definition of your country, following the instructions found in the Tone Configuration Guide located in the "doc" subdirectory of your NBE installation. You would not be able to rename the .xml file to the name of your country, so you'll need to refer to the tone set using the original country name (ex GERMANY if you modified the file called germany.xml to match your country tones.)
3. Tone generation set
The gateway must generate in-band tones in some conditions (ex. ringback played back on inbound calls). Currently, audio files must be provided by the gateway installer if deployed outside North America. See the Tone Configuration Guide located in the "doc" subdirectory of your NBE installation for details.
4. Caller ID detection
The delivery mode of Caller ID changes from country to country. The NBE Release Notes provides the list of countries for which Caller ID is supported.
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