The model 3201 and model 3241 G.SHDSL DiamondLink routers offer high-speed standards-based G.SHDSL for long reach network extension. The model 3201 provides rates up to 2.3 Mbps and the model 3241 offers rates up to 4.6 Mbps. Connect them back-to-back or use them with one of Patton's concentrator solutions such as, The Patton model 3096RC T-DAC or the Patton modem 3224 IP DSLAM. In addition, the DialmondLink routers can be used as CPE with third party DSLAMS. The model DiamondLink routers offer a standards-based DSL that supports the fundamental and advanced access requirements needed in the market today:
1. Provide a routed packet-based CPE with IP centric services and functionality.
2. Support compatibility with the ability to connect to standard DSLAMs using G.SHDSL.
3. Manage devices from any workstation or PC across the Internet with Patton's web-based network management systems or standard SNMP management.
G.SHDSL is the newest standard for symmetric DSL. It works over 2-wires and improves both the distances achievable by DSL and the spectral compatibility with other services often present in a twisted pair bundle. The ITU standard for G.SHDSL is G.911.2. Annex A describs the transmission and performance requirements for North America and Annex B describes performance and transmission requirements for Europe. SHDSL has been standardized by three different standardization bodies: ANSI(T1E1.4/2001-174) for North America, ETSI(TS101524) for Europe and ITU-T(G.991.2) worldwide.
Must the DiamondLink G.HDSL Routers or 3086 IADs be used in pairs? ( Link to this FAQ )
In order to establish a G.HSDSL connection, there must be a modem on each end of the link. This does not mean that a DiamondLink/IAD modem must be on each end. The DiamondLink/IAD modem can be used with the following options on the other end of the twisted pair:
1. Another DiamondLink G.HDSL Router
2. Another Patton G.SHDSL CPE or IAD device such as the Patton Model 3086 G.SHDSL IAD
3. One of Patton;s G.SHDSL DSLAM options such as the 2096RC Digital Access Concentrator or the 3224 ipDSLAM.
4. A third party G.SHDSL CPE modem.
5. A third party G.HSDSL DSLAM.
As long as the third party modems and DSLAMs implement the G.SHDSL standard, the DiamondLink router can be used in conjunction with them across the twisted pair. Operation above the standardized 2.6 Mbps is dependent upon vendor implementation and interoperability at these higher rates will vary.
Do the G.SHDSL products support Network Address Translations(NAT)? ( Link to this FAQ )
Yes, both of these models support NAT (RFC3022) with Network Address Port Translation(NAPT). They also support MultiNat with 1:1, Many:1, Many:many mapping as well as Port/IP redirection and mapping.
Yes. They automatically learn, age and filter 1,024 source addresses. Destination addresses of incoming frames are compared with the Source Address in the address table and discarded if an entry exists; otherwise, they are forwarded over the DSL Extension.
What are the ethernet capabilities of the 3201,3241 and 3086? ( Link to this FAQ )
They are capable of connecting to Ethernet Hubs and Switches, remote PCs and any other network enabled device. The MDI-X switch is used to allow the units to connect to either a hub(DCE) or PC(DTE) device eliminating the neede to decide whether a straight-through or cross-over connection is needed. With a simple push of the MDI-X switch, the unit itself will change the connection from straight-through to a crossover or back again. The Ethernet port automatically senses 10 or 100 Base-T Ethernet connections. Additionally they can automatically sense full or half-duplex Ethernet connections.
What are some typical applications for the Model 3201 and Model 3241 DialmondLink Routers? ( Link to this FAQ )
Some typical applications are:
1. Enterprise/Campus users deploying point-to-point network extensions
2. PTTs who want to offer standards-based DSL service using 3201 or 3241 as CPE to DSLAMs.
3. ILECs/CLECs connecting service to secondary metropolitan areas where a full-blown DSLAM is not cost-effective or practical.
4. ISPs wishing to expand their service offering advanced IP feature set in a small, flexible, low-cost DSL router
5. To deliver data services to customers within a office complex.
What HDSL standards do the DiamondLink G.HSDSL routers And IpRocketlink IAD use? ( Link to this FAQ )
The DiamondLink G.HSDSL routers supports all the SHDSL standards developed by ANSI (T1E1.4/2001-174, ETSI(TS 101524) and ITU-T(G.991.2). The routers offer selectable support for both Annex A (North America) and Annex B(Europe). They implement TC-PAM 16 as the standard line coding.
Is surge protection provided with the DiamondLink routers? ( Link to this FAQ )
Yes, the DiamondLink routers are protected in compliance with all FCC Part 68 and UL1950 specifications. In addition, the DiamondLink routers are designed to meet ITU-T recommendation K12.
These models can easily be configured using several methods. The are user selectable to allow ATN, PPP or HDLC WAN data link connections and allow a variety of local and remote configuration and management options:
1. Web-based configuration via embedded web server. You just need a web browser to access the web pages.
2. CLI menu for configuration, management and diagnostics
3. Local/Remote CLI (VT-100 or telnet)
4. SNMPV1(RFC 1157) MIB II (RFC 1213)
5. Logging via SYSLOG and Vt-100 console. Console port set at 9600bps 8N1, no flow control
6. EOC access for End-to-end management, configuration and control.
Additionally the 3086 IAD can be configured for DIP switch control for serial only environments.
What protocols are supported by the G.SHDSL products? ( Link to this FAQ )
Point-toPoint Protocol over HDLC
PPPoA(RFC2364) Point-to-Point Protocol over ATM
PPPoE(RFC 2516) Client for autonomous network connection. Eliminates the requirement of installing client software on a local PC and allows sharing of the connection across a LAN.
User configurable PPP PAP (RFC1661) or CHAP(RFC1994) authentication.
Multiprotocol over ATM AAL5 and Multiprotocol Bridges encapsulation RFC 2684 (Formerly RFC 1483) and RFC 1577 Classical IP over ATM. Default RFC-1483 route mode. Logical Link Control (LLC)/Subnetwork Access Protocol (SNAP) enxapsulation. Default VC mux mode.
ATM UNI 3.0, 3.1 and 4.0 signaling ATM QoS with UBR, CBR, nrt-VBR, and rt-VBR and per-VC queuing ans haping. IISP V.1.0 Q.2931 UNI L3 and Q.2971 UNI L3 support.
LAN Emulation Client (LEC) V.1 with LEC via PVC or ILMI connection.
Peak cell rate shaping on a per-VCC basis up to 32 active VCCs across VPI 0-255, VCI 0-65525. Single default PVC:8/35 with PCR=5,500 cells
I.610 OAM network management including AIS/RDI, loop-back and performance monitoring.
Enhanced ILMI 4.0 for auto-configuration of ATM PVCs.
FRF.12 Frame RElay Fragmentation support, LMI for Frame RElay PVC Link Management, FRF.5 Frame Relay to ATM Network internetworking, and FRF.8 Frame Relay to ATM Service Internetworking.
Complete internetworking with IP (RFC741), TCP(RFC793), UDP(RFC768), ICMP(RFC950), ARP(RFC826)
IP router with RIP (RFC1058), RIPv2 (RFC2453) for up to 64 static routes
Built-in Ping and Traceroute facilities
Integrated DHCP Server(RFC2131)
DHCP relay agent (RFC2132/RFC1542) with 8 individual address pools.
DNS Relay with primary and secondary Name Server selection.
NAT (RFC3022) with Network Address Port Translation(NAPT), MultiNat with 1:1, Many:1, Many:Many mapping, Port/IP redirection and mapping.
Based on the ETSI and International Telecommunications Union (ITU) G.SHDSL G.991.2 standard, the Patton 3086 ipRocketLink enables nx64 (n=3..36 with software key to offer up to n = 72) over a single pair of wires and presents a unique dual port user interface.
The 3086 is available in a variety of version to provide customers with two local data interfaces. The 3086 line offers models that provide both a Synchronous Serial port (either V.35, X.21 or T1/E1) and Ethernet port and incorporates a high-speed business class router. All 3086 models can transport data over the Ethernet and the WAN ports at the same time. The 3086 offers both V.35/X.21/T1/E1 WAN interfaces and 10/100 Base-T Ethernet ports. The sync serial port is available in either V.35 or X.21 versions. The ethernet port gives access to any IP network via the DSL link using ATM, PPP, HDLC or Frame Relay transport.
What is the primary application for the Model 3086 ipRocketLink? ( Link to this FAQ )
They can be used in Internet applications(connection to ISP) or in the connection of remote branches using DSL access and IP/FR/ATM/PPP. Applications include:
Internet/Extranet Access
IP/FR and TDM Access (using TDM or FRF.5/FRF.8 internetworking)
IP/FR and Voice over DSL
Metro Intranet Access
What are the interfaces available on the model 3086s? ( Link to this FAQ )
Each 3086 comes with all of the following interfaces:
Control Port: An EIA-561 Rj-45 support external RS-232, VT-100 CLI configuration port
G.SHDSL Port: A two-contact shielded RJ-111F(RJ-45F available)
Ethernet Port: Eight contact shielded RJ-45F, 10/100 Mbps (auto-sensing with support for full/half duplex operation.
In addition each 3086 coms with one of the following interfaces:
X.21 on a DB-15F with n x 64 kps support up to 2.3 Mbps - DTE/DCE configurable
V.35 on a M34F with a nx64 kbps support up to 2.3Mbps
V.35 on a DB-25F with nx64 kbps support up to 2.3 Mbps
E1 (G.703/G.704) on a RJ-45 and Dual BNC with nx64 support up to 2.048 Mbps
T1 on Rj-45 with a nx64 kbps support up to 1.544 Mbps
Microproducts and General Communications Equipment
General
How to check if Microsoft's USB fix is installed on my computer? ( Link to this FAQ )
You can check whether the Microsoft fix was installed through:
· 'My Computer' - 'Properties' - 'Hardware' - 'Device Manager' - 'Universal-Serial-Bus-Controller' - choose one of the controllers and select 'Properties' - 'Driver' - 'Driver Details'.
· The correct fix is: usbuhci.sys version is 5.1.2600.14.
· The incorrect fix is: usbuhci.sys version is 5.1.2600.0.
· Note: If WinXP is installed with SP1 the usbuhci.sys version should be: 5.1.2600.1038.
What services do I need from my phone company to use modem on hold? ( Link to this FAQ )
Modem On Hold: Allows you to receive incoming calls while keeping your Internet connection on hold. Note: you need to have the "call waiting" feature enabled on your phone line. Your phone company can arrange to have this installed, if not currently available.
Outgoing Calls: Allows you to make outgoing calls while keeping your Internet connection on hold. Note: You need to have the "three-way call" feature on your phone line. Your phone company can arrange to have this installed, if not currently available.
What do I do if I have a USB modem on WinXP and the modem frequently disconnects? ( Link to this FAQ )
Two Wire Operation of Models 1004A, 1009, 2085 and 2089? ( Link to this FAQ )
For Half-Duplex 2-wire operation, Please confirm that the communications program being used by the computers controls Request To Send (RTS). This is very important, when a Converter detects an ON (+ voltage) for RTS the transmitter is switched from HIGH IMPEDANCE to LOW IMPEDANCE. If two or more Computers have RTS - ON the line will be clamped and there will be NO TRANSMISSION possible.
The Converters operates HALF-DUPLEX only when they are configured for
2-wire operation. For Half-Duplex operation all Computers must have RTS-OFF, a negative voltage for RTS.
Typical Half Duplex communications will function as follows :
A Master unit will Poll with addresses of slave units.
The Master unit turns ON RTS sends the address of a slave unit, then turn RTS OFF.
The Slave unit receives it’s address, turns ON RTS sends the information then turns OFF RTS.
Please use the following configuration switch settings for 2-wire operation of the Model 1004A and 2085
S1-1, 2, 3 and 4 OFF
S1-5 ON for RTS control
S1-6 one unit ON the other OFF
S1-7, and 8 ON for 2-wire operation.
Please use the following configuration switch settings for 2-wire operation of the Model 1008 and 2089
S1-1, 2 and 3 ON
S1-4 OFF
S2-1 and 2 ON
S2-3 and 4 OFF
The 2-wire twisted pair circuit should be connected:
XMT+ ---- XMT+
XMT- ---- XMT-
What is the meaning of DCE (data communication equipment) and DTE (data termination equipment)? ( Link to this FAQ )
DCE Data Communications Equipment, is typically a Modem or other device that connects to a Network. DTE Data Terminal Equipment, is typically a Computer.
What is the different between wired as DCE and DTE? ( Link to this FAQ )
DCE and DTE reference is used to identify the source or origination of signals. A local DTE (PC) connects to a DCE (Modem), the Modem connects to a Network, the remote end of the Network has a second Modem used to connect to the remote PC.
A Null Modem is used to connect the one DCE device to another. A Null Modem crossover connects the Receive and Transmit Data and the Control Signals.
What pin-out should be used to connect DCE to DTE? ( Link to this FAQ )
The cable should be connected straight through.
How can the distance limitations on any unit be extended? ( Link to this FAQ )
Distance limitations on any unit that operates in pairs can be extended using a tail circuit. This involves using two pairs of the unit daisy-chained together through the DTE Interface. However, since the units are both DCE, a crossover cable must be used. We have run tests on some units using 3 repeaters which extended the distance 4 times the original.
Why won't my RS485 or RS422 device work properly, and which signals do Patton use as + and -? ( Link to this FAQ )
There is confusion regarding the polarity of the RS485 and RS422 transmit and receive signal pairs. The RS-485 and RS-422 specifications define the transmit and receive signal pairs as XMT-A, XMT-B, RCV-A, and RCV-B. Unfortunately most manufacturers have ARBITRARILY assigned XMT-A as either XMT+ or XMT-. Similarly for the RCV signal pair. The specification never defines the A and B signals as either negative or positive. They only state that in the mark condition, XMT-A is more negative than XMT-B. This is only a relative definition. What does this mean? You might want to swap the polarity if your application does not function properly. If the other RS485 or RS422 device does use -A and -B for the polarity identification, note that the Patton signals correspond as follows: XMT+ -----> XMT-A XMT- -----> XMT-B RCV- -----> RCV-B RCV+ -----> RCV-A. This only applies if the remote end is not the same as the local end. If you are using any of these converters on both ends: 2089, 2085, 222N, 222N9, 2084, 2086, and their corresponding rack cards, this does NOT apply.
What's required for a Short Range Multipoint Modem to operate Two Wire? ( Link to this FAQ )
First of all, the communication equipment must control Request To Send (RTS). This is very important, when a modem detects an ON (+ voltage) on Pin 4 (RTS) it switches the transmitter from High Impedance to Low Impedance. If two or more pieces of equipment have RTS - ON, the line will be clamped and there will be no transmission possible. These modems operate in Half-Duplex only when they are configured for Multi-Point operation. For Half-Duplex operation all equipment must have RTS-OFF, a negative voltage on Pin 4 of the RS-232/V.24 serial port. Typical Half Duplex communications will function as follows:
A Master unit will Poll with addresses of slave units. The Master unit turns ON RTS sends the address of a slave unit, turn RTS OFF. The Slave unit receives it's address, turns ON RTS sends the information then turns OFF RTS.
Why doesn't my Short Range Modem work with a Laptop? ( Link to this FAQ )
The problem is the P/C AC power source has only two wires are connected: +/- DC, no earth ground. In this case voltages of 75 to 85VAC between the P/C chassis (Signal Ground) and Earth can be found. Grounding the P/C chassis fixed the problem.
Remote Access Servers
General
How do I find the MAC address of my Patton RAS? ( Link to this FAQ )
Go into the GUI. Click on the "Interfaces" tab on the left hand side. Then you will want to click on the first "Details..." tab and there you will see the MAC address listed as the "Physical Address:".
Drivers for Windows 2003 Server for the Dialfire 2977? ( Link to this FAQ )
For Windows 2003 (server) drivers for the 2977 are included in the OS driver database.
2960, 2996 New 3.5.1 Code Modem Connection Troubles ( Link to this FAQ )
If you are having users that are having troubles connecting to your RAS server with code version 3.5.1 please be sure to set the TX level on your RAS box to 14.
You can do this by doing the following.
Click DIAL IN -> Modify Defaults -> Scroll to bottom of page and locate TX LEVEL. Input 14 here, and click submit query.
Has Patton checked their RAS products for the recent SNMP vulnerabilities issued by CERT? ( Link to this FAQ )
How do I use SNMP to kill a user's connection? ( Link to this FAQ )
The OID to kill a call is as follows:
1.3.6.1.4.1.1768.5.100.1.3.x
where x=call ID number on the dial-in page of the RAS
This OID needs to be set to 10.
This is valid for both the 2800, 29xx and 3120 series.
How do I change the date on the remote access server? ( Link to this FAQ )
The 29xx, 2800 and 3120 models do not support the concept of time/date like a calendar. The only time it knows is the time since the remote access server was last rebooted.
In the system log, the time stamping increments by 100 every second. So dividing the time stamp by 6000 will tell you how many minutes after the last reboot the event occurred.
What is the minimum system requirements for a 2977 PT1/PT2 PE1/PE2? ( Link to this FAQ )
A Pentium 333Mhz processor or higher.
256 MB RAM for Windows 2000
2MB RAM per 2977 port
What are the minimum system requirements for a 2977 B4U or B4ST? ( Link to this FAQ )
Pentium 333 Mhz processor or higher
256 MB RAM for Windows 2000
2MB RAM per 2977 port
What are the MFR Version 2 line settings for the 2960 in the US? ( Link to this FAQ )
The MFR2 line settings are specific to a bit oriented signal on an E1 line. E1 lines are used everywhere in the world but Canada, Japan, parts of Taiwan, and the USA. The line you will have installed will be a T1 (message oriented or robbed bit) line.
What types of High Speed Facilities do the US use? ( Link to this FAQ )
The US uses a T1 (message oriented or robbed bit) line. A message oriented line is a PRI (Primary RateInterface) ISDN line with 23B channels and 1 D channel. Instead of the phone company sending the ring voltage in-band with the data, the digital packet with the hook status is sent in its own channel (D channel) as an out-of-band signal (Common Channel Signaling). The gain from losing one channel (D channel which happens to be channel 24 on a T1) is that the 23 B channels are full 64K able DS0s and are not interrupted by calls coming in on the line. This is the line you want to use if you have users using a ISDN BRI (Basic Rate Interface) and a Terminal Adapter. This line also will tell you information like who is calling and what number was dialed. Robbed bit signaling is a type of in-band signaling (Channel Associated Signaling) used in T1 when the D channel is buried with the B channels, using the least-significant bits to indicate the hook condition. The least significant bits are "robbed" from each DS0 leaving a throughput of 56kb per second. Robbed bit signaling leaves you with 24 DS0s rather than 23 (remember you can only make ISDN BRI calls on a PRI that's the advantage, because the robbing of the bits only allows each DS0 56K and there is no digital channel to send digital packets on).
What are the HTTP/SNMP password protection options? ( Link to this FAQ )
There are two (2) passwords. The first is a monitor password which allows one to view, but not change, all configuration options and statistical data---all passwords are hidden. The second is a superuser password which allows full configuration and password control. These passwords are the same for the SNMP RO and RW community strings.
New software images can be found at http://upgrades.patton.com. The software is upgraded using FTP from any computer. The steps are as follows.
FTP into the 2960
The username to enter is KillImage
The password is the superuser password for that 2960
Set the transfer mode for FTP to BIN (for Binary). On most FTP clients this done by typing in BIN at the FTP> prompt.
Use the FTP put command to put the file into the 2960. On most FTP clients this is done by typing put (where name-of-image-file is the name of the new software load.)
The DSP's are best thought of as a resource pool. At ring time the 2960 is told by the calling switch whether the call is digital or analog. At that time a DSP is allocated and assigned to that call. The way the DSP's are kept in the resource pool and allocation is in a Round-Robin fashion.
On the web management pages there is an option to change monitor privileges. The default is readonly(2). If this is changed to none(0), the monitor user can not log in to the web pages.
What is the pin-out for the RJ45 to DB-9 converter? ( Link to this FAQ )
To connect to the RS232 port you need a straight through cable and a RJ48C to DB-9 converter. These were included with your remote access server but if you have lost them, the pin-out for the converter is as follows:
RJ45 DB9 1 - 6 Data Set Ready 2 - 1 Carrier Detect 3 - 4 Data Terminal Ready 4 - 5 Signal Ground 5 - 2 Received Data 6 - 3 Transmitted Data 7 - 8 Clear to Send 8 - 7 Request to Send
What kind of data connections does the 2960 support? ( Link to this FAQ )
The 2960 series supports the following types of connections:
ASCII/VT100
Async PPP
Sync PPP
Frame Relay
ISDN
The 2960 supports all of the types of dial-in connections simultaneously. If the incoming E1/T1 supports ISDN then some users can connect using an analog modems and other users can connect over the same incoming line using ISDN.
How do I configure SYSLOGD in my Linux box to create a debug log? ( Link to this FAQ )
On the Patton you need to do the following:
Go to System Log->Modify.
Set IP address for Syslog Daemon to the IP address of the machine running syslogd. Select Submit Query
Set minimum priority for syslog daemon to the desired level of logging. The lower the number next to the option the more information you get. Verbose is quite verbose. Debug is also probably more logging than you want. Select submit query.
If you want the messages to go into a separate file called local1-local7 then you set the Unix Facility.
The following instructions are valid for Rehat Linux V6.2. The commands may be slightly different for your version of Unix/Linux.
On the Linux machine, if syslogd is running it will automatically start logging information to the /var/log/messages. If you want that information to go to a separate file as indicated by selecting the Unix facility above then do the following:
Go to /etc and edit syslog.conf.
To log messages to local1.log add the following line:
local1.* /var/log/local1.log
For each of local1-local7 you would need to add a line like the one above.
Stop and start syslogd.
To kill the process type: $ killall syslogd
Restart syslogd by going to /sbin and typing: $ ./syslogd or just $ syslogd from any directory if you have /sbin in your path.
Create an empty file for syslogd to write messages to:
cd /var/log
touch local1.log
You would need to touch each localx file you will have syslogd write messages to.
Following is an example config for the number of active calls. Our SNMP MIB definition is contained in links on the SNMP page. The definitions include the OIDs necessary for MRTG or other programs that use OIDs to get or change MIB variables.
This goes and gets the active users on the 2800
# BE SURE TO CHANGE THE IP ADDRESS, WORKDIR, Directory
# and timezone for your system.
WorkDir: /usr/local/www/data
WriteExpires: Yes
Directory[ActiveCalls]: p26
Timezone[ActiveCalls]: GMT+5
Target[ActiveCalls]: 1.3.6.1.4.1.1768.5.25.0&1.3.6.1.4.1.1768.5.25.0:monitor@1.2.3.4
MaxBytes[ActiveCalls]: 32
AbsMax[ActiveCalls]: 32
YLegend[ActiveCalls]: Active Calls
Options[ActiveCalls]: gauge
Unscaled[ActiveCalls]: dwmy
Title[ActiveCalls]: diActive Users
PageTop[ActiveCalls]:
PageTop[ActiveCalls]: diActive Users on Patton
Happy MRTGing. You will soon have real-time graphs that look like this:
Please click to access a scan of the 2960 and 2800 mib trees. If you look at the OID for the dial-in page 1.3.6.1.4.1.1768.5.25.0, you can follow it down the page.
1.3.6.1.4.1.1768 = This is the branch to the Patton Enterprise mib.
5 = This is the calldialin area
25 = diActive (active users on the 2800)
0 = this instance
Here is what the snipet of the SNMP MIB for this (it is under the common.mib)
diActive OBJECT-TYPE
SYNTAX INTEGER
ACCESS read-write
STATUS mandatory
DESCRIPTION "The total number of active calls."
::= { calldialin 25 }
The word "calldialin 25" tells you it is under the calldialin branch, in this case it is integer 5, and it is variable 25.
To get all of the items that you can manage in the 2800, goto the HTTP managment system and click on SNMP. All of the MIBS are there at the top.
Click and save all three. Then you can find everything under the sun!
Here are some additional OIDs that you might find useful:
1.3.6.1.2.1.1.0 - System Description w/ Software Version
1.3.6.1.2.1.1.3 - Uptime Ticks Since Box Rebooted
1.3.6.1.4.1.1768.5.17 - Total Number of calls
1.3.6.1.4.1.1768.5.25 - Number of calls
1.3.6.1.4.1.1768.5.39 - Max number of calls
1.3.6.1.4.1.1768.16.2 - Number of DSPs Avail
1.3.6.1.4.1.1768.12.2.1.11 - framrelTXOctets
1.3.6.1.4.1.1768.12.2.1.12 - framrelRXOctets
Do you have an MRTG script for the the box temperature on the 2960 or 2996? ( Link to this FAQ )
Box Temperature Example:
## Temperature for 2996 Site 1.2.3.4
WriteExpires: Yes
Timezone[29xxTemp]: GMT+5
Target[29xxTemp]:1.3.6.1.4.1.1768.20.1.31.0&1.3.6.1.4.1.1768.20.1.31.0:monitor@1.2.3.4
Title[29xxTemp]: 1.2.3.4 Box Temperature
PageTop[29xxTemp]: 2996 Box Temperature
MaxBytes[29xxTemp]:90
AbsMax[29xxTemp]:90
Options[29xxTemp]: growright, nopercent, integer,absolute, gauge
Ylegend[29xxTemp]: Temp C
ShortLegend[29xxTemp]: Degrees -C
Unscaled[29xxTemp]: dwmy
Legend1[29xxTemp]:Current Temp
Legend2[29xxTemp]:Current Temp
Legend3[29xxTemp]: 5 Min Average Temp
Legend4[29xxTemp]: 5 Min Average Temp
How is the MTU(maximum tranmission unit) determined on a call? ( Link to this FAQ )
The remote access server has a default MTU of 1524. This is the maximum The MTU of the ethernet media. We recommend that this not be changed. The MTU will be negotiated during LCP negotation for a dial-in user. During LCP negotiation we will tell the remote end we are capable of 1524.
There are two ways in which a customer can receive an MTU that is lower:
1. The RADIUS software returns a Framed-MTU attribute that specifies a lower value. In releases 2.3.3 and lower, we will change the MTU in response to this attribute. In 2.4.1 and above this RADIUS attribute is ignored.
2. The remote modem indicates that an MTU of 1524 is not acceptable and wants 512. We 'give in' to that request and assign 512 as the MTU.
A lower MTU on a dial-in call is not necessarily a bad thing. Most packets that are larger than the MTU of a connection can be broken down and sent down the connection in smaller chunks.
The problem occurs when the remote access server receives a packet larger than the connection's MTU and the packet has the Do Not Fragment bit set. This bit tells the remote access server that it must not split the packet into smaller chunks. Because the remote access server can not split the packet into smaller chunks AND can not send it as-is over the connection, it drops the packet.
What the dial-in user may see is the inability to load certain web sites.
Console into the unit via the RS-232 port
Reboot the RAS
When prompted type bbb
>From the prompt Model29xx/31xx> 'type' factory
>From the prompt Model29xx/31xx> 'type' reset
This procedure will set the unit back to factory defaults
We no longer recover lost passwords.
SIPx Nano
General
How do I configure my Smartnode (FXS) gateway to register with the SipX Nano? ( Link to this FAQ )
system
ic voice 0
low-bitrate-codec g729
profile acl LOCKOUT
profile napt NAPT
profile ppp default
profile call-progress-tone US_Dialtone
play 1 1000 350 -13 440 -13
gateway sip GW_SIP
bind interface eth0 router
no shutdown
port ethernet 0 0
medium auto
encapsulation ip
bind interface eth0 router
no shutdown
port ethernet 0 1
medium auto
encapsulation ip
bind interface eth1 router
no shutdown
port fxo 0 0
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO_00 switch
no shutdown
port fxo 0 1
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO_01 switch
no shutdown
port fxo 0 2
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO_02 switch
no shutdown
port fxo 0 3
use profile fxo us
caller-id format bell
encapsulation cc-fxo
bind interface IF_FXO_03 switch
no shutdown
How do I see how many user licenses I have in SipX? ( Link to this FAQ )
Enter the following command:
rpm -ql -all -i |grep Name |grep patton
It should look like this:
Name : ecspatton Relocations: /usr
Name : ecspatton-15 Relocations: /usr
How do I change the root password in the Linux shell of the Nano? ( Link to this FAQ )
Enter the following command and follow the steps listed:
"passwd"
It should look like this:
[root@sipx ~]# passwd
Changing password for user root.
New UNIX password:
Retype new UNIX password:
passwd: all authentication tokens updated successfully.
How do you restart the services in the Nano? ( Link to this FAQ )
You can enter the following command:
service sipxpbx restart
After you have loaded the software, you will need to install bind. You first want to do a "yum install bind". Make sure that you are on a network that has connectivity to the web. After that is done, you will need to install the following files in order for Bind to function. The following files need to be edited according to your network. Please see the notes inside each file.
Note: You will want to copy the below example files into a text document to modify them.
Note: Hitting "ctrl+h" will open a replace box. You can enter the above statements that need to be changed into the "find what:" field and eneter the replacement statements into the "Replace with:" field. This will do all of them at once. You can do this for each statement.
zone "patton.local" IN {
type master;
file "patton.local.zone";
allow-update { none; };
};
zone "200.168.192.IN-ADDR.ARPA" {
type master;
file "192.168.200.200.zone";
};
Change with below requirements then cut and paste the above text in the "nano /etc/named.conf" file.
### Replace the following ###
- all "patton.local" with domian name
- "200.168.192." with inverted IP address
- "192.168.200.200" with IP address
Note: Hitting "ctrl+h" will open a replace box. You can enter the above statements that need to be changed into the "find what:" field and eneter the replacement statements into the "Replace with:" field. This will do all of them at once. You can do this for each statement.
nano /var/named/example.zone:
$TTL 1D
@ IN SOA ns1.patton.local. root.patton.local. (
200602132 ; serial#
3600 ; refresh, seconds
3600 ; retry, seconds
3600 ; expire, seconds
3600 ) ; minimum TTL, seconds
NS ns1.patton.local. ; Inet Address of nameserver
patton.local. MX 10 mail ; Primary Mail Exchanger
localhost A 127.0.0.1
ns1 CNAME sipx
mail CNAME sipx
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
; sipX Servers for SIP domain 'patton.local'
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
sipx.patton.local. IN A 192.168.200.200
patton.local. IN NAPTR 2 0 "s" "SIP+D2T" "" _sip._tcp.patton.local.
patton.local. IN NAPTR 2 0 "s" "SIP+D2U" "" _sip._udp.patton.local.
_sip._tcp.patton.local. IN SRV 1 0 5060 sipx.patton.local.
_sip._udp.patton.local. IN SRV 1 0 5060 sipx.patton.local.
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
Change with below requirements then cut and paste the above text in the "nano /var/named/(hostname).zone" file.
### Replace the following ###
- "patton.local" with Sip domain name
- "sipx.patton.local." with host name
- "192.168.200.200" with Nano IP address
Note: Hitting "ctrl+h" will open a replace box. You can enter the above statements that need to be changed into the "find what:" field and eneter the replacement statements into the "Replace with:" field. This will do all of them at once. You can do this for each statement.
nano /var/named/ip.zone:
$TTL 1D
@ IN SOA ns1.patton.local. root.patton.local. (
200602132 ; serial#
3600 ; refresh, seconds
3600 ; retry, seconds
3600 ; expire, seconds
3600 ) ; minimum TTL, seconds
NS ns1.patton.local. ; Inet Address of nameserver
1 IN PTR localhost.
200 IN PTR sipx.patton.local. ; Record of class IN by default
Change with below requirements then cut and paste the above text in the "nano /var/named/(IPaddress of Nano).zone" file.
### Replace the following ###
- "patton.local" with Sip domain name
- "sipx.patton.local." with host name
- "200" with last number in the IP address of the Nano
Note: Hitting "ctrl+h" will open a replace box. You can enter the above statements that need to be changed into the "find what:" field and eneter the replacement statements into the "Replace with:" field. This will do all of them at once. You can do this for each statement.
Also, make sure the domain in the SIPx GUI is set properly to your FQDN. You can check this by going into the GUI under System/Domain.
How do I check to see if all my services are running in the linux shell? ( Link to this FAQ )
You can enter the following command:
service sipxpbx configtest
How do I check my SIPx version in the Linux shell? ( Link to this FAQ )
You can enter the following command:
sipx-config --version
How do I change the superadmin password in the SIPx GUI? ( Link to this FAQ )
1. Gain Access to a terminal on the sipnano
2. Type: sipxconfig.sh --database reset-superadmin at the prompt.
3. Access the nano via web browser
4. Login with superadmin and leave the password blank.
5. To change the password, go into "Users", click on superadmin and change the pin to the password of your choice.
What are the factory default settings for the SIPxNano? ( Link to this FAQ )
Default Nano Settings
Shell:
Hostname: sipx.patton.com
IP Address: 192.168.200.200
Netmask: 255.255.255.0
Gateway: 192.168.200.1
Nameserver: 0.0.0.0
Sipx Login: root
Admin Password: superuser
Admin Email: superuser@patton.local
Time Zone Continent: US
Time Zone: Eastern
SIP Domain name: patton.local
Organization Name: PBX
Section Name: VoIP Services
Country Code: US
State or Province: Maryland
Locality (city): Gaithersburg
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.
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
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 ''
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 -
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.
Under Telephony, SIP - Uncheck Send Registration Request with Expire Time:
Under Telephony, phone one (or two), user information.
Set the phone number to the number to be used for phone one. (For example: 555 for line one).
Set the Caller id to be used for phone one.
Under Telephony, phone one (or two), supplementary services set "IP dial srv " to YES.
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.
Save all the changes.
Repeat steps 3-5 for phone 1 using a different phone number (For example: 444 for line two).
Reload the SL4020.
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 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.
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.
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.
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.)
FEATURE
KEYPAD
FEATURE
KEYPAD
Call Hold
F(flash) 1
Call Alternative
F*
Conference
F7
Conference Drop
F8
Call Transfer
F4
.
.
Do Not Disturb ON
*82
Do Not Disturb OFF
#82
Distinctive ON
*90
Distinctive OFF
#90
Call Waiting ON
*91
Call Waiting OFF
#91
Incoming Caller ID Display ON
*92
Incoming Caller ID Display OFF
#92
Self Caller ID Block Service ON
*93
Self Caller ID Block Service OFF
#93
Anonymous Call Rejection ON
*94
Anonymous Call Rejection OFF
#94
Incoming Call Block ON
*95
Incoming Call Block OFF
#95
Call Forward Selective ON
*96
Call Forward Selective OFF
#96
Call Forward All ON
*97
Call Forward All OFF
#97
Call Forward Busy ON
*98
Call Forward Busy OFF
#98
Warm Line ON
*99
Warm Line OFF
#99
IP Dialing ON
*80
IP Dialing OFF
#80
Speed Dialing ON
*81
Speed 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 keyyyyyy = 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
*73
Wait for 3 short confirmation tones before hanging up.
*** 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.
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 -
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
How to upgrade firmware on the SL4050 phones ( Link to this FAQ )
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.
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
In Windows, Click on "Start", then "Run". In the "Open" input box, enter "cmd" and press OK. A DOS window will open.
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.
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.
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.
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.
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.
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:
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.
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:
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.
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.
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
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.
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
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.
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
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
Determine Rule
ask your government or find on a website the Rule according which in your Timezone Daylight Saving is handled
Make Timer Command
timer
Variables in the Timer Command
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"
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
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 !
Connect to the SmartNode using the console cable.
Cycle the power
At the prompt "Press ^C to abort boot script, press enter to start immediately" press ctrl and c.
Type in the command 'fis load'
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 laterIf 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.
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'.
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)
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.
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).
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
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:
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
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:
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
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 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.
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
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".
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 )
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
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
xDSL Equipment
General
If I can't get my 1095 mDSL modems to work, what should I try first? ( Link to this FAQ )
When you encounter a problem with the 1095s, try the following steps:
First to confirm that the units are operating properly, you should connect two units using a Point to Point configuration. You should connect the units using a two-wire twisted pair cable, pins 4 and 5.
Next turn both units off, set switches S3-7 OFF. This will reset the units Software to factory defaults.
Power up both units for about 2 seconds
Power off both units and return S3-7 to the ON position
The units are now ready to accept the switch setting that you set.
Next both units must be configured for the same data rate. Factory default is 768Kbps, Switch S3 - 1 ON, S3 - 2, 3, and 4 OFF S3 - 5 and 6 ON.
Now one unit must be configured for Central Office ( CO) INTERNAL clock ( S2 - 6 and S2 - 7 ON ), Next set the other unit for Customer Premises (CP) RECEIVE RECOVER clock ( S2 - 6 ON and S2 - 7 OFF ).
Check the front panel Test Mode Switches, they should be in the NORMAL position.
Turn on the power to the model 1095’s, after 30 Seconds the NS ( No Signal ) LED’s should turn OFF, CTS (Clear To Send ), CD ( Carrier Detect ) and DTR ( Data Terminal Ready ) LED will turn Green.
After the (NS) LED turn off, and CTS, CD and DTR have turned Green, select a unit to operate as the LOCAL modem, the other modem will be the REMOTE. Move a TEST MODE switch on the LOCAL modem to REMOTE, the TM (Test Mode) LED will light on both units.
Next select the 511E test mode on the LOCAL modem, the ER (ERror) led on the LOCAL modem will now FLASH at about a one second rate. After 45 seconds the Pattern Generator will turn off, the ERror LED will flash at a high rate until the 511 Test switch is turned off.
Return the Test Mode Switches to the NORMAL po
If I can't get my 1092 iDSL modems to work, what should I try first? ( Link to this FAQ )
When you encounter a problem with the 1092s, try the following steps:
The testing of the Model 1092 and 1092A are the same except for the 2w/4w selection for the model 1092A.
First do a Hardware Reset, turn the power off, set configuration switch (S1-6 and 7 "OFF")
Turn on the power for 5 or 10 seconds. Turn the power OFF and reconfigure the model 1092A using the following configuration. One unit must be set as Master the other unit as Slave. MASTER UNIT SETTINGS
Switch S2 settings are the same as the MASTER UNIT.
Connect the twisted pair circuit between the model 1092A's, turn on the power, after 10 to 15 seconds the NS (no signal)- LED will go off and the CD (carrier detect) LED will turn GREEN.
Do a REMOTE DIGITAL LOOP-BACK test using the 511/511E test switch (Page 30 of the user manual).
How do I upgrade the software of a 1095 mDSL modem? ( Link to this FAQ )
You will need a cable to connect the Model 1095 control port to a DB-9 PC serial port and a flash image (xxxxxx.IMG). The cable should be wired:
Then follow these instructions to load the code into the 1095 Flash Memory.
Copy the (xxxxxx.IMG) to a TEMP file on the PC that will be used to program the Model 1095.
Use Windows HyperTerm, or some other Terminal Emulation Program set for 9600 bps, NO parity, 1 stop, X-ON/ X-OFF.
Turn the Model 1095 power OFF 4. Set configuration Switch S2- 6 and 7 OFF (Reserved)
Turn ON the Model 1095 Power, The TM and ER LED will turned on. The 1095 will send "BEGIN" to the PC.
From Hyper Term select Send Text File on the Transfer menu.
Open the image File, xxxxxx.IMG. You will see DOT's on the PC screen (about 4 lines of DOT's.) Next the 1095 will send COMPLETE ERRORS : 00
What is Octet Mode in the IM1/F for the 1092? ( Link to this FAQ )
Octet Mode is a special protocol is used between the model 1092s. The connection to the G.703 line is still 64Kbps and the connection between the Model 1092s must be set for 128Kbps. Octet mode use bipolar violations to define the start of an 8 bit byte, The IM1/F recognizes the violation and converts the signal into two Bytes (16 bits), one framing byte and one data byte to send to the receiving unit (frame+Data) (frame+Data), the receiving IM1/F strips the frame byte and pass the Data and the bipolar violation to the G.703 interface at 64Kbps. For Byte timing to function properly, requires each 1092 to have IM1/F modules.
2701/B, C, D and I Lock up or Loose Link Periodically - Test Mode Light Comes On. ( Link to this FAQ )
If you are experiencing the TM LED turning on unexpectedly, change configuration switch S2-6 from OFF to ON.
This will DISABLE response to V.54 command.
V.54 should only be used when the Model 2701/X’s are connected Back to Back in a campus configuration.
General 
