Talkswitch Switch 324 User Manual

TA L K S W I T C H D O C U M E N TAT I O N  
TALKSWITCH  
VOIP NETWORK  
TROUBLESHOOTING  
GUIDE  
RELEASE 3.24  
C T.T S 0 0 5 . 0 0 8 0 0 1  
®
A N S W E R S W I T H I N T E L L I G E N C E  
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TABLE OF CONTENTS  
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T A L K S W I T C H V O I P N E T W O R K T R O U B L E S H O O T I N G G U I D E  
1. INTRODUCTION  
This guide assumes you are having difficulties with VoIP calls in a branch-to-branch  
configuration and you have already configured the following configuration settings in  
both the proxy server and remote site locations:  
IP addresses for private, public and proxy server network settings.  
VoIP and Line Hunt Group numbers.  
Services in the router IP mapping tables.  
This guide describes the following steps:  
Verifying existing TalkSwitch configuration settings.  
Attempting a VoIP call and troubleshooting specific problems.  
During the VoIP call attempt, you will be redirected to specific places in the configuration  
settings in order to help pinpoint the exact problem. The following figure depicts a simple  
branch-to-branch VoIP setup with one remote site location and the proxy server location.  
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2. VERIFYING CONFIGURATION SETTINGS  
Verify the TalkSwitch unit configuration settings contained in this section for the remote  
site and proxy server locations. This section also lists additional configuration setting  
information to help you determine possible errors in your TalkSwitch settings.  
2.1 IP CONFIGURATION  
The System IP Settings section displays the TalkSwitch unit private IP address. We  
recommend a static IP address for a TalkSwitch unit behind a firewall. The Public WAN IP  
Address section displays the IP address seen from outside the network and may be a public,  
or TalkSwitch unit IP address.  
2.1.1 System IP Settings  
If the unit uses a static IP address, verify that the IP addresses under Use the following IP  
and DNS information are correct and that the Default Gateway is within the Subnet mask  
range.  
If DHCP is enabled (Obtain IP and DNS information automatically is selected) the unit IP  
addresses are grayed out and display the latest addresses retrieved by the DHCP server.  
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Verify that the MAC address is reserved or that the firewall supports UPnP (firewall specific  
information).  
If the Unit IP Address is public, ensure the unit is not behind a firewall.  
Test: Ping the IP address to verify that it is valid.  
2.1.2 Public WAN IP Address  
If the IP address is static, verify the address.  
If the IP address is dynamic, check the address using an address-server checker (such as  
checkip.talkswitch.com). If FQDN is enabled, confirm the spelling of the domain name and  
ensure a utility exists to update the domain when it changes.  
Test: Ping the FDQN name to verify that it is valid and matches the value in the Current  
public WAN IP address field.  
2.2 VOIP CONFIGURATION  
The TalkSwitch Profile section displays the Proxy and Registrar IP addresses or domain  
names. The Username and Password section contains authentication details for all  
TalkSwitch units making up the branch-to-branch VoIP configuration.  
2.2.1 Proxy/Registrar  
One unit or location must be set as the proxy server for all locations in the branch-to-  
branch VoIP configuration. Selecting the This TalkSwitch is the Proxy / Registrar box  
populates the server name boxes with the unit public IP address. Unchecking then  
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rechecking this box updates the Server Names entries. The Registrar Authentication option  
turns authentication on or off.  
If this location is the proxy/registrar, make sure the box is checked and verify the server  
name addresses and that both the proxy and the registrar entries are the same.  
If FDQN is used, make sure the entries are typed correctly.  
Verify that all locations have the proper proxy/registrar IP addresses.  
Note: Leave the Outbound Proxy IP field blank, as it is not required for this type of  
configuration.  
2.2.2 Username and Password  
All locations must have matching username and password entries used for authentication  
purposes. If no Realm/Domain is entered, TalkSwitch will sub in ‘norealm’ for the  
authentication packets.  
If authentication is enabled by selecting yes(digest) in the Registrar Authentication  
option, ensure the username and password are the same in all locations.  
Note: Authentication is not required in a branch-to-branch VoIP configuration.  
2.2.3 VoIP Lines  
The VoIP Lines option is found in System Information -> VoIP Configuration.  
Ensure the number of VoIP lines is set to a value other than zero.  
2.3 VOIP NUMBERS  
The Activate VoIP Number option is found in System Information -> VoIP Numbers.  
Ensure all locations have at least one VoIP number activated and that no two locations  
have the same VoIP number assignment.  
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2.4 LINE HUNT GROUPS  
The Set Line Hunt Group options are found in System Information -> Line Hunt Groups.  
Choose a hunt group and TalkSwitch VoIP Numbers as the Line Type. By default, hunt group  
88 is set to use the TalkSwitch VoIP lines.  
2.5 SIP AND RTP PORT ASSIGNMENTS  
The VoIP Ports options are found in Options -> Troubleshooting -> Advanced.  
TalkSwitch uses 5060 for SIP signalling and 6000+ for RTP packets. These values are set in  
the firewall routing table.  
2.6 FIREWALL PORT MAPPINGS  
Verify the firewall settings for services listing both SIP and RTP ports.  
5060 should be forwarded to the IP address of Box 1 (UDP).  
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Confirm the RTP ranges for each TalkSwitch unit.  
Range of RTP Ports  
Unit  
Start RTP  
End RTP  
Type  
Unit 1  
6000  
6006  
UDP  
Unit 2  
Unit 3  
Unit 4  
6010  
6020  
6030  
6016  
6026  
6036  
UDP  
UDP  
UDP  
Note: Port 9393 may also need to be mapped to the TalkSwitch unit for TCP/IP access  
configuration.  
2.7 REGISTRAR  
You can verify registration of the units within the TalkSwitch configuration software. On  
the VoIP Configuration software page you have the options to View Registration Status  
and in a branch-to-branch scenario you will be able to View Registrar Entries.  
2.7.1 View Registrar Entries  
This button will display all VoIP numbers registered to the CVA with their current public  
IP address and SIP port. Confirm all units are listed with their corresponding IP addresses.  
2.7.2 View Registration Status  
This button will display the unit’s VoIP numbers and current registration status.  
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3. TESTING YOUR CONFIGURATION  
This section describes using registration and placing a remote to proxy site VoIP call to  
pinpoint the location of possible system configuration errors which may be causing the  
problem.  
3.1 VERIFY REGISTRATION  
The steps below follow a sequence to ensure proper registration of the remote and proxy  
units. If the remote and proxy units are registered, then initiate a VoIP call as described  
in Initiate the VoIP Call on page 8, otherwise, follow the steps below to check the following  
configuration settings, in order:  
Authentication  
Proxy location SIP port mapping and remote location IP address  
Remote location SIP port mapping  
If the unit’s client VoIP numbers are not registered, check to see if authentication settings  
are causing the problem (if you are not using Registrar Authentication, skip the procedure  
below):  
1. At the proxy registrar location, go to System Information -> VoIP Configuration,  
then set the Registrar Authentication field to none.  
2. Perform a configuration save at the proxy location, followed by a configuration save  
at the remote location. This will attempt to register the remote location with the  
proxy location.  
3. Check the registration status again at the proxy location.  
4. If the proxy and remote locations are now registered, the authentication values at the  
remote location were causing the problem. You can now initiate a VoIP call.  
The following steps will verify the registration status of the remote location to verify if  
proxy location SIP port 5060 is valid and if the remote location IP address is configured  
properly:  
1. Go to System Information -> IP Configuration at the remote location and verify  
that the public IP address setting is correct.  
2. Go to System Information -> VoIP Configuration at the remote location and con-  
firm that the correct IP addresses are entered for the Proxy and Registrar Server  
names.  
3. At the proxy registrar location, verify that the firewall port mappings for SIP have  
5060 pointing to the proxy location private IP address settings.  
4. Go to System Information -> VoIP Configuration at the proxy location, then click  
the View Registrar Entries button. If registered, go to the next section, otherwise,  
redo this section starting with Step 1 above.  
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The following steps will verify the registration status of the remote location to verify if  
SIP port 5060 is configured properly:  
1. Go to System Information -> VoIP Configuration at the remote site, then click the  
Registration Status button. If registered, go to section 3.1 below and initiate a  
VoIP call.  
2. Verify that the firewall port mappings for SIP have 5060 pointing to the proxy  
location private IP address settings.  
3. Perform a configuration save at the proxy location, followed by a configuration save  
at the remote location.  
4. If the proxy and remote locations are now registered, the port mapping values at the  
remote location were causing the problem. You can now initiate a VoIP call.  
3.2 INITIATE THE VOIP CALL  
From the remote site location, dial a valid VoIP number for the proxy server location that  
is set to ring an extension only. This will verify the remote and proxy location SIP  
mappings, as well as the public IP and proxy server IP addresses at the remote location. If  
the proxy location doesn’t ring or no ringing is heard from the remote location, confirm  
the remote and proxy units are registered and attempt the call again.  
3.3 ANSWER THE VOIP CALL  
With ringing heard at the proxy and remote locations, check for audio to verify the  
RTP settings.  
3.3.1 Audio received at proxy location  
If audio is received at the proxy location, then the RTP settings in the proxy location  
router mapping table are correct.  
3.3.2 No audio received at proxy location  
If no audio is received at the proxy location, verify the following:  
Proxy location RTP settings. For more information, refer to SIP and RTP Port  
3.3.3 Audio received at remote location  
If audio is received at the remote location, then the RTP settings in the remote location  
router mapping table are correct and VoIP is properly configured.  
3.3.4 No audio received at remote location  
If no audio is received at the remote location, verify the following:  
Proxy location RTP settings. For more information refer to SIP and RTP Port  
If you encounter any difficulties not covered in this guide, please contact  
Technical Support.  
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