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Frequently Asked Questions


To search our FAQs, enter a partial description of what your looking for in the search box on the left and click "Search FAQs".

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 SmartLink VoIP 
 Debug and Logging 
 Using Syslog to troubleshoot SmartLinks ( Link to this FAQ )
 The SmartLink supports syslog for simple troubleshooting.

To enable syslog logging, look under the GUI "System" menu, then select "Configuration". Set "Enable Syslog" to "yes" and put the IP address of your Syslog server in the "Syslog Server" field. For additional troubleshooting, you can also set "Enable Debug" to "yes", set "Debug Server" to the IP address of your Syslog server, and set the "Debug Connection Port" to 514. Click "Save" and then "System/Reload", "Reset".

If you are not familiar with Syslog, an easy to use freeware Syslog package is provided by Kiwi. See http://www.kiwisyslog.com. If Syslog doesn't give you enough information, we suggest you use hub or mirror a port on your Ethernet switch and use Ethereal. Ethereal will show you everything! 
 Gateways 
 The manual for the SL4020 shows a significant number of configuration parameters that I don’t see on the web configuration interface ( Link to this FAQ )
 Based on login, the Smartlink 4020 provides different configuration menus. The system login provides complete system configuration access. The user configuration login provides a subset of the system menus.

If you purchased your SmartLink 4020 from a VoIP service provider as a part of a calling service - The SmartLink 4020 configuration settings must interoperate with your VoIP service provider’s network. In an effort to provide you with reliable services, you provider may have decided not to provide customer access to the configuration interfaces.

 
 How do I access the web configuration interface?  ( Link to this FAQ )
 The web configuration interface can be access by pointing your browser to the IP address of the SL4020. The web interface works with current versions of Internet Explorer and Netscape. It does not work with the Firefox browser. From the factory, the SL4020 WAN Ethernet interface is set to be a DHCP client and the LAN is set to be a DHCP server.
    Set your PC’s LAN interface card to obtain an address automatically.
    Connect your PC’s Ethernet cable tothe LAN interface
    http://192.168.1.1
    You will then be prompted to enter a login password.
You will need the configuration password to access the system. If you purchased your SmartLink 4020 from a VoIP service provider as a part of calling package, you will have to get the configuration password from your service provider. From the Patton factory, the default system password is “root”  
 What do I need to specify as the callerid for call forwarding, distinctive ring, caller id blocking and do not disturb? ( Link to this FAQ )
 The SL4020 supplementary calling features of selective call forwarding, distinctive ring, caller id blocking, and do not disturb require specification of the calling party. There are several places in the SIP invite header that could carry that caller information. The "FROM" header of a SIP Message is as follows:

���From: CallerId;tag=G1024jh3-NXr3s......
The CallerID field of this header is used in the "Caller ID" fields of supplementary service features. Some implementations identify the callerID as the "Display Name"

For example: The "FROM" header of a SIP Message may look like:

���From: 123;tag=G1024jh3-NXr3s......

To use call forwarding, distinctive ring, caller id blocking and do not disturb specify "123" as the caller id.  
 I want to be able to lift the handset and have the phone call immediately connected to the call destination. I do not have a SIP server involved.  ( Link to this FAQ )
 There two things here, dialing by IP address and hotline dialing. The trick is putting the two together.
You can do direct IP dialing using the keypad on the phone. There's another FAQ posted on how to do IP address dialing.
In this case we need to change tell the SL4020 not to look for a SIP server and to set up a hotline call. Here is how you set up hotline dialing by IP address -
  1. Under Telephony, SIP - Set the SIP Registration Server Address: to the IP address (or domain name) of the Smartnode / FXO gateway. Save the changes on this screen. This setting directs the Smartlink to send the call to the FXO gateway.
  2. Under Telephony, SIP - Uncheck Send Registration Request with Expire Time:
  3. Under Telephony, phone one (or two), user information.
    1. Set the phone number to the number to be used for phone one. (For example: 555 for line one).
    2. Set the Caller id to be used for phone one.
  4. Under Telephony, phone one (or two), supplementary services set "IP dial srv " to YES.
  5. Under Dial out type, set to Hotline and code the SIP user part of the SIP call header. This will be combined with the IP address set in step #1 to form the complete SIP header.
  6. Save all the changes.
  7. Repeat steps 3-5 for phone 1 using a different phone number (For example: 444 for line two).
  8. Reload the SL4020.
  9. When the handset is lifted, a SIP call will be made to the destination sip address.
    SIP calls placed to 555@IP address of the SL4020 will ring phone one.

     
 How to extend dial tone (dial tone extension) from one location to another location without a SIP server. ( Link to this FAQ )
 The dial tone generated by a PSTN connection or a PBX can be extended over the Internet or an Intranet by using a pair of Smartnode 452x units or a Smartnode 452x and a Smartlink 402x unit. This FAQ describes how to perform dial tone extension using a Smartnode 452x and a Smartlink 402x. Smartnode to Smartnode configuration information is available under "Line Extensions" of the SN452x applications notes page. Once the setup is complete, the end user will be able to lift the handset on the analog phone, hear the dial tone from the PSTN or PBX and then dial a number just as if they were directly connected to the PSTN or PBX. Flash-hook is supported so features like call waiting and call hold will work. Tax pulse tone (not typically used in the USA) is not supported. The guide below assumes that WAN and LAN connectivity has already been established.

Hardware –

  • Smartlink 4021 or Smartlink 4022
  • Smartnode 4522, 4524, 4526 or 4528 with FXO ports (order code “JO” indicates FXO ports)
  • PSTN analog connection OR PBX with analog phone port
  • Analog phone

Physical Connectivity
Analog phone ----SL402x----- Internet/Intranet -----SN452x-------PSTN or PBX

In the SL4020

  1. Under Telephony, SIP –
    • Set “SIP Registration Server Address:” to the IP address or domain name of the SmartNode 4520
    • Uncheck “Send Registration Request with Expire Time:”
    • Save the changes
  2. Under Telephony, Phone 1
    • Set the phone number to the phone extension (example 111).
    • Optional - You can set caller id if you wish.
    • Under “Dial Out Type” , select Hotline from the drop down menu
    • Under “Hot Line Number”, enter the phone number that has been routed to the FXO port on the SN452x.
    • Save the changes
  3. Go to System,Reload and restart the SL4020

In the SmartNode 452x....Please use the Smartnode CO configuration information under "Line Extensions" of the SN452x applications notes. You will need to add call routing and mapping table to strip the called number that is passed from the Smartlink.

routing-table called-e164 TAB-IN
route ... dest-interface IF_FXO0 STRIP
mapping-table called-e164 to called-e164 STRIP
map ...% to \1
map default to ''

 
 How many VPN tunnels can be configured on the SL4020 ( Link to this FAQ )
 The configurable limit is 8 now. Based on our preliminary study of code and hardware we believe this limit could be extended significantly. It will eventually be limited by memory and network resources.  
 How do I automatically forward calls to a different extension? ( Link to this FAQ )
 From the web interface - Click on Telephony, SIP extensions and select "Conditional Call Forwarding Timer:". Set the timer to the number of seconds to wait before call forwarding is triggered.

Suppose you want to forward calls to phone number 123456. Pick up the phone set, you will hear the dial tone. You have to dial *73 and you will hear the dial tone again. Dial 123456 and you will hear beep beep beep.

On hook the phone set. Calls will now forward to the number that was entered. Dial #73 to cancel call forward.  
 How do I directly dial an IP address? ( Link to this FAQ )
 The SL4020 must be configured to enable dialing by IP address. This can be done on user configuration menu under Telephony, Line. Once enabled, IP SIP addresses can be dialed directly by IP address using the telephone keypad.
For example: If you want to call 123456 at 192.168.1.3 port 5060.

You have to dial: *72123456*192*168*1*3*5060
*72 is the code to indicate that an IP call follows.

Optionally, you can skip "*5060" at the end, because it is the default value.  
 How do I use Speed Dial? ( Link to this FAQ )
 The SL4020 supports association of numbers to call with telephone handset keys. The configuration of speed dialing can be done using the telephone keypad or via the web. Up to eight speed dial numbers can be set.

Setting up speed dial using your telephone handset:

1. Press *81 on your telephone keypad to enable the speed dialing function. This only has to be done once.

2. Set up to eight speed dial numbers from your telephone keypad by entering *71xyyyyyyy where:

*71 = Is the code to set speed dial keys
x = The telephone keypad number to be set
yyyyyyy = The number to be called

For example: You want to set the telephone key number "2" to speed dial the destination 123456.
1. Dial *712123456 on the telephone and then hang up. Speed dial is now setup.
2. To use speed dial just press 2 and wait for the call to complete.

Speed dial can be disabled by entering #81 from your telephone keypad.

Note: SIP URLs cannot be included in speed dial settings.  
 Smart Link 4020 Calling Features ( Link to this FAQ )
  Patton SL4020 Calling Features

The SL4020 family supports advanced calling features that can be turned on and off from phones attached to the SL4020. (Your telephony service provider must enable your service for these calling features to work.)
FEATUREKEYPADFEATUREKEYPAD
Call HoldF(flash) 1Call AlternativeF*
ConferenceF7Conference DropF8
Call TransferF4..
Do Not Disturb ON*82Do Not Disturb OFF#82
Distinctive ON*90Distinctive OFF#90
Call Waiting ON*91Call Waiting OFF#91
Incoming Caller ID Display ON*92Incoming Caller ID Display OFF#92
Self Caller ID Block Service ON*93Self Caller ID Block Service OFF#93
Anonymous Call Rejection ON*94Anonymous Call Rejection OFF#94
Incoming Call Block ON*95Incoming Call Block OFF#95
Call Forward Selective ON*96Call Forward Selective OFF#96
Call Forward All ON*97Call Forward All OFF#97
Call Forward Busy ON*98Call Forward Busy OFF#98
Warm Line ON*99Warm Line OFF#99
IP Dialing ON*80IP Dialing OFF#80
Speed Dialing ON*81Speed Dialing OFF#81
Call Return*60..
Configure Warm Line Number*70*70yyyyy
where yyyyy = number to call
.
Configure Speed Dialing Number*71*71xyyyyy
where x = speed dial key
yyyyy = number to call
.
Configure IP Dialing*72*72xxx*xxx*xxx*xxx*yyyy
where xxx = IP Addr. Octets
*yyyy(Optional) = port number
.
Set Call Forward Number*73Wait for 3 short confirmation tones before hanging up..
Access Voicemail*86..

 
 How to perform factory hardware reset ( Link to this FAQ )
 

*** WARNING – This procedure will download and reload of new firmware. All LAN and WAN network connectivity information will be lost. Do not use this procedure unless directed to do so by customer service ***

The SmartLink 4020 can be hardware reset to force the download of new system firmware. This procedure requires access to firmware version software, typically loaded on a local device for downloading. Upon completion of the procedure, the LAN port will be set as a DHCP server with an IP address of 192.168.1.1 and the WAN port will be set as to be a DCHP client.

Procedure:

1. Power the SL4020 off.

2. Press the RESET button with a pencil and hold it.

3. To erase all configuration parameters, power on and hold the reset button for about 8 seconds.

4. Release the button.

5. The SL4020 will go to factory default, and the IP will reset to WAN: 172.16.0.1 and the LAN will be reset to 192.168.1.1.

6. To connect to the LAN port you may need to change the network settings on PC to DHCP client or to fixed IP addressing like: IP address 192.168.1.2, subnet 255.255.255.0, gateway/router 192.168.1.1.

6. Using your browser connect to the SL4020. For example: http://192.168.1.1.

7. Download new firmware to the SL4020 using HTTP or TFTP.

8. Reload the system when propmted.

9. Reconfigure the LAN and WAN network settings for your environment.

10. Reload the SL4020.

11. If you changed your PC's network settings, change them back to the original values.

 
 General 
 Default Password ( Link to this FAQ )
 The default password for the 4021, 4022, and M-ATA units on the web interface is "root" Please change this password for security reasons. 
 How can I get my Smartlink to register with my Vbuzzer Account? ( Link to this FAQ )
 To get your Smarlink to register with your Vbuzzer account all you need to do is set it up according to the following:

SIP PAGE:
    *Sip Registration Server Address: vbuzzer.com
    Sip Port: 80
    Sip Domain: vbuzzer.com
    Voice Port: 5004
    Send Registration Request with Expire Time: 3600

PHONE PAGE:
     Phone Number: vbuzzer username
     CallerID Name: vbuzzer username
     User Name: vbuzzer username
     Password: vbuzzer password
     Port: 80 

 Telephones 
 Why do I get a login error when using the default blank password on my Patton SL4050 IP phone with IE7? ( Link to this FAQ )
 There is a bug with IE7 that does not like it when you use a blank password. What you need to do is obtain a new web browser like Mozilla and it will except a blank password. There you can set the password to what you would like and you should then be able to access the unit via IE7. 
 How do you transfer calls with a Patton SL4050/2 (SmartLink2 LINE SIP PHONE)? ( Link to this FAQ )
 Transfering calls with a SL4050/2 (Patton SmartLink Series 2-Line VoIP SIP Phone):

1 - First you will establish the call.

2 - Then to transfer, press the "Hold" button. This will play the default hold music for the caller being transfered.

3a - If you want to do an Announced transfer, you will then dial the extension you are transfering to, wait to talk to the person at that extension, and when ready, hit the "Transfer" button and hang up.

3b - If you want to do an Unattended/Blind transfer, you can dial the extension and then just hit the "Transfer" button and hang up the phone.
 
 Why does my Patton Smartlink IP Phone disconnect as soon as a call is placed to an older model PBX or softswitch? ( Link to this FAQ )
 This happens because the PRACK function is enabled by default. Under the "SIP Settings" tab, you will want to disable the PRACK function.

**Prack ensures that media information is exchanged and that the network checks before connecting the call**  
 How do I factory reset a SL4050 telephone ? ( Link to this FAQ )
 Lift the handset and dial the number 255*000. Then press "OK" three times.

The telephone will reset itself and reboot. Now you can connect to the telephones's IP-address and port 9999. The login credentials are an empty username and an empty password.

Note: Internet Explorer 7 and some Linux Browsers can not send an empty login/password. 
 How do I dial an IP address on the SL4050 SIP telephones? ( Link to this FAQ )
 1) Lift the handset (or press SPEAKER)
2) Dial the IP Address by substituting the asterisk "*" for the dots in the IP address. For example: 192*168*1*100
3) Press OK 
 I would like to be able to push one button and immediately get a dial tone from the PSTN so that I can make calls. I DO NOT have SIP server.  ( Link to this FAQ )
 Sounds like you want to do something like press "9" for an outside line and then dial number......
This can be accomplished by using a combination of features on the Smartlink phones or gateways along with a Smartnode serving as the gateway to the PSTN. The resulting design results in two calls being made - one SIP call from the Smartlink device to the SmartNode gateway and a second PSTN call from the Smartnode gateway to the destination address on the PSTN.
How it works- The SL4050/SL4020 makes a SIP call to the Smartnode when the speed dial button is pushed. The Smartnode accepts the SIP call and presents dial tone from the FXO line. The user enters the number to be dialed which is transmitted as DMTF tones to the Smartnode that passes the tones to the PSTN. The PSTN completes the call to the number dialed.
Since there isn't SIP server, direct IP address or DNS calling can be used to send the SIP call directly to a Smartnode gateway with an PSTN connection.
To make calling easier on on the end user associate a speed dial button, like the telephone keypad number "9", with the IP address or DNS name of Smartnode.
Optionally the SL4050/10 phone has 10 line buttons that can be set as speed dial buttons. All the end user would do is push a line button and the next thing they would hear is a PSTN dial tone pass through by the Smartnode.
Please see the "FXO Interface Configuration" in the Smartnode configuration guide for information on setting up the Smartnode to pass dialtone from the PSTN to a SIP call.
From the SL4050 web configuration interface -
  1. Select Line Key Settings
  2. Pick a free line key and select "one touch dial"
  3. Enter the DNS name or IP address of the Smartnode gateway that has the PSTN FXO connection
  4. Click on "submit" to save the changes
 
 How to upgrade firmware on the SL4050 phones ( Link to this FAQ )
 
  1. On the phone - Press the menu button once and then press the down arrow button three times. The LCD screen should display "IP Address:" with the IP address of your phone below. Note the IP address and then push the menu button once again to clear the screen.
  2. Download the firmware from the Patton website and save it on your PC. Note the location name of the saved file. For example: c:\SL4050-10-S-020905-2005-07-22.azf
  3. In Windows, Click on "Start", then "Run". In the "Open" input box, enter "cmd" and press OK. A DOS window will open.
  4. In DOS window on the command line, enter: "TFTP -i xxx.xxx.xxx.xxx put filename "
    where xxx.xxx.xxx.xxx is the IP address of your phone as noted from step number one above and "filename" is the name and location of the downloaded firmware file.
  5. With a local LAN connection between your PC and phone the download of firmware may take as little as 5 seconds. After the download is complete, the LCD screen will display "initializing" as the phone reloads. The phone's LCD screen will display the date and time after the download and reboot have completed.
  6. Close the DOS window on you PC.

Example:
A. Using the menu button on the phone you learn that your phone's IP address is: 192.168.1.21.
B. You downloaded the firmware image file "SL4050-10-S-020905-2005-07-22.azf" from the Patton website and save the file in the root directory "C:"
The command to load the phone is:
TFTP -i 192.168.1.21 put c:\SL4050-10-S-020905-2005-07-22.azf

You can verify the version of firmware loaded on the phone through the web interface on the phone. The firmware version is shown in the upper right corner of every screen.

 
 SmartNode VoIP/ToIP 
 Call Routing 
 Incomplete entry in routing table ( Link to this FAQ )
 In case of using a routing table with conflicting entries, the error NO MATCHING ENTRY (INCOMPLETE ENTRIES PRESENT) appears. This can happen when the route entries are as follows and the number 123 was dialed.

    routing-table called-e164 OUT
      route 1234 dest-interface IF_BRI_03
      route 123 dest-interface IF_SIP

To avoid this, following can be set to succeed the entry for 123.

    routing-table called-e164 OUT
      route 1234 dest-interface IF_BRI_03
      route 123T!0 dest-interface IF_SIP

Adding the T!0 forces the entry 123, although there is another entry which incompletely matches. 
 Why I can't call out to my SIP-provider ? ( Link to this FAQ )
 In order to send a call over the SIP protocol you need to have the whole number complete as SIP does not support the so called "overlap-dialing" (dial digit by digit).
ISDN telephones can send the number already "en-bloc" (if you dial first an lift the handset afterwards) or send it digit-by-digit. Analog telephones always send the number digit-by-digit. In the SmartNode configuration you will need to configure now a routing-table to collect every digit and, if there is no more digit after a timeout, route the call out to SIP.

Configuration example:

context cs
routing-table called-e164 RT_DIG_COL
route .T dest-interface IF_SIP

NOTE: IF_SIP is the name of your SIP-interface and can differ in your configuration. Please use here the name as configured in your configuration.

You need to route now the calls from the fxs/isdn interface to this routing-table in order to collect the digits like this:
route call dest-table RT_DIG_COL
 
 How to solve the problem '400 From_tag is required' ( Link to this FAQ )
 If the SIP-server sends back "400 From_tag is required" most likely the problem are missing "<>" - brackets around the FROM - header.

By default SmartNode does not use those brackets and the FROM-header looks like this:

From: sip:123@a.b.c.d:5060;tag=XYZ

However there is a way how to add them. The solution is to add a calling-name in order to add those brackets. Below you will find an example of how to create such a mapping-table:

mapping-table calling-e164 to calling-name MAP_CALLING_NAME
map (.%) to \1


Be aware you will need to use the mapping-table within a routing-table. You will have such a routing-table already in your configuration right before you route the calls to the sip-interface. All you need to do to use the mapping-table is use it as a function like this:

routing-table called-e164 RT_TOWARDS_SIP
route .T dest-interface IF_SIP MAP_CALLING_NAME

This will add the angle brackets around the from-header.
 
 How can I configure load balancing between different call-router interfaces ( Link to this FAQ )
 You may configure call load balancing between multiple interfaces using a service distribution group. It works not only for ISDN but also for mixed voip-isdn and only voip networks.
Attend the following two points:
- The load balancing is working only for outgoing calls.
- The time out have to be disabled.

Example:

service distribution-group SER_DIST_ISDN
cyclic
max-concurrent 1
route call 1 dest-interface "interface-name"
route call 2 dest-interface "interface-name"
 

 How can I remove or restrict Caller-ID (CLIP)? ( Link to this FAQ )
 There are two possibilities:
1. Set the ISDN Presentation Indicator (PI) to restricted:
172.16.40.125(ctx-cs)[switch]#mapping-table pi to pi MT-PI-TEST
172.16.40.125(map-tab)[MT-PI-T~]#map default to restricted

2. Delete the Calling-Party Nummer using a E.164 mapping table:
172.16.40.125(map-tab)[MT-PI-T~]#ble calling-e164 to calling-e164 MT-CNPN-TEST
172.16.40.125(map-tab)[MT-CNPN~]#map default to "" 
 Using Timeout and Termination Characters in Call-Routing Tabels ( Link to this FAQ )
 Call-Routing tables offer two possibilities to terminate overlap dialed numbers.
1. A dialling timeout
2. A special termination caracter like # or *
The timout and the caracter can be configured as follows:

172.16.40.125>enable
172.16.40.125#configure
172.16.40.125(cfg)#context cs
172.16.40.125(ctx-cs)[switch]#digit-collection timeout 5
172.16.40.125(ctx-cs)[switch]#digit-collection terminating-char #

For example:
172.16.40.125(ctx-cs)[switch]#routing-table called-e164 RT-CDPN-EX
172.16.40.125(rt-tab)[RT-CDPN~]#route 123T dest-interface Line0

According to this rule the dialed keys '12345#' will be immediatly matched and the number '12345' will be used without waiting for the timeout.

Special Cases:
The Termination Character can also be part of the rule, in which case it will NOT have the effect of cancelling the timeout period.

Examples:
Rule: #21#T
Dialled Keys: #21#1234
Effect: Timeout is aktive, used number: #21#1234

Rule: #21#T
Dialled Keys: #21#1234#
Effect: No Timeout, used number: #21#1234

Note: The first two dialled '#' do not cancell the timeout, they are part of the rule.

For a general digit-collection with timeout or termination caracter without any restrictions use the following rule:
Rule: T

In this case...
Dialled Keys: 1234
Effect: Timeout active, used number: 1234

or...
Dialled Keys: 1234#
Effect: No Timeout, used number: 1234

Do NOT use a rule as follows:
Rule: .*T

In this case...
Dialled Keys: 1234#
Effect: Timeout is STILL active because '#' matches the regular expression '.T', the used number will be: 1234#

 
 Codecs 
 Why do I hear a crackling noise when using the G.729 codec? ( Link to this FAQ )
 On the SmartNodes (4110, 4520, 463X, 465X 4830, 491X, 492X, 493X, IC-4FXS) is not possible to use two low-bit-rate codecs at the same time on an FXS port. Thus you must choose to use either G.723 or G.729. G.711 is always supported.
Try this:

enable
configure
system

ic voice 0
low-bitrate-codec g729

In your VOIP profiles, we suggest you use either the G.723 codec or the G.729 codec, but not both, and it should match your low-bitrate-codec selection. 
 Do the SmartNodes support G.729B?  ( Link to this FAQ )
 G.729 is defined in a standard with two Annexes.
G.729 is the original 8kb/s CS-ACELP Codec
Annex A defines a reduced complexity Codec>br> Annex B defines the silence suppression scheme for G.729

All versions are supported by the SmartNode family. The configuration allows selection of g729 and optional silence supression. The configuration maps as follows with the capability exchange in VoIP signalling.

If g729 is selected in the VoIP profile:
G.729 and G.729a are signaled

If g729 and silence supression are selected in the VoIP profile:
G.729, G.729a, G.729b and G.729ab are signaled

 
 Even if G.711 codecs are not specified in gateway (H.323, ISoIP), the SmartNode sends H.323 setups with G.711 (plus the additional explicitly specified codecs) to the remote H.323 party. Why? ( Link to this FAQ )
 The H.323 standard requires that H.323 endpoints (gateways, terminals etc.) be capable to process G.711. That is why the SmartNode always sends G.711 in the terminal capability set. If a specific codec is to be enforced a codec has to be specified in the interface (H.323) with the keywork exclusive.
 
 How much bandwidth does a VoIP call use? ( Link to this FAQ )
 The required bandwdidth depends on various factors such as:
- Codec
- Codec samle length
- Protocol stack (IP, PPP, Frame-Relay, etc.)
- Tranmission Network (DSL, ATM, etc.)
- Echo Cancellation

As a general rule of thumb the bandwdith for one call in one direction is between 10 and 110 kb/s.

An excellent overview of how these parameters can be tuned and effect the VoIP bandwdith can be found in the following TechNote.
http://www.patton.com/technotes/smartnode_qos.pdf
 
 What are the available Voice Codecs? ( Link to this FAQ )
 
Codec
Net Bandwidth per call (kbps)
Used Bandwidth per call (kbps)
Min. Compression Delay (ms)
Usage
G.711 a-law
64
96
10
Uncompressed, best voice quality, European audio-digitizing
G.711 u-law
64
96
10
Uncompressed, best voice quality, American audio-digitizing
G.723.1
6.3
17
30
Good voice quality at lowest bandwidth, like analog phone, acceptable delay

G.729
G.729a
G.729b

8
40
10
Best relationship between voice quality and used bandwidth, low delay. G.729b is the silence suppression scheme for G.729a and is supported by the SmartNode family. For the sake of simplicity SmartWare and SmartNode always use the term G.729a, implicitly meaning G.729a and G.729b.
transparent
64
96
10
Transparent ISDN data, no echo cancellation
 
 What must be observed with codec selection with H.323 FastConnect procedure in case of overlap dialing? ( Link to this FAQ )
 When using H.323 FastConnect (which is usually the case) together with overlap dialing the voice codec is sometimes not known to the B-side SmartNode in the route lookup. The configuration fragment below exemplifies how to configure a SmartNode such that it also accepts such incoming calls:
context cs
no number-prefix national
no number-prefix international
use tone-set-profile default
...
interface h323 h323_1
routing dest-interface bri1
remoteip 172.19.128.21
codec g711alaw64k
interface h323 h323_2
routing dest-interface bri1
remoteip 172.19.128.21
...
gateway h323
codec g711alaw64k 10 20
faststart
no ras gatekeeper-discovery auto
bind interface eth0 router
use voip-profile default
no shutdown

The bold section shows an H.323 interface without codec. This interface is used by the B-side SmartNode in case overlap dialling is used on the A-side and FastConnect is enabled.

 
 How does voice codec selection for H.323 work with SmartWare? ( Link to this FAQ )
 The selection of a specific codec is somewhat tricky with the H.323 protocol. In SmartWare a specific codec can be entered at 2 different locations:
a. in the h323 gateway configuration
b. in the h323 interface configurationThe following example exemplifies a typical configuration:
...
interface h323 myif
routing dest-interface isdn
codec g711alaw64k
use tone-set-profile default
...
gateway h323
alias h323-id inalp1400
alias e164 01233000
codec g723_6k3 30 30
codec g729
faststart
ras
gatekeeper-discovery manual 172.19.32.42 1719 RRS
bind interface eth0 router
no shutdown
use voip-profile default

The codec configuration in the gateway specifies the 2 codecs G.723 6.3kbps and G.729 as the set of allowed codecs. This set of codecs is sent to the remote H.323 gateway (or gatekeeper) during the capability exchange phase. The codec specified in the interface 'myif' is the preferred codec to be used. SmartWare places this codec is on top of the list of allowed codecs which is sent during the capability exchange phase. Note that the preferred codec specification only works when FastConnect procedure is used. Otherwise it is without effect.

 
 Debug and Logging 
 What are the different Layer 1 states of the ISDN port? ( Link to this FAQ )
 ISDN LAYER 1 STATES

ISDN-PORT NETWORK MODE (NT1):

State G1 = NT1 DEACTIVATED
State G2 = NT1 PENDING ACTIVATION
State G3 = NT1 ACTIVATED AND WORKING
State G4 = DEACTIVATION REQUEST RECEIVED

ISDN-PORT USER MODE (TE)

State F1 = TE INACTIVE, NO POWERFEEDING
State F2 = TE POWER FEED, WAITING FOR INFO 0
State F3 = TE DEACTIVATED
State F4 = TE IS WAITING FOR SIGNAL
State F5 = SIGNAL RECEIVED, SYNCRONIZATION
State F6 = SYNCHRONIZED, READY FOR RECEIVING
State F7 = ACTIVATED, WORKING
State F8 = LOSS OF FRAME ALIGNMENT

NOTE:
THE ACTIVATION PROCEDURE CAN BE INITIATED BY THE TE OR THE NT.
THE DEACTIVATION CAN ONLY BE INITIATED BY THE NT. 
 How do I debug QoS? ( Link to this FAQ )
 Debugging QoS is different from any other debug commands. It is a two step process. You must be in configuration mode.
1) Go into your service policy and specify "debug queue statistics detail 7"
2) Then do a show command: "show service-policy interface eth0". You can repeat this command as often as you want to view the current statistics. 
 How do I use the ACL debugs to debug a VPN Connection? ( Link to this FAQ )
 Debugging VPNs and ACLs is a bit different than using the other debug commands. It is a two step process to enable ACL debugging. You must first be in configuration mode.
1) Go into "context ip" and then into the ethernet interface and type the following debug commands:
"debug acl in"
"debug acl out"

2) Then you can enable and disable debugging of the ACLs by the using the command "debug acl" or "no debug acl".
Note: VPNs tunnels only work between the two networks configured as a VPN (usually two private networks on eth1 like 192.168.1.0 and 192.168.2.0). You cannot ping or test the VPN from the console port or the SmartNode administrator command. You must test between PCs on the two private networks. For instance, a PC at 192.168.1.10 should be able to ping a PC at 192.168.2.10 through the VPN tunnel. You cannot PING a PC on one of the VPN tunnels from the console or admisistrator account.

Additionally, "debug ipsec" provides the IPSEC debug monitor which is normal a one-step debug command.

See the command "terminal monitor-filter" to allow you to filter out the ACLs you want to see. For example, to see only the packets to an IP address 123, you can simply use the command: terminal monitor-filter .*123.* 
 General 
 How do I start a debug session in the SmartNode ( Link to this FAQ )
 1. Open a telnet session using the command line interface of Windows. Command: telnet "SN-ip-address|SN-hostname".

2. After opening the tool, open the properties of telnet and check the quick edit mode check box. Then change to layout tab and set the height of the screen buffer size to the maximum value of 9999.

3. Login to the SmartNode using your account data.

4. Change to the administrator mode using the command "enable".

5. Print the running-config and version out using the "show running-config" and "show version" command.

6. Start one or more required debug monitors simultaneous in the same terminal windows by using the "debug ...." command.

7. After recording the problem stop all debug monitors by using the "no debug all" command.

8. Mark the full out put in the telnet session.

9. Open a text file and try to interpret what may go wrong or send us the text file to support@patton.com.

10. As last you may close the telnet session.

 

 How can I update a SmartNode 4552/4554 in redboot mode ? ( Link to this FAQ )
 Hold down the reset button while you power cycle the unit and make sure you hold it down for at least 10 seconds. Connect a PC to the WAN port of the SN4554. You will have to set your PC statically to an IP address on the same subnet (172.16.40.2 255.255.0.0). The WAN interface IP address in reboot mode should be 172.16.40.1 255.255.0.0. You will then have to open a telnet session to the above IP address. Once there you should see the RedBoot> prompt.

Follow these instructions to reset back to factory.

1. RedBoot> fis load
2. RedBoot> go -s factory-config


 
 Why is my defined tone not played correctly?
How many custom tones can be configured? ( Link to this FAQ )
 The following could be the issue in case a configured tone in a call progress tone profile is not played as it is set.
In total 13 different tones, a combination of frequency and volume, can be stored in the SmartNode tone database. This includes as well the tones of the default call progress tone profiles.

For example

play 1 2000 440 -19 and
play 2 2000 440 -10

will use two different tones out of the 13 available.
In case the 13 tones are defined, no more tones can be added to the tone database even though they can be configured in an additional call progress tone profile. This results in a random tone played in the specified call state.

Possible solutions:

1. Remove all the unused call progress tone profiles and be sure that a minimum number of tones are configured.

2. Modify the default profile instead of creating a new tone set profile. In that way, the default profile will play the required frequencies and the number of tones is kept to a minimum.

3. In case you prefer to have call progress tone profiles with your own labels, delete all unused default call progress tone profiles, using the commands:

no profile call-progress-tone defaultBusytone
no profile call-progress-tone defaultReleasetone
...

 
  Which SmartNode CLI commands are not supported with the timer function ( Link to this FAQ )
 The internal timer configuration command is only able to execute commands that produce an immediate result. Some commands that execute asynchronously cannot be executed by the timer. The following commands (among others) cannot be executed by the timer:
  • ping
  • traceroute
  • dns-lookup
  • copy any kind of files from or to a TFTP server
  • reload without the forced option
 
  How can I configure Daylight Savings on Smartnode Products ( Link to this FAQ )
 

Daylight Savings on Smartnode

  1. On the SmartNode a timezone can only be defined with the offset from GMT. For Daylight Saving Purposes there can be Timers defined which adjust the Timezone Offset according to the Daylight Savings Rule.
  2. The Rules which can be applied are like
    • last Sunday of October
    • first Saturday of March
    • second Sunday of April
    • first Sunday after March 20
  3. Rule which cannot be applied
    • last Saturday before "the fast of Yom Kippur"
      • because "the fast of Yom Kippur" is not every year on the same date.
    • last Sunday of February
      • because february has in leap years one day more
  4. Determine Rule
    • ask your government or find on a website the Rule according which in your Timezone Daylight Saving is handled
  5. Make Timer Command
          timer  
  6. Variables in the Timer Command
          
  7. Example Timer Commands
    • Rule: On last sunday in march the clock is turned on 02:00 plus one Hour Daylight
    • Two Ways to make a Timer with the same effect:
            timer DaylightSavingsOn 02:00 mar 31st previous sunday every year "sntp-client gmt-offset + 03:00:00"
            timer DaylightSavingsOn 02:00 mar 25th next sunday every year "sntp-client gmt-offset + 03:00:00"
      
    • Rule: On second sunday in october the Daylight Savings time ends on 03:00 (from GMT+3 to GMT+2)
    • Two Ways to make a Timer with the same effect:
            timer DaylightSavingsOff 03:00 oct 14th previous sunday every year "sntp-client gmt-offset + 02:00:00"
            timer DaylightSavingsOff 03:00 oct 8th next sunday every year "sntp-client gmt-offset + 02:00:00"
      
  8. Remarks
    • There are always two ways to determine the same date
    • Because of setting the clock back, the timer above occures two times in one year, the first time in Daylight Savings Time to turn Daylight Savings Time Off, the second time one hour later without effect: the offset is already set to "+ 02:00:00" and nothing changes.
    • The system functions even across the border of a month
      • "mar 27th next sunday" -> is in the range of mar 27th till apr 2nd

  9. Save the modified configuration
    • Please be aware the timers above change the system configuration. Without saving the configuration the 'old' configuration will take effect in case of a reboot of the SmartNode.
    • The two additional timers below will save the current (modified) configuration. Note: Because the system time is changed immediately these timers have different times. Effectively they will be executed one minute after the time change.
      • timer SaveConfigDaylightSavingsOn 03:01 mar 25th next sunday every year "copy running-config startup-config"
        timer SaveConfigDaylightSavingsOff 02:01 oct 25th next sunday every year "copy running-config startup-config"
        

Example for Switzerland

  • Switzerland is in timezone GMT+1
  • Daylight savings shift the clock about one hour to GMT+2
  • Daylight savings begins on last sunday in march at 02:00
  • Daylight savings ends on last sunday in october at 03:00
  • Two Timers: first turns daylight savings time ON, second turns daylight savings time OFF
          timer DaylightSavingsOn 02:00 mar 25th next sunday every year "sntp-client gmt-offset + 02:00:00"    
          timer DaylightSavingsOff 03:00 oct 25th next sunday every year "sntp-client gmt-offset + 01:00:00"    
    
    
  • Show timers to see that the switching will be on the right date
          #show timer
          Timer                 Next Execution
          -----------------------------------------------------------------------------
          DaylightSavingsOn     2007-03-25T02:00:00 (in 178d12:11:32)
          DaylightSavingsOff    2006-10-29T03:00:00 (in 31d13:11:32)  
    

Command changed on R3.T

  • The following command is in software branch after september 2006 release R3.T deprecated but still working
    
          sntp-client gmt-offset (+|-) hh:mm:ss   
    
  • New following command should be used
          clock local offset (+|-)hh:mm   
    
  • The example timer commands above looks then
          timer DaylightSavingsOn 02:00 mar 25th next sunday every year "clock local offset +02:00"    
          timer DaylightSavingsOff 03:00 oct 25th next sunday every year "clock local offset +01:00"    
    

 

 How can I reset a forgotten password on SN1xxx / SN2xxx / SN4552 ? ( Link to this FAQ )
 A forgotten password can only resettet by booting with the factory configuration. The current configuration will be lost !

Press and hold the reset button until the SmartNode begins to boot. The SmartNode boots now with the factory-configuration. The login is then 'administrator' with an empty password 
 How can I reset a forgotten password on SN4xxx series ? ( Link to this FAQ )
 A forgotten password can only resettet by booting with the factory configuration. The current configuration will be lost !

  1. Connect to the SmartNode using the console cable.
  2. Cycle the power
  3. At the prompt "Press ^C to abort boot script, press enter to start immediately" press ctrl and c.
  4. Type in the command 'fis load'
  5. Type in the command 'go -s factory-config'
Now you can login with the username 'administrator' and an empty password.
Note: You can load every other configuration which is stored in the nvram of the SmartNode. Replace the 'factory-config' with the configuration which you want to load.

This will only work for code dated 11/30/05 or later
If you are having problems, and cannot upgrade to the 11/30/05 code or later, please contact support@patton.com for assistance. 
 How can I reset the IP adresses of a SmartNode 4552 to the factory default? ( Link to this FAQ )
 Press the reset button for 5 seconds until the Power LED starts to blink then release the reset button.
This will restart the unit with the factory configuration.  
 How can I check if the routing tables are loaded successfully? ( Link to this FAQ )
 In the context cs the command 'no shutdown' causes the routing tables to be re-loaded and errors to be printed to the telnet/console (if command 'debug session-router' entered before). 
 There is an error message after download:"Cannot delete file /flash/cli/spec.xml" ( Link to this FAQ )
 If for some reason the CLI file (containing the commands) did not exist when user or download batch filed tried to delete it the CLI issues the above error message. 
 What TCP & UDP Service Ports do I need to open to my SmartNode? ( Link to this FAQ )
 
Function/Application Protocol Source Port Destination Port Transport Port
H.323 - H.225 call setup (default) any 1720 tcp
SIP and H.323 Audio data streams (RTP and RTCP) 4864-5119 4864-5119 except 5060 udp
H.323 - Gatekeeper RAS (default) any 1719 tcp
H.323 - RAS Gatekeeper discovery (optional) any 1718 tcp, udp
ISoIP (Patton-Inalp ISDN over IP) ports (optional) any 1106 and 1107 tcp, udp
SNMP–Network Mgmt (optional) any 161 tcp, udp
NTP–Network Time Protocol (optional) any 123 tcp, udp
SIP–Internet to Phone (optional) any 5060 tcp, udp
SIP–Phone to Internet (optional) 5060 any tcp, udp
TFTP–Used for software upgrades and saving or loading configuration files (optional-for admin) any 69 udp
Telnet for administration (optional-for admin) any 23 tcp
ping (optional-for troubleshooting) any 8 icmp
tracert (optional-for troubleshooting) any 11 icmp
 
 Can I setup multiple VoIP Gateways on a SmartNode? ( Link to this FAQ )
 It is possible to configure multiple SIP gateways on a SmartNode and register them with individual settings to different SIP Servers (multiple domain support). With H.323 only ONE gateway can be configured. The SmartNode can register with a single Gatekeeper. 
 DynDNS is expiring my dynamic DNS enteries. How do I refresh to prevent this? ( Link to this FAQ )
 DynDNS.org removes dynamic entries if it is not refreshed or changed after 35 days. Some ISP's i.e., Comcast only changes the IP address about every 3 months. According to dyndns.org the "Static DNS" service should be used in such cases. The "static" service still allows an update, it just takes a bit longer to propagate. So even if you really have a dynamic address but it changes in intervals larger than 30 days, use the static service. 
 Sometimes a command entered into the CLI does not appear in the 'show running-config'. ( Link to this FAQ )
 If the command entered happens to be a default value (e.g. sntp-client poll-interval 60) the command is not displayed in the running-config but nevertheless active. This means that only commands and values other than defaults are displayed in the running-config.

 
 Overflow handling, if input to SmartNode is higher than maximum output. Can the overflowing calls be sent via the fall-back ISDN-line? ( Link to this FAQ )
 The WAN access speed is expected to be large enough to handle all voice connections on the SmartNode. Moreover the voice portion of the used access bandwidth is expected to be small with respect to total bandwidth. For example a DSL access with 700kbps or more may be used for 4 voice channels (SOHO, SME) applications. A subscriber with an ISDN PRI interface is expected to have at least 2Mbps access speed. In such a configuration there can be no voice overflow in the SmartNode. The fallback connection is used if a call cannot be established at set-up, because the network is down or congested or because the opposite gateway is not available.

 
 How does a SmartNode work in combination with IP-Phones? ( Link to this FAQ )
 SmartNodes and IP-Phones can operate in the same H.323 Zone. IP-Phones may be connected to the customer LAN or directly to the access Network. If the SmartNode is used as an ISDN over IP access device, it works independently of H.323 IP phones connected to the same network.

 
 How can the SmartNode be remotely managed when the IP network is down? ( Link to this FAQ )
 SmartNodes offer a local console interface which can be connected to a separate device, for example analog modem. In this way the SmartNode can be managed through a dial-up connection if the IP Network is down or the SmarNode is mis-configured.

 
 Keypad facility does not work with H.323. ( Link to this FAQ )
 In ISDN supplementary service can be invoked by means of thekeypad protocol. A service can be invoked with the digitsequence *21#. The phone sends these digits as information element 'Keypad Facility' and not as informationelement 'called party number'. As the H.323 protocol does not dispose of a way to transport 'keypad facility'information it gets lost.On the contrary H.323+ (H.323 Annex M3) is a tunneling protocol for the transparent transport of ISDN over H.323and thus inherently supports 'keypad facility'.Note that on some phones it is required to explicitly switch to keypad facility mode in order to send a specificdigit sequence as info element 'keypad facility'.

 
 Is DTMF supported in SmartWare? ( Link to this FAQ )
 Yes. Generally, DTMF can always be transported either out-of-band or in-band. in-band provides the most accurate timing reproduction of DTMF, but it is not suitable when a compressing codec is used, e.g. G.729 or G.723. In these cases, out-of-band has to be used for reliable DTMF transport. Smartware supports all mechanisms available today: In H.323: SmartWare uses a mechanism called H.245 alphanumeric for the transparent (i.e. loss-less) transport of dialed DTMF digits across an H.323 network. DTMF digits are extracted from the (digital) ISDN signal, transported to the remote H.323 party which inserts the DTMF digit into the signal again. SmartWare allows to configure an H.323 interface to 'relay' DTMF signals as described above or to leave the signal unchanged. The latter may cause problems when compression codecs are in use which may distort the DTMF signal such that a receiving IVR application is no longer able to decode the signal properly.

In SIP: You can choose between RFC2833 transport of DTMF, within the real-time data (default and most commonly used), or you can choose to have DTMF sent as SIP INFO messages - this way a SIP proxy that does not route RTP traffic will be able to see it. 
 The routes on my MS Windows machines suddenly change when a SmartNode is booted on the same subnet. Why? ( Link to this FAQ )
 The SmartNode is a router as per RFC1812/RFC1256 and thus sends "ICMP Router Discovery" messages to the subnet to which it is attached. Some MS Windows versions react to such messages and automatically adjust their routing tables. In order to avoid such unwanted routing table changes on MS Windows machines the "ICMP Router Discovery" can be switched off as follows (per interface):
context ip
interface eth0
no icmp router-discovery
 
 How does the gatekeeper registration work in detail? ( Link to this FAQ )
 Up to 3 different gatekeeper IP addresses can be configured on a SmartNode.If the SmartNode is unregistered from the gatekeeper by means of an URQ message, after sending a UCF message the SmartNode tries (instantaneously) up to 3 times (every 20s) with an RRQ message to register with the next configured gatekeeper (round robin). Once successfully registered the SmartNode re-registers every 90s again with a RRQ message. If 3 RRQ registration requests in a row fail the SmartNode switches to the next configured gatekeeper. If a RRQ is not answered with a RCF or RRJ the SmartNode resends 2 further RRQ messages in 20s intervals.

 
 Signaling works fine but there is no voice at all. What is the problem? ( Link to this FAQ )
 When using NAPT on a SmartNode there is one global IP interface (with the public IP address) and one local IP interface (with the private IP address). The H.323 gateway must be bound to one of the interfaces since it needs to know on which interface it must send broadcast RAS messages for gatekeeper discovery. When the gateway is bound to the local interface (the one with the private IP address) then signaling with a gateway or gatekeeper in the public network works fine, but RTP packets (voice packets) will use the private IP address and thus voice will not be routed to the destination. Thus make sure that the gateway is bound to the correct interface (generally the global interface when using NAPT).

 
 I need to know the Ethernet MAC address but do not have physical access to the SmartNode. Is there a way to retrieve the MAC address remotely? ( Link to this FAQ )
 Yes, the command 'show port ethernet' shows the current Ethernet configuration along with the MAC addresses.

 
 What are the necessary configuration settings for TDM data over IP (Dial-up ISDN tunneled over an IP network)? ( Link to this FAQ )
 
gateway h323 h323
q931-tunneling isoip-2 (mandatory to preserve BC over the H.323 IP network)
codec transparent 50 50
interface h323
...
codec transparent exclusive
dejitter-mode static
dejitter-max-delay 400
no echo-canceller
no silence-compression
 
 Hardware Interfaces 
 What are the ADSL VPI and VCI settings in my country ( Link to this FAQ )
 
Country and provider specified ADSL settings
TLD Country Provider ATM Protocol VPI VCI
ar, Argentina Arnet PPPoE 0 33
ar, Argentina Speedy PPPoE 8 35
at, Austria ? PPPoA 8 48
au, Australia Telstra PPPoE 8 35
be, Belgium Belgacom PPPoE/Bridging * 8 35
bh, Bahrain Telecom Company Batelco PPPoA 8 35
br, Brazil Brasil Telecom (brturbo) PPPoE 0 35
br, Brazil do rio grande do sul são PPPoE 1 32
br, Brazil Speedy da Telefonica PPPoE 8 35
br, Brazil Velox da Telemar PPPoE 0 33
ch, Switzerland All (Swisscom, Wholesales) PPPoE 8 35
cl, Chile Speedy [Telefonica Terra] PPPoE 8 32
cz, Czech ? PPPoA or PPPoE 8 48
de, Deutschland Deutsche Telekom T-DSL (including Wholesales) PPPoE 1 32
de, Deutschland Alice DSL PPPoE 1 32
de, Deutschland Mannesmann arcor PPPoE 1 32
es, Spain Telefonica PPPoE 8 32
fr, France France Telecom - Wanadoo PPPoA for PC/ix86 8 35
fr, France France Telecom - Wanadoo PPPoE for Apple/MAC 8 35
fr, France Free, zone degroupee Routed IP(gateway: XXX.YYY.ZZZ.254) 8 36
gr, Greece ? PPPoA 8 35
it, Italy Telecom Italia - Alice, Tele2 PPPoA 8 35
nl, Netherlands bbned PPPoE? 0 35
nl, Netherlands BaByXL/TISCALI PPPoE? 0 34
nl, Netherlands Versatel PPPoA 0 32
nl, Netherlands ? PPPoA ? 8 48
nz, New Zealand ? PPPoA 0 100
pl, Poland Netia-Net24 PPPoE 8 35
pl, Poland TPSA PPPoA 0 35
pt, Portugal Portugal Telecom - Telepac PPPoE/Bridging 0 35
qa, Qatar Qtel PPPoA 8 35
tr, Turkey ? PPPoA or PPPoE 8 35
uk, Britain British Telecom line, Any ISP PPPoA or PPPoE 0 38
us, United States ? PPPoA 8 35
ve, Venezuela CANTV NET PPPoA 0 33
za, South Africa Telkom SA Ltd PPPoE 8 35
 
 When do I need ISDN line power? ( Link to this FAQ )
 An ISDN S-Bus can provide up to 8W of 40V DC power. Many ISDN Phones draw their operating power from the ISDN S-Bus. This is usually not the case for PBX systems which are typically local mains powered. Check the technical specifications of your ISDN terminal equipment to find out if line power is required. 
 Does the ISDN port provide ISDN line power? ( Link to this FAQ )
 It depends on the model. SmartNode 4634 and 4638 can be software configured to provide line power to terminals. Smart-DTA always provides line power. All other SmartNodes don't provide line power. However, they do forward line power received on the TE port to the NT port. So if a public ISDN-PSTN line is connected to the TE port and this line provides power then the power will be available to the Terminals connected to the NT port of e.g. the SN4552. Optionally if there is no line connected to the TE port you can install the PM-BRI-EXT S-Bus phantom power supply to power Terminals connected to the NT port.

An ISDN S-Bus can provide up to 8W of 40V DC power. Many ISDN Phones draw their operating power from the ISDN S-Bus. This is usually not the case for PBX systems which are typically local mains powered. Check the technical specifications of your ISDN terminal equipment to find out if line power is required. 
 What is the difference between BRI NT and TE? ( Link to this FAQ )
 The ISDN BRI interface is a User-Network Interface (UNI). It has an asymmetric behavior. The two sides are denominated by NT and TE. TE ports Are found on ISDN Terminals (ISDN phones or PBX trunk ports) NT ports Are found on the NT (Network Termination) box. An NT port always connects to a TE port and vice versa. The pinout of the ports is such that you can use straight cable to connect to the respective ISDN equipment. 
 Why does Ethernet not work when connected a Laptop with PC-Card Ethernet interface to a SmartNode using a crossover cable? ( Link to this FAQ )
 Some PC-Card Ethernet interfaces do not provide enough voltage to be recognised as a proper Ethernet signal by the SmartNode network interface. We recommend to connect the Laptop to the SmartNode using straight cables via Ethernet hubs or switches.

 
 The analog facsimile attached to a terminal adapter (which in turn is connected to a S0 interface of the SmartNode) does not send the called party number. Why? ( Link to this FAQ )
 Analog faxes wait for a dial tone prior to send the called party number. Despite this the terminal adapter sends a setup message when hooking off the phone receiver (without called party number information element). If the routing tables of the SmartNode are configured such that any called party number is accepted no dial tone will be sent to the fax which means that the fax never sends the called party number dialled. Make sure that the routing tables are programmed such that a dial tone (continuous tone signal) is generated.

 
 When connecting a BRI SMartNode to a PBX trunk line I have bit-slips and problems with fax connections. How to solve this synchronization problem? ( Link to this FAQ )
 In installations where a PBX is connected to the PSTN and to a SmartNode ISDN VoIP Gateway at the same time, synchronization problems can occure. The problem exists beacuse the PBX expects a synchronous clock on all trunk lines. The SmartNode ISDN VoIP Gateway however can only deliver a synchronous clock if it is connected to a reference network/clock. If this is not the case the SmartNode clock and the PSTN clock will not be synchronous leading to bit slips between different trunk lines of the PBX. These slips do not cause problems with voice calls, however fax and modem calls are impaired.

- The only universally applicable solution to this problem is to have one SmartNode BRI (or PRI) port connected to a reference clock . This solution will work with every PBX. The reference clock may come from an internal S-Bus on the PBX or from a PSTN connection

- In the case of one BRI port used for Voice, Fax and Modem over IP, many SmartNode models provide an extra BRI port for this refclock connection. E.g. the SN4552 and SN4630 series.

- If more then 4 active BRI ports are required, the solution can be provided with the SN2400 and 1-4 IC-4BRV interface cards.

- With some PBXs a reconfiguration of the trunk ports is possible that allows to deliver the ref clock to the SmartNode over the trunk line (PBX port Layer3 Usr (TE) but Layer 2 clock-master). This requires a reconfiguration of the PBX which is not possible on all PBX systems.

 
 What is FXS & FXO? ( Link to this FAQ )
 In analog telephony there are two common types of interfaces: FXS and FXO. FXS stands for "Foreign eXchange Subscriber" interface is used to connect subscriber equipment such as telephones, modems and Fax machines. FXO stands for "Foreign eXchange Office" is used to connect to the Public Switched Telephone Network (PSTN) and can also be used to connect to a PABX or multiplexer FXS port. Another third interface, which we will not discuss here, is known as an E&M (Ear & Mouth) interface which is used to provide a leased line or tie-line interface connection between PABX systems.

An FXO device plugs always plugs into an FXS line. You cannot plug FXS into FXS, or FXO into FXO; it will not work.

FXS Information

FXS is what is most commonly known as Plain Old Telephone Service (POTS). It is what your local phone company delivers to your home on a twisted pair. In other words, FXS looks a line from the telephone company switch (PSTN); it hooks to a telephone.

FXS interfaces provide to the subscriber:

  • Battery current and ring voltage
  • Dial tone (knows when to give dial-tome (seizure) when it sees current flowing from an FXO port closure.
  • Optional: CallerID (both caller number and name)
  • Optional: Call Waiting / Call Waiting ID
  • Optional: Message waiting indicator

FXS interfaces receive:

  • Hook Flash (to be notified of features, e.g, to set-up a three-way conference call or toggle between two incoming calls)
  • DTMF (touch tones)


FXS "alterts" an incoming call by:

  • Presenting ringing voltage to the line (attached device) – just like a PBX it does not and cannot pass any dialed digits.

FXS goes off-hook by:
  • Loop closure - Identifying that the line has been seized by the attached telephone going off hook. It can then receiving dialed digits (via DTMF).

Typically FXS devices do not indicate when they want to clear a call down, they rely on the two parties noticing that the call has ended (through the other party saying goodbye or the line going quiet) and each end device clearing itself down.

FXO Information

Your telephone is an FXO device and it connects to the FXS of the telephone company. Your phone provides on-hook/off-hook indication (loop closure) to the phone company. This is why you get a dial-tone when you pick up the phone.

FXO interfaces provide:

  • onhook/off-hook indication (loop closure)
  • HookFlash (to request features of PBX or PSTN, e.g., three-way conference calling) A quick loop closure or wink which is about a quarter of a second.
  • DTMF (touch tones)

FXO interfaces receive:

  • Dial tone as an indication from the FXS port that it achknowldeges the loop-closure.
  • Optional: Ring indication (voltage to ring the phone)
  • Optional: CallerID (both caller number and caller name)
  • Optional: Call Waiting Indicator (tone indicating a second incoming call)
  • Optional: Call Waiting ID (Caller ID of second incoming call)
  • Optional: Message waiting indicator (blinking light to indicate voice mail)

FXO makes a call by:

  • Seizing the telephone line (going off hook)
  • Dialing DTMF digits to identify the destination to call
  • Hanging up at the end of the call

FXO receives a call by:
  • Identifying when ringing voltage is being supplied by the PBX / CO switch (ringing the telephone)
  • Answers the call by “going off hook”. Call is then connected.
Examples
  • A standard analog (plain old telephone) is FXO
  • PBX/Switch lines from a PBX (that drive current) that you plug analog phones into are FXS
  • The PBX analog ports lines that plug into the CO are FXO
  • The SmartNode 2300 IC-4FXS card is FXS

 
 Is a reboot required when changing the mode of the ISDN interfaces (net/usr)? ( Link to this FAQ )
 Yes, even twice since the PLDs must be reprogrammed before parsing the CLI file.

 
 IGMP 
 How do I configure the IGMP functionalities ? ( Link to this FAQ )
 If you want route a multicast stream, you need to configure this on the ethernet interfaces.
There are two commands, one for the sending interface (to the clients) and the other for the receiving interface (eg. from the WAN).

interface IF_IP_WAN
ipaddress 192.168.1.20 255.255.255.0
icmp router-discovery
igmp interface-type proxy-upstream

interface IF_IP_LAN
ipaddress 10.0.0.1 255.255.255.0
icmp router-discovery
igmp interface-type proxy-downstream

Note: On interface IF_IP_LAN there are the receiving units connected, therefore the SmartNode must send the stream out there (downstream). Interface IF_IP_WAN is the receiving interface (upstream). 
 Licenses 
 Why do I need a License Key for Release 3.10? ( Link to this FAQ )
 SmartWare is the embedded software running on the SmartNodes. SmartWare offers a number of feature options such as QSIG, VPN and IP forwarding.
Up to SmartWare release 2.20 feature options had to be paid but where not keyed. Starting with release 3.10 a license key has to be installed to enable the feature options.

Note that some product bundles include some of the feature options i.e. SN1200/2VIL/UI includes IP Forwarding. The "I" in the model code stands for the IP forwarding license.

Q. Do I need a License Key for every SmartNode?
Yes the License Keys are specific to the feature option and the serial number of the SmartNode. The keys can not be transferred from one unit to another.

Q. Where can I buy Licenses?
Feature Options can be purchased through the regular SmartNode distribution channels.

Q. Where can I get License Keys for feature options purchased together with SmartWare 2.20?
The License Keys for SmartNodes delivered with SmartWare 2.20 can be requested using the following web form:
Liscense Request Form

 
 How do I install a license key? ( Link to this FAQ )
 License Key installation is described in the Software Configuration Guide 3.10 in the Chapter "Basic System Management" section "Managing Feature License Keys".

To install the licenses, simply copy the install command and license key ("install license 00010001gB...") from this message and paste them into an open CLI telnet or console session. Note that the CLI session must be in the "configure" mode.
You can verify that your license are installed using the following command:
  show licenses

Occasionally, e-mail clients can add spaces or tabs that will currupt a license key. If you have problems with the cut and paste method, you can alternatively copy a license file from your TFTP server as follows:
  copy tftp://tftp-server-ip-address/tftp-server-path/license-file licenses:

Example:
 copy tftp://192.168.1.1/sn4xxx_00a0ba001234.lic licenses:

  You should then see:
  Download...100% 
 When installing the License the SmartNode returns an error ( Link to this FAQ )
 There are two possible reasons for that. 1. You may be trying to install the wrong key. Make sure the keys you are installing match the serial number of the SmartNode. 2. You may have an early access build of SmartWare release 3.00 or 3.10. Please upgarde to a commercial release build number and try again.

 
 I have two different keys for the same feature on the same SmartNode. Which one is correct? ( Link to this FAQ )
 When a licence key is issued several times the resulting cipher key is different. However both keys will work and enable the same feature.

 
 The License does not work correctly on my SmartNode 4000 Series? ( Link to this FAQ )
 Some SmartNode 4000 series units have a serial number notation using colons ":" that do not work with the early access builds of Release 3.10. Upgrade first to the commercial version of 3.10 and the install the license keys.

 
 What happens if I do not install License Keys after the upgrade from Release 2.20 to 3.10? ( Link to this FAQ )
 IP forwarding will be disabled. That means you can still access the SmartNode on all IP interfaces but the SmartNode is not routing IP packets between interfaces. Also if you have been using other feature options such as VPN or QSIG these functions will be disabled as well. Q What does a License Key look like? you will receive a file for each SmartNode including the install commands for each purchased feature option and the actual license key string. Q. Where do I find the SmartNode serial number? The Serial number is marked on the product label on the bottom of the SmartNode. You can also find the serial number by login into the SmartNode and do a "show version".

login: administrator
password:mypassword
172.16.40.125>enable
172.16.40.125#show version

Productname: SN1400
Software Version: SmartWare R3.00 BUILD21244
Supplier:
Provider:
Subscriber:

Information for Slot 0:
SN1400 (Admin State: Application Started, Real State: Application Started) Hardware Version : 0001, 0001
Serial number: 100000020508
Software Version : SmartWare R3.00 BUILD21244

 
 Network Address Translation (NAT) 
 Do SmartNodes have a built-in NAPT application level gateway for H.323? ( Link to this FAQ )
 H.323 is a non-well behaving protocol in that it signals transport ports (RTP ports) inband in IP packets. When using NAPT (Network Address and Port Translation) this poses a problem since the ports are used by NAPT for address mapping. Thus H.323 does usually not pass a NAPT unless the NAPT is enhanced with H.323 aware functionality that leaves H.323 port ranges untouched.SmartNodes have NAPT but no H.323 aware application level gateway. However, it is possible to run NAPT and H.323 gateway concurrently on a SmartNode since NAPT affects only packets that are routed from IP interface to another (WAN to LAN).

 
 Can I do VoIP over NAT (Network Address Translation)? ( Link to this FAQ )
 Yes, If you are on a private network, your firewall or NAT (Network Address Translation) router must be “H.323 aware” or you'll need a SIP proxy if you are using SIP. To help determine if your LAN uses NAT, you can use a web browser and go to the following URL: http://www.patton.com/support/showmyip
This shows both the public and private IP address of your PC.

Note: H.323 aware routers and firewalls support "snooping", in which the H.323 control channel is continuously examined and session requests are authenticated. Once authenticated, the requested ports to be used for the H.323 session are opened for the duration of the conference. Upon termination of the conference, the ports are immediately closed by the firewall.
This is often referred to as an Application Level Gateway since this operation requires the firewall to be protocol-aware. Your H.323 aware router must support H.323v3. Both the firewall and the NAT/PAT software in your router must be H.323v3 aware.  
 NAT is not working anymore after I upgraded to 3.10 ( Link to this FAQ )
 If you are able to ping all interfaces of the SmartNode but NAT does not seem to be working, please verify that the IP routing license is installed. Without this license IP forwarding is blocked and therefore also the NAT does not work.
192.168.0.1#show licenses

IP Routing [iprouter]
License serial number: 546
Status: Active
 
 SIP 
 Why does my SIP-provider not accept my call ? ( Link to this FAQ )
 If SmartNode routes a call to a SIP-Proxy it sends in the SIP FROM-field the calling-party number. Either this calling-party number is provided from the connected telephone/PBX or it will be set as "anonymous" if not available.

This FROM-field is used by the proxy for authenticate purposes. This means the server challenges the SmartNode based on this number. SmartNode needs to have then a password configured in the SIP-gateway configuration. If the number in the FROM-field is different than the user-name in the gateway configuration the authentication fails and the proxy denies the call.

To avoid this you may need to set the FROM-field manually and with every call to the username as provided from the SIP-proxy operator. The command is to enter in the SIP-interface and is called:

address-translation outgoing-call from-header user-part fix (username) host-part call

NOTE: Please replace the (username) with the provided username from your provider. 
 How many SIP users can be supported on a SmartNode? ( Link to this FAQ )
 For all intents and purposes a maximum number of 100 "SIP users" can be supported on a SmartNode 
 Can I bind multiple SIP Gateways to the same IP Interface? ( Link to this FAQ )
 In some cases you may want to create multiple SIP gateways to subscribe to multiple SIP Telephony Services at the same time, or to seperate LAN SIP calls from Global/Internet SIP calls.

In order to bind multiple SIP gateways to the same IP interface the signaling port of the different gateways has to be different. Use the "call-signaling-port" command for this purpose

172.16.40.125(cfg)#gateway sip SIP1
172.16.40.125(gw-sip)[SIP1]#bind interface eth0
172.16.40.125(gw-sip)[SIP1]#no shutdown

172.16.40.125(cfg)#gateway sip SIP2
172.16.40.125(gw-sip)[SIP2]#bind interface eth0
172.16.40.125(gw-sip)[SIP2]#call-signaling-port 5062
172.16.40.125(gw-sip)[SIP2]#no shutdown

If you do not change the signalling port you will get the following error message when you try to bind or activate the second gateway:

% ANOTHER GATEWAY IS ALREADY BOUND TO THE SAME PORT

Note: The ports are allocated even if a gateway is in shutdown. You must still use different signalling ports on each gateway!

Note: The signalling port numbers must be even values e.g. 5060, 5062, 5064 etc.
As an alternative, you may want to create different SIP "services" within one gateway - this allows to have mulitple virtual gateways on the same interface, using all the same call signaling port. 
 Supplementary Calling Features 
 Why does the SmartNode not forward the AOC-d messages from Cirpack to ISDN ( Link to this FAQ )
 In case a Cirpack SIP server is used and AOC-d is enabled, it may be that the first AOC message is sent to the ISDN equipment only.

This has to do that the Cirpack sends amount raises and not the amount sum.

To fix this following needs to be done on the Cirpack.

Set the parameter SendITX to "SUM"
 
 Why does the Siemens Hipath PBX not show the Caller-Name with Qsig as signaling protocol ( Link to this FAQ )
 
The only Siemens Qsig Protocol, which is interoperable with the SmartNode products is ISO-QSIG.

You must choose as QSIG version ISO-QSIG.

All other Siemens Qsig versions show restrictions in the supported supplementary services, such as the caller name.
 
 How does a SmartNode CDR looks like? ( Link to this FAQ )
 
Below you may find an example of a SmartNode CDR stream recorded in a RADIUS server.


Wed Dec 20 09:03:17 2006
   Acct-Status-Type = Start
   Acct-Session-Id = "SIP_PROXY.in-050505"
   Called-Station-Id = "0041319852525"
   Calling-Station-Id = "0041319852555"
   Calling-presentation-indicator = "allowed"
   Calling-screening-indicator = "network provided"
   Setup-time = "Wed Dec 20 09:03:16 2006"
   NAS-IP-Address = 192.168.1.182
   Client-IP-Address = 192.168.1.182
   Acct-Unique-Session-Id = "fce3c77c99ee8cf1"
   Timestamp = 1166601797

Wed Dec 20 09:03:37 2006
   Acct-Status-Type = Alive
   Acct-Session-Time = 0
   Acct-Session-Id = "SIP_PROXY.in-050505"
   Called-Station-Id = "0041319852525"
   Calling-Station-Id = "0041319852555"
   Calling-presentation-indicator = "allowed"
   Calling-screening-indicator = "network provided"
   Called-ip-address = 192.168.1.180
   Called-Unique-Id = "202f48e5b0cbb4bc3ee4738a6d95002a@192.168.1.182"
   Setup-time = "Wed Dec 20 09:03:16 2006"
   Connect-time = "Wed Dec 20 09:03:36 2006"
   NAS-IP-Address = 192.168.1.182
   Client-IP-Address = 192.168.1.182
   Acct-Unique-Session-Id = "fce3c77c99ee8cf1"
   Timestamp = 1166601817

Wed Dec 20 09:04:07 2006
   Acct-Status-Type = Stop
   Acct-Session-Time = 31
   Disconnect-source = "terminator"
   Disconnect-cause = 16
   Acct-Session-Id = "SIP_PROXY.in-050505"
   Called-Station-Id = "0041319852525"
   Calling-Station-Id = "0041319852555"
   Calling-presentation-indicator = "allowed"
   Calling-screening-indicator = "network provided"
   Called-ip-address = 192.168.1.180
   Called-Unique-Id = "202f48e5b0cbb4bc3ee4738a6d95002a@192.168.1.182"
   Setup-time = "Wed Dec 20 09:03:16 2006"
   Connect-time = "Wed Dec 20 09:03:36 2006"
   Disconnect-time = "Wed Dec 20 09:04:07 2006"
   NAS-IP-Address = 192.168.1.182
   Client-IP-Address = 192.168.1.182
   Acct-Unique-Session-Id = "fce3c77c99ee8cf1"
   Timestamp = 1166601847
 
 How do I send a hook FLASH to a SIP Provider to use services like three-way conferencing? ( Link to this FAQ )
 By default the SmartNode handles hook FLASH events by itself, i.e. a call is held locally, and if it is transferred, it is looped locally as well.

If you want to transmit the DTMF towards the far end, you must disable the additional servivces on the fxs interfaces.

Example:

interface fxs IF_FXS_00
no call-hold
no call-waiting
no call-transfer
no additional-call-offering

This is often used in fxs/fxo line extensions.
To transport a hook flash to the SIP network, you also need to set the option in your voip profile.
Example:

profile voip default
dtmf-relay rtp


Options:

dtmf-relay rtp - DTMF's and flash are transmitted by RFC2833 RTP events. This is the default setting.
dtmf-relay signaling - DTMF's and flash are transmitted by SIP INFO messages.
Regardless of what is configured, the SmartNode accepts incoming events of both methods.  
 How can a FLASH be relayed from an FXS port to the PSTN on an FXO port? ( Link to this FAQ )
 A common application is to accept calls from a PSTN on an FXO port and then ring a telephone connected to a FXS port. In order to send a FLASH out the FXO port to the PSTN, you must disable all supplementary calling features on the FXS interface. For example:

interface fxs IF-FXS-PHONE1
route call dest-table TAB-OUTGOING-LINE1
no call-hold
no call-waiting
no additional-call-offering
caller-id-presentation mid-ring
use profile tone-set US
 
 How can I do Call Transfer and FLASH codes on the SmartNode? ( Link to this FAQ )
 SmartWare FLASH Codes

-FLASH 0 - keep current, reject incoming
-FLASH 1 - drop current, accept incoming
-FLASH 2 - hold current, accept incoming
To toggle between the active and the held call, press flash-hook, followed by the "2" key.

Additional Call Offering

To enable aditional call offering, configure the fxs port of the SN with the command: additional-call-offering

1) Press FLASH, then the first call is placed on hold and you will hear a new dial tone.
2) Dial the number of the second call.
3) If you press FLASH, you may change between the two calls.
4) When you hang-up on the phone, the two other parties are connected together. Sorry, three way conferencing is not yet supported. 
 Upgrading/TFTP 
 How can I upgrade a SN4552 from 5.T to 5.1 ( Link to this FAQ )
 If the message OUT OF MEMORY appears, you may do the following

1. Store the current running-config in a separate file
2. copy the config which is attached below, to the startup-config in the SmartNode and reload it. Be sure about the ip addresses. If they have to be set fix, enter the required values to the file before upgrading
3. downgrade the SmartNode to R4.2 by using the CLI
4. After the reload, upgrade to 5.1, also by CLI



#----factory configuration----#
dns-relay
sntp-client
sntp-client server primary 129.132.2.21 port 123
version 4

system

ic voice 0
low-bitrate-codec g729

profile napt NAPT_WAN

profile dhcp-server DHCPS_LAN
network 192.168.1.0 255.255.255.0
include 1 192.168.1.10 192.168.1.99
lease 2 hours
default-router 1 192.168.1.1
domain-name-server 1 192.168.1.1

context ip router

interface WAN
ipaddress dhcp
use profile napt NAPT_WAN
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface LAN
ipaddress 192.168.1.1 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context ip router
dhcp-server use DHCPS_LAN

port ethernet 0 0
medium auto
encapsulation ip
bind interface WAN router
no shutdown

port ethernet 0 1
medium auto
encapsulation ip
bind interface LAN router
no shutdown

 
 Using Encrypted TFTP ( Link to this FAQ )
 

Encrypted Configuration Download

- An external encryption tool on the PC is used to encrypt the configuration file:

enctool encrypt <plain-config-file> <enc-config-file> [<key>]

- The encrypted confiugration file can then be downloaded with TFTP triggered by

- the CLI copy command:

copy tftp://<host>/<path> <config-file>

- Auto Provisioning

- SNMP

- HTTP

- On the SmartNode the encryption is detected and the configuration file is automatically decrypted

before stored to flash.

- A custom encryption key can be

- downloaded to the SmartWare

- specified with the PC encryption tool

- The encryption key may include the MAC address and/or serial number of the SmartNode using the

placeholders $(system.mac) and $(system.serial) resp.

- An encrypted configuration file can be uploaded to a TFTP server on request, specifying the encrypted

flag:

copy <config-file> tftp://<host>/<path> encrypted

- On the PC the encryption tool can be used to decrypt the file:

enctool decrypt <enc-config-file> <plain-config-file> [<key>]

- A log file lists the last up/downloads:

show log file-transfer


Use Cases

Install a custom encryption key (optional)

You can install a custom encryption key with the SmartNode. The encryption key is used to automatically

decrypt an encrypted configuration file that is downloaded later. A default encryption key is already

installed on the SmartNode.

To install an encryption key you have to create a file on your TFTP server that contains the key. Then you

have to download this key file to the SmartNode using the ‘copy’ command of the SmartNode:

The key file shall contain a key string of at most 24 characters on a single line. Spaces, tabs and LF/CR

characters are trimmed. The key must not contain LF/CR or the null character and must not start or end

with a space or tab. If the key contains more than 24 characters, only the first 24 characters are

considered.

Part Nr. 80-0165, Rev. 1.13 12-07-05 49/54

The key may contain variables that are resolved when the key file is downloaded to a SmartNode. Using

this mechanism you can specify device-specific encryption keys. We currently support the following

variables:

- $(system.mac): The MAC address of the first ethernet port. Execute the show port ethernet

command on a SmartNode to display the MAC address of a SmartNode. This value without the colon

separators and with all lower-case hexadecimal letters is used instad of the variable on the SmartNode.

- $(system.serial): The serial number of the SmartNode. Execte the show version command on

the SmartNode to display the serial number.

When your key file contains the following line…

123$(system.serial)abc$(system.mac)XYZ

show port ethernet shows the following…

Ethernet Configuration

-------------------------------------

Port : ethernet 0 0 0

State : OPENED

MAC Address : 00:0C:F1:87:D9:09

Speed : 10MBit/s

Duplex : Half

Encapsulation : ip

Binding : interface eth0 router

and show version the following….

Productname : SN1200

Software Version : R3.20 TB2005-06-24_MEYER SIP

Supplier :

Provider :

Subscriber :

Information for Slot 0:

SN1200

Hardware Version : 0004, 0001

Serial number : 100000020002

Software Version : R3.20 TB2005-06-24_MEYER SIP

the encryption key on this SmartNode will be interpreted as…

123100000020002abc000cf187d909XYZ

Then you have to download the created key file to the SmartNode. Open a telnet session and type in the

following commands:


>enable

#copy tftp://<ip>/<path> key:

where <ip> is the IP address of your TFTP server and <path> is the path to the key file relative to the

TFTP root.


Encrypt a configuration file

Use the encryption tool to encrypt a configuration file on your PC. Therefore you have to enter the

following command.


enctool encrypt <plain-file> <encrypted-file> [<key>]

where <plain-file> is the path of the non-encrypted input configuration file and <encrypted-file> is the path

of the encrypted output configuration file. <key> specifies the encryption key which shall be used to

encrypt the configuration file. If ommitted the default key is used.


Download an encrypted configuration file

Now you can download the configuration file as usual using the CLI copy-command, the autoprovisioning

feature, HTTP or SNMP download. The SmartNode automatically detects that a downloaded

file is encrypted and tries to decrypt the file using the pre-installed key.


Upload an encrypted configuration file

The SmartNode immediately decrypts a configuration file after downloading it. This is the configuration

file is stored non-encrypted in the flash memory. Thus when you upload a configuration it is uploaded

non-encrypted.

You may upload an encrypted configuration file specifying the encrypted flag at the end of the copy

command:


#copy startup-config tftp://<ip>/<path> encrpted

This encrypts the configuration file before sending it to the TFTP server. Use the enctool decrypt

command on the PC to regain the original configuration.


File Transfer Logs

We introduced an additional log file that stores the history of all file transfers (up to 50 entries). To show

all recently executed file transfer operations enter the following command:


#show log file-transfer

 
 How can I update a SmartNode 4xxx in redboot mode ? ( Link to this FAQ )
 If a firmware update fails the SmartNode will enter the redboot mode. Connect a computer to the console interface and restart the SmartNode. You will see the Prompt "RedBoot".

    Download the required .bin file at

    Image for SN4552; SN4554; SN4562;S-DTA
    Image for SN4xxx_FXS_FXO
    Image for SN4xxx_BRI
    Image for SN4960

    Default IP Address WAN port 172.16.40.1 netmask 255.255.0.0 for SmartNodes without console port

  • 1. Type in 'fis list'

  • 2. If there is an entry type 'fis delete -n 1'

  • Optional
  • 3. Set the IP address using
    'ip_address -l (ip-address)/(masklength) -g (gateway-address)'

  • 4.1 Load the new firmware for the SN4xxx Series (SN4110, SN4520, SN4600, SN4960)
    'load -r -b 0x1800100 -h (ip-address) /image.bin'

    4.2 Load the new firmware for the SN4552, SN4554, SN4562, S-DTA
    'load -r -b 0xc00100 -h (ip-address) /image.bin'

  • Replace (ip-address) with the ip-address of your tftp-server.

  • 5. Program the firmware into the flash with the command 'fis create' when prompted type 'yes'

  • 6. Load the new firmware with 'fis load'

  • 7. Restart the smartnode with 'go'


  • Reset and how to access the bootloader/redboot
    The reset button has three functions:

    • Restart the unit with the current startup configuration—Press (for less than 1 second) and release the Reset button to restart the unit with the current startup configuration.

    • Restart the unit with factory default configuration—Press the Reset button for 5 seconds until the Power LED starts blinking to restart the unit with factory default configuration.

    • Restart the unit in bootloader mode (to be used only by trained SmartNode technicians)— Starting with the unit powered off, press and hold the Reset button as you apply power to the unit. Release the Reset button when the Power LED starts blinking so the unit will enter bootloader mode.
 
 How can I update the SmartNode 1xxx /2xxx in bootloader mode ? ( Link to this FAQ )
 If a firmware update fails the SmartNode will enter the bootloader mode. You can access the SmartNode ONLY with ethernet on the last known ip-address. The username is "admin" and the password "inalp".

After successful login you can download the new firmware with these commands:
sd 1--> enter the download agent
ssip (ip-address) --> replace (ip-address) with the ip-address of your tftp-server. eg. 192.168.1.100
download (path) b --> replace (path) with the path (if necessary). Please note the space and the letter b !
boot --> boots the SmartNode with the new firmware.  
 What happens if the software upgrade on a SmartNode fails? ( Link to this FAQ )
 Each SmartNode is equipped with a bootloader application. If an upgrade fails and no valid firmware is available on the system the SmartNode will start in this bootloader mode. The bootloader will allow you to install a new firmware.
Please refer to the user documentation on how to operate in bootloader mode.
Note that the bootloader can not be replaced. 
 Where can I get a TFTP Server to load in my configuration or upgrade my SmartNode Software? ( Link to this FAQ )
 We recommed two good TFTP Servers: First the Solar Winds TFTP Server is found on the SmartWare CD. You can also download it from the SolarWinds web site. Use SolarWinds version 3.0.9 or higher. SolarWinds is especially nice if you want it to run a TFTP server all the time as an NT service.

Additionally,
TFTPD is available on-line for easy download. Use version 2.60 or higher. TFTPD32 a very small, fast, easy to use and contains a TFTP Client, TFTP Server, Syslog, and SNTP server which are all useful for testing. TFTPD32 is a stand-alone executable that is quick to get running.

Thank you SolarWinds and Philippe Jounin! See their web sites for more great software.

 
 The software download fails in the middle of the process. Why? ( Link to this FAQ )
 Some firewalls may reset a session when it takes too much time to complete. On low speed links the software download via TFTP may indeed take a long time and thus the firewall on the link may prematurely reset the session.

 
 Is there a tool to convert SmartWare R2.20 configurations to R3.10 configurations? ( Link to this FAQ )
 Yes, Release 3.10 adds SIP and a lot of new session router features. The configuration must be converted. An easy to use on-line tool and instructions can be found at Smart Convert

 
 VPN 
 Why is the DES/3DES VPN encryption Key, displayed in the running-config, different to my input? ( Link to this FAQ )
 Quote: DES Key Parity from Phil Harding

A single DES key is 64 bits (8 bytes) long, however the actual key material used by the DES algorithm amounts to only 56 bits in length. The least significant bit of each byte is a parity bit, and should be set such that there is always an odd number of bits set (1's) in each key byte. Only the 7 most significant bits of each byte are effective for security purposes.


Therefore the SmartNode change the entered DES/3DES encryption key to match to above explained rule.  
 Can I do encrypted VoIP calls with the SmartNode IPSec? ( Link to this FAQ )
 Yes, with SmartWare software releases dated 3/1/06 and later. For earlier relases, VoIP calls terminated on the SmartNode route the RTP outside the VPN tunnel. A VPN feature license has to be installed for this feature to work. 
 How many VPN tunnels can I configure on a SmartNode? ( Link to this FAQ )
 The number of VPN tunnels that you are able to create is only limited to the amount of available RAM. The SmartNode does not have a preset limitation of VPN tunnels. In practice the SmartNode will support a minimum of 10 VPN tunnels but also 100 tunnels are working. Keep in mind that with a large number of tunnels the available bandwidth for each tunnel is reduced. Note that you have to install the VPN license key to have access to the VPN configuration. 
 Wan and DSL 
 How can I connect with PPPoE to T-DSL (Germany) ? ( Link to this FAQ )
 T-Online uses the username in three steps.

a) 12 digits connect-number
b) 9 to 12 digits T-Online-number (if <12 digits add a hash (#) in front)
c) user-suffix is always 0001 folowing by @tonline.de

The username is therefore:
aaaaaaaaaaaabbbbbbbbbbbbcccc@t-online.de

---------------------------------------
profile napt NAPT

profile ppp default
mru min 1508 max 1508

context ip router
interface IF_IP_WAN
ipaddress unnumbered
mtu 800
point-to-point
icmp router-discovery
use profile napt NAPT
tcp adjust-mss rx 1000
tcp adjust-mss tx 1000

context ip router
route 0.0.0.0 0.0.0.0 IF_IP_WAN 0

subscriber ppp SUB_PPPOE_SUNRISE
dial out
authentication chap
authentication pap
identification outbound <username> password <password>
bind interface IF_IP_WAN router

port ethernet 0 0
encapsulation pppoe

pppoe

session SES_SUNRISE
bind subscriber SUB_PPPOE_SUNRISE
no shutdown

port ethernet 0 0
no shutdown
 
 How can I connect with PPPoE to Sunrise (Switzerland) ? ( Link to this FAQ )
 Use this configuration and adapt the username and the password to your requirements.
-----------------------------------------------

profile napt NAPT

profile ppp default
  mru min 1508 max 1508

context ip router
 interface IF_IP_WAN
  ipaddress unnumbered
  mtu 800
  point-to-point
  icmp router-discovery
  use profile napt NAPT
  tcp adjust-mss rx 1000
  tcp adjust-mss tx 1000

context ip router
  route 0.0.0.0 0.0.0.0 IF_IP_WAN 0

subscriber ppp SUB_PPPOE_SUNRISE
  dial out
  authentication chap
  authentication pap
  identification outbound <username> password <password>
  bind interface IF_IP_WAN router

port ethernet 0 0
  encapsulation pppoe

  pppoe

    session SES_SUNRISE
      bind subscriber SUB_PPPOE_SUNRISE
      no shutdown

port ethernet 0 0
  no shutdown
 
 



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