Frequently Asked Questions
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 G.SHDSL Products |
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| General |
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| What are the DiamondLink G.SHDSL Routers? ( Link to this FAQ ) |
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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.
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| What is G.SHDSL? ( Link to this FAQ ) |
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| 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.
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| Must the DiamondLink G.HDSL Routers or 3086 IADs be used in pairs? ( Link to this FAQ ) |
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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.
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| Do the G.SHDSL products support Network Address Translations(NAT)? ( Link to this FAQ ) |
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| 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.
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| Do the G.SHDSL products support bridging? ( Link to this FAQ ) |
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| 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.
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| What are the ethernet capabilities of the 3201,3241 and 3086? ( Link to this FAQ ) |
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| 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.
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| What are some typical applications for the Model 3201 and Model 3241 DialmondLink Routers? ( Link to this FAQ ) |
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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.
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| What HDSL standards do the DiamondLink G.HSDSL routers And IpRocketlink IAD use? ( Link to this FAQ ) |
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| 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.
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| Is surge protection provided with the DiamondLink routers? ( Link to this FAQ ) |
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| 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.
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| How are the G.SHDSL products configured? ( Link to this FAQ ) |
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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. |
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| What protocols are supported by the G.SHDSL products? ( Link to this FAQ ) |
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- 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.
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| What is the 3086 IpRocketLink G.SHDSL IAD? ( Link to this FAQ ) |
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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.
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| What is the primary application for the Model 3086 ipRocketLink? ( Link to this FAQ ) |
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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
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| What are the interfaces available on the model 3086s? ( Link to this FAQ ) |
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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
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| Microproducts and General Communications Equipment |
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| General |
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| How to check if Microsoft's USB fix is installed on my computer? ( Link to this FAQ ) |
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| 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.
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| What services do I need from my phone company to use modem on hold? ( Link to this FAQ ) |
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| 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. |
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| What do I do if I have a USB modem on WinXP and the modem frequently disconnects? ( Link to this FAQ ) |
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| Two Wire Operation of Models 1004A, 1009, 2085 and 2089? ( Link to this FAQ ) |
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| 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-
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| What is the meaning of DCE (data communication equipment) and DTE (data termination equipment)? ( Link to this FAQ ) |
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| DCE Data Communications Equipment, is typically a Modem or other device that connects to a Network. DTE Data Terminal Equipment, is typically a Computer.
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| What is the different between wired as DCE and DTE? ( Link to this FAQ ) |
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| 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.
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| How should two DCE devices be connected? ( Link to this FAQ ) |
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| 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.
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| What pin-out should be used to connect DCE to DTE? ( Link to this FAQ ) |
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| The cable should be connected straight through. |
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| How can the distance limitations on any unit be extended? ( Link to this FAQ ) |
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| 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.
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| Why won't my RS485 or RS422 device work properly, and which signals do Patton use as + and -? ( Link to this FAQ ) |
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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.
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| What's required for a Short Range Multipoint Modem to operate Two Wire? ( Link to this FAQ ) |
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| 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.
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| Why doesn't my Short Range Modem work with a Laptop? ( Link to this FAQ ) |
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| 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.
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| Remote Access Servers |
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| General |
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| How do I find the MAC address of my Patton RAS? ( Link to this FAQ ) |
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| 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:". |
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| Drivers for Windows 2003 Server for the Dialfire 2977? ( Link to this FAQ ) |
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| For Windows 2003 (server) drivers for the 2977 are included in the OS driver database. |
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| 2960, 2996 New 3.5.1 Code Modem Connection Troubles ( Link to this FAQ ) |
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| 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. |
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| Has Patton checked their RAS products for the recent SNMP vulnerabilities issued by CERT? ( Link to this FAQ ) |
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| How do I use SNMP to kill a user's connection? ( Link to this FAQ ) |
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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.
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| How do I change the date on the remote access server? ( Link to this FAQ ) |
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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.
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| What is the minimum system requirements for a 2977 PT1/PT2 PE1/PE2? ( Link to this FAQ ) |
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| A Pentium 333Mhz processor or higher.
256 MB RAM for Windows 2000
2MB RAM per 2977 port
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| What are the minimum system requirements for a 2977 B4U or B4ST? ( Link to this FAQ ) |
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| Pentium 333 Mhz processor or higher
256 MB RAM for Windows 2000
2MB RAM per 2977 port
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| What are the MFR Version 2 line settings for the 2960 in the US? ( Link to this FAQ ) |
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| 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.
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| What types of High Speed Facilities do the US use? ( Link to this FAQ ) |
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| 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).
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| What are the HTTP/SNMP password protection options? ( Link to this FAQ ) |
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| 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. |
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| How can I upgrade my software? ( Link to this FAQ ) |
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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.)
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| How are DSP's allocated? ( Link to this FAQ ) |
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| 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.
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| Why can I not log in as monitor? ( Link to this FAQ ) |
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| 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.
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| What is the pin-out for the RJ45 to DB-9 converter? ( Link to this FAQ ) |
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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
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| What kind of data connections does the 2960 support? ( Link to this FAQ ) |
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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.
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| How do I configure SYSLOGD in my Linux box to create a debug log? ( Link to this FAQ ) |
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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.
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| Do you have any sample scripts for MRTG? ( Link to this FAQ ) |
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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:
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| How do I figure out what the SNMP OID is? ( Link to this FAQ ) |
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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
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| Do you have an MRTG script for the the box temperature on the 2960 or 2996? ( Link to this FAQ ) |
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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
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|
| 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.
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|
| How do I recover Factory password? ( Link to this FAQ ) |
|
| 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.
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|
|
| 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
profile call-progress-tone US_Alertingtone
play 1 1000 440 -19 480 -19
pause 2 3000
profile call-progress-tone US_Busytone
play 1 500 480 -24 620 -24
pause 2 500
profile call-progress-tone US_Releasetone
play 1 250 480 -24 620 -24
pause 2 250
profile tone-set default
profile tone-set US
map call-progress-tone dial-tone US_Dialtone
map call-progress-tone ringback-tone US_Alertingtone
map call-progress-tone busy-tone US_Busytone
map call-progress-tone release-tone US_Releasetone
map call-progress-tone congestion-tone US_Busytone
profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
profile pstn default
profile sip default
profile dhcp-server DHCP
network 192.168.1.0 255.255.255.0
include 1 192.168.1.10 192.168.1.99
lease 2 hours
default-router 1 192.168.1.1
domain-name-server 1 192.168.1.1
profile aaa default
method 1 local
method 2 none
context ip router
interface eth0
ipaddress dhcp
use profile napt NAPT
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
interface eth1
ipaddress 192.168.1.1 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
dhcp-server use profile DHCP
context cs switch
routing-table called-e164 RTEST
route .%T dest-interface IFSIPX
routing-table called-e164 FXS_OUT
route default dest-interface LINE1
interface sip IFSIPX
bind gateway GWSIP
service SIPX
route call dest-table FXS_OUT
remote sipx.patton.com
use profile tone-set US
interface fxs LINE1
route call dest-table RTEST
use profile tone-set US
context cs switch
no shutdown
gateway sip GWSIP
bind interface eth0 router
service SIPX
domain sipx.patton.com
realm 500
registration manual sipx.patton.com
user 500 authenticate password 6YIIjHz41cI= encrypted register
gateway sip GWSIP
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 fxs 0 0
caller-id format bell
use profile fxs us
encapsulation cc-fxs
bind interface LINE1 switch
no shutdown
port fxs 0 1
shutdown
port fxs 0 2
shutdown
port fxs 0 3
shutdown
port fxs 0 4
shutdown
port fxs 0 5
shutdown
port fxs 0 6
shutdown
port fxs 0 7
shutdown
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|
| How can I configure my Smartnode (FXO) as a PSTN gateway with the SipX Nano? ( Link to this FAQ ) |
|
system
ic voice 0
low-bitrate-codec g729
profile napt NAPT
profile ppp default
profile call-progress-tone US_Dialtone
play 1 1000 350 -13 440 -13
profile call-progress-tone US_Alertingtone
play 1 1000 440 -19 480 -19
pause 2 3000
profile call-progress-tone US_Busytone
play 1 500 480 -24 620 -24
pause 2 500
profile call-progress-tone US_Releasetone
play 1 250 480 -24 620 -24
pause 2 250
profile tone-set default
profile tone-set US
map call-progress-tone dial-tone US_Dialtone
map call-progress-tone ringback-tone US_Alertingtone
map call-progress-tone busy-tone US_Busytone
map call-progress-tone release-tone US_Releasetone
map call-progress-tone congestion-tone US_Busytone
profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20
profile pstn default
profile sip default
profile dhcp-server DHCP
network 192.168.1.0 255.255.255.0
include 1 192.168.1.10 192.168.1.99
lease 2 hours
default-router 1 192.168.1.1
domain-name-server 1 192.168.1.1
profile aaa default
method 1 local
method 2 none
context ip router
interface eth0
ipaddress 192.168.1.8 255.255.255.0
use profile napt NAPT
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
interface eth1
ipaddress unnumbered
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
route 0.0.0.0 0.0.0.0 192.168.1.8 0
context cs switch
routing-table called-e164 RT_FROM_PSTN
route default dest-interface IF_SIP TO_100
mapping-table called-e164 to called-e164 TO_100
map default to 100 (***This will map all incoming calls to the auto attendant***)
interface sip IF_SIP
bind gateway GW_SIP
service default
route call dest-service HG_FXO
remote 192.168.1.7 (***this is the IP of the SipX Nano***)
interface fxo IF_FXO_00
route call dest-table RT_FROM_PSTN
use profile tone-set US
interface fxo IF_FXO_01
route call dest-table RT_FROM_PSTN
use profile tone-set US
interface fxo IF_FXO_02
route call dest-table RT_FROM_PSTN
use profile tone-set US
interface fxo IF_FXO_03
route call dest-table RT_FROM_PSTN
use profile tone-set US
service hunt-group HG_FXO
timeout 5
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
drop-cause user-busy
route call 1 dest-interface IF_FXO_00
route call 2 dest-interface IF_FXO_01
route call 3 dest-interface IF_FXO_02
route call 4 dest-interface IF_FXO_03
context cs switch
no shutdown
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
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| 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
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| 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.
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| How do you restart the services in the Nano? ( Link to this FAQ ) |
|
You can enter the following command:
service sipxpbx restart
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|
| How do I create BIND files for the SIPx? ( Link to this FAQ ) |
|
When installing Bind on Nano box:
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.
- /etc/named.conf
- /var/named/example.zone
- /var/named/ip.zone
- /etc/resolv.conf
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.
Examples:
nano /etc/named.conf:
options {
directory "/var/named";
dump-file "/var/named/data/cache_dump.db";
statistics-file "/var/named/data/named_stats.txt";
};
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.
nano /etc/resolv.conf:
search domain.com
nameserver
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.
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|
| How do I check to see if all my services are running in the linux shell? ( Link to this FAQ ) |
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| You can enter the following command:
service sipxpbx configtest |
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| How do I check my SIPx version in the Linux shell? ( Link to this FAQ ) |
|
| You can enter the following command:
sipx-config --version
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|
| 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.
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|
| 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
GUI:
User ID: superadmin
PIN: patton
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| How do I change the date in the SIPxNano? ( Link to this FAQ ) |
|
First you must stop the sipxpbx services.
“service sipxpbx stop”
Then enter the following command:
“date mmddttttyyyy”
Example: date 032611562007
mm- 2 digit month ex.(03)
dd – 2 digit day ex.(26)
tttt – time ex.(11:56)
yyyy - year ex.(2007)
After you are done, you must restart the sipxpbx services.
“service sipxpbx start”
To check the date, just enter "date".
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|
| What is a voice attendent? ( Link to this FAQ ) |
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| The automatic Internal Voice Automation inside of the SIPx IP PBX. |
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|
|
| SmartLink VoIP |
|
| Debug and Logging |
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| Using Syslog to troubleshoot SmartLinks ( Link to this FAQ ) |
|
The SmartLink supports syslog for simple troubleshooting.
To enable syslog logging, look under the GUI "System" menu, then select "Configuration". Set "Enable Syslog" to "yes" and put the IP address of your Syslog server in the "Syslog Server" field. For additional troubleshooting, you can also set "Enable Debug" to "yes", set "Debug Server" to the IP address of your Syslog server, and set the "Debug Connection Port" to 514. Click "Save" and then "System/Reload", "Reset".
If you are not familiar with Syslog, an easy to use freeware Syslog package is provided by Kiwi. See http://www.kiwisyslog.com. If Syslog doesn't give you enough information, we suggest you use hub or mirror a port on your Ethernet switch and use Ethereal. Ethereal will show you everything! |
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| 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.
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| 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”
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|
| What do I need to specify as the callerid for call forwarding, distinctive ring, caller id blocking and do not disturb? ( Link to this FAQ ) |
|
The SL4020 supplementary calling features of selective call forwarding, distinctive ring, caller id blocking, and do not disturb require specification of the calling party. There are several places in the SIP invite header that could carry that caller information. The "FROM" header of a SIP Message is as follows:
���From: CallerId;tag=G1024jh3-NXr3s......
The CallerID field of this header is used in the "Caller ID" fields of supplementary service features. Some implementations identify the callerID as the "Display Name"
For example: The "FROM" header of a SIP Message may look like:
���From: 123;tag=G1024jh3-NXr3s......
To use call forwarding, distinctive ring, caller id blocking and do not disturb specify "123" as the caller id. |
|
| I want to be able to lift the handset and have the phone call immediately connected to the call destination. I do not have a SIP server involved. ( Link to this FAQ ) |
|
There two things here, dialing by IP address and hotline dialing. The trick is putting the two together.
You can do direct IP dialing using the keypad on the phone. There's another FAQ posted on how to do IP address dialing.
In this case we need to change tell the SL4020 not to look for a SIP server and to set up a hotline call. Here is how you set up hotline dialing by IP address -
- 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.
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|
| 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 ''
|
|
| 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.
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|
| How do I directly dial an IP address? ( Link to this FAQ ) |
|
The SL4020 must be configured to enable dialing by IP address. This can be done on user configuration menu under Telephony, Line.
Once enabled, IP SIP addresses can be dialed directly by IP address using the telephone keypad.
For example: If you want to call 123456 at 192.168.1.3 port 5060.
You have to dial: *72123456*192*168*1*3*5060
*72 is the code to indicate that an IP call follows.
Optionally, you can skip "*5060" at the end, because it is the default value.
|
|
| How do I use Speed Dial? ( Link to this FAQ ) |
|
The SL4020 supports association of numbers to call with telephone handset keys. The configuration of speed dialing can be done using the telephone keypad or via the web. Up to eight speed dial numbers can be set.
Setting up speed dial using your telephone handset:
1. Press *81 on your telephone keypad to enable the speed dialing function. This only has to be done once.
2. Set up to eight speed dial numbers from your telephone keypad by entering *71xyyyyyyy where:
*71 = Is the code to set speed dial keys
x = The telephone keypad number to be set
yyyyyyy = The number to be called
For example: You want to set the telephone key number "2" to speed dial the destination 123456.
1. Dial *712123456 on the telephone and then hang up. Speed dial is now setup.
2. To use speed dial just press 2 and wait for the call to complete.
Speed dial can be disabled by entering #81 from your telephone keypad.
Note: SIP URLs cannot be included in speed dial settings. |
|
| Smart Link 4020 Calling Features ( Link to this FAQ ) |
|
|
Patton SL4020 Calling Features
The SL4020 family supports advanced calling features that can be turned on and off from phones attached to the SL4020. (Your telephony service provider must enable your service for these calling features to work.)
| 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. | . |
| Access Voicemail | *86 | . | . |
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| How to perform factory hardware reset ( Link to this FAQ ) |
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| *** 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 less than 4 seconds (just count to two). If you press the button for greater than 4 seconds the system will drop back into to bootloader mode, but your configuration parameters will be retained.
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.
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| General |
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| Default Password ( Link to this FAQ ) |
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| The default password for the 4021, 4022, and M-ATA units on the web interface is "root"
Please change this password for security reasons. |
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| How can I get my Smartlink to register with my Vbuzzer Account? ( Link to this FAQ ) |
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| To get your Smarlink to register with your Vbuzzer account all you need to do is set it up according to the following: SIP PAGE:
*Sip Registration Server Address: vbuzzer.com
Sip Port: 80
Sip Domain: vbuzzer.com
Voice Port: 5004
Send Registration Request with Expire Time: 3600
PHONE PAGE:
Phone Number: vbuzzer username
CallerID Name: vbuzzer username
User Name: vbuzzer username
Password: vbuzzer password
Port: 80 |
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| Telephones |
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| How do you transfer calls with a Patton SL4050/2 (SmartLink2 LINE SIP PHONE)? ( Link to this FAQ ) |
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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.
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| 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 ) |
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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** |
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| How do I factory reset a SL4050 telephone ? ( Link to this FAQ ) |
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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. |
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| How do I dial an IP address on the SL4050 SIP telephones? ( Link to this FAQ ) |
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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 |
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| 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 ) |
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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
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| How to upgrade firmware on the SL4050 phones ( Link to this FAQ ) |
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- 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.
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| SmartNode VoIP/ToIP |
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| Call Routing |
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