SmartNode™ Interface Card IC-T1V Interface Card for ISDN PRI Interface
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Features & Benefits
1 ISDN balanced 100Ohm T1, RJ45
23 calls per PRI
G.168 echo cancellation
Support of various voice codecs: G.711 A-Law/µ-Law, G.723.1, G.729A, Transparent
Voice Signalling: Euro ISDN DSS-1 / ETSI PRI Q.931 / Q-SIG (PSS-1)
EMC & CE Compliant
1-year warranty
Optional Upgrade: 2-5 year Warranty
Optional Upgrade: 1-5 year Enhanced Warranty (includes free Advance Replacements)
Overview
The IC-T1V is a PMC standard compatible interface card for the SmartNode 2000 series. It provides one ISDN PRI interface and connects through a PCI packet and PCM circuit interface to the SmartNode base unit.
The IC-T1V is a high performance ISDN networking card that supports up to 23 simultaneous voice or fax calls. It meets the requirements of medium and large sized enterprises that want to connect their ISDN equipment, such as an ISDN PBX, via a PRI interface to a multi-service IP network.
The ISDN PRI interfaces can be configured as user or network side interfaces. Its on board dedicated micro-controller and DSPs off load the CPU of the SmartNode 2000 base unit and hence guarantee the conversion of voice/fax circuits into related IP-packets in real-time with minimal delay and jitter values.
Configuration Example The example shows the connection of a PBX via an IC-T1V Card into an IP network.
How to extend dial tone (dial tone extension) from one location to another location without a SIP server.
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
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
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
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
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.
Telephones
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.
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 -
Select Line Key Settings
Pick a free line key and select "one touch dial"
Enter the DNS name or IP address of the Smartnode gateway that has the PSTN FXO connection
Click on "submit" to save the changes
SmartNode VoIP/ToIP
Call Routing
How can I remove or restrict Caller-ID (CLIP)?
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
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:
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#
Debug and Logging
How do I use the ACL debugs to debug a VPN Connection?
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.*
SIP
How many SIP users can be supported on a SmartNode?
For all intents and purposes a maximum number of 100 "SIP users" can be supported on a SmartNode
Upgrading/TFTP
What happens if the software upgrade on a SmartNode fails?
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.
VPN
Can I do encrypted VoIP calls with the SmartNode IPSec?
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.
Gateways 