SmartNode™ 4110 Series Analog VoIP Gateway with up to 8 FXS/FXO ports The SmartNode 4110 Analog VoIP Media Gateway supports up to eight FXS or FXO telephone connections. Connect PSTN Lines, PBXs, and standard phones for voice and fax over any IP network. Now the corporate, small, or remote office can access Internet telephony services, eliminate toll charges and route calls to and from the PSTN, Internet, or LAN.
Need help? Call Patton at +1 301 975 1000 or email sales@patton.com.
Features & Benefits
Up to 8 FXS and/or FXO ports—Compact, reliable stand-alone VoIP gateway with different port options. Supports simultaneous voice or fax calls on all ports.
Advanced Local Call Switching—Virtual interfaces and routing tables provide industry leading flexibility in call handling programming. Local call switching, soft fallback to alternative routes. Simultaneously connects to multiple SIP services/IP PBXs.
Complete SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem handling, DTMF relay. Codecs G.729, G.723, etc.
Easy Management & Provisioning—Web-based management, SNMP, command line interface. Automated mass provisioning for efficient large-scale deployments.
Outstanding Interoperability—Proven integration for voice and T.38 fax with 3CX®, Asterisk™, PingTel™ and other leading IP PBX systems and soft switch vendors.
Overview
The business-class SmartNode 4110 VoIP Media Gateway supports up to eight transparent phone calls and leverages VoIP for carrier and corporate access. Connecting to any analog phone, fax, or PBX, the SN4110 is an effective and flexible solution for toll-bypass, remote/branch office voice connectivity, and enhanced carrier services.
The SN4110 series is the perfect choice for phone-to-IP connectivity supporting up to 8 FXS ports or a combination of 4 FXS and 2 or 4 FXO ports. With its FXS analog ports the SN4110 connects to any legacy telephone or PBX and provides dial-tone, ringing, caller-ID and other services. When equipped with FXO ports, the local PSTN can be accessed enabling local calling and enhanced toll-bypass applications while using a single connected telephone. Flexible call integration allows per-port telephone numbers, programmable call progress tones, and distinctive ringing. With Telephony-over-IP (ToIP) call switching, calls can automatically be routed to the PSTN or the IP network while providing flexible numbering plans and end-to-end feature transparency. PPPoE, DHCP, and VLAN offers universal IP connectivity and optional IPSEC VPN with AES/3DES guarantees secure voice over the public network.
Patton's SmartNode 4110 delivers the legacy phone interfaces, service transparency, and flexible PSTN integration for true converged packet voice.
Applications
Application—Remote Office/Branch Office Voice Extension and Access
In enterprise networks, transparent access to PBX features while using existing equipment is key to low-cost operations. Now, instead of installing a separate PBX at the remote office, the SmartNode 4110 is able to provide transparent extension while simultaneously connecting multiple locations. The extensions can be managed centrally and benefit from PBX services such as calling groups, least cost routing, and call forwarding. PSTN access allows local calls to be processed without using corporate remote PBX resources. Additionally, the corporate PBX can break-out and bypass any long distance charges by using the remote office for the local gateway.
Specifications
Voice Connectivity
• 2, 4, 6, or 8 FXS ports
2-wire Loopstart, RJ-11/12
Short haul loop 1.1 KM @3REN
EuroPOTS (ETSI EG201 188)
Programmable AC impedence, feeding, and ring voltage; On-Hook Voltage 29VDC
Caller-ID Type-1/2 FSK and ITU V.23/Bell 202 generation
• 2 or 4 FXO ports
2-wire loop-start presented as an RJ-11/12
2.5kV line isolation Surge Protection: Voice Ports: Tip & Ring protected by 270 V side actor
Off-hook and ring detection, Automatic line gain, Programmable ring count
End of Call detection, Line drop, busy tone, battery reversal detection
Hook-Flash Sending, Hook-Flash relay, DTMF send, detect, and relay
Caller ID FSK CLI reception and relay (Bellcore/ANSI and ETSI/ITU), Call routing based on Caller ID
Second dial-tone for two-stage DTMF dialing, Call routing based on DTMF numbers
Data Connectivity
10/100 Full Duplex, Autosensing, Ethernet RJ-45 port
Voice Processing (signalling dependent)
• Voice codes
G.711 A-Law/µ-Law (64kbps)
G.726 (ADPCM 40, 32, 24, 16 kpbs)
G.723.1 (5.3 or 6.3 kbps)
G.729ab (8kbps)
• Up to 8 parallel voice connections • G.168 echo cancellation • Carrier tone detection and generation • Silence suppression and comfort noise • Configurable dejitter buffer • Configurable tones (dial, ringing, busy) • RTP/RTCP (RFC 1889)
• Virtual Interfaces • Routing Criteria: Called party number (Destination) Calling party number (Source) Time of day, day of week, date • Longest prefix match, wildcard match, regular expression match • Number Manipulation Functions: Replace numbers Add/remove digits Regular Expressions • Fallback Routing: Soft Fallback to alternative interface or Call Router table
• Web GUI • Industry standard CLI with local console (CRJ-45, RS-232) and remote Telnet access • TFTP configuration & firmware loading • SNMP v1 agent (MIB II and private MIB) • Built-in diagnostic tools (trace, debug)
How to extend dial tone (dial tone extension) from one location to another location without a SIP server.
The dial tone generated by a PSTN connection or a PBX can be extended over the Internet or an Intranet by using a pair of Smartnode 452x units or a Smartnode 452x and a Smartlink 402x unit. This FAQ describes how to perform dial tone extension using a Smartnode 452x and a Smartlink 402x. Smartnode to Smartnode configuration information is available under "Line Extensions" of the SN452x applications notes page. Once the setup is complete, the end user will be able to lift the handset on the analog phone, hear the dial tone from the PSTN or PBX and then dial a number just as if they were directly connected to the PSTN or PBX. Flash-hook is supported so features like call waiting and call hold will work. Tax pulse tone (not typically used in the USA) is not supported.
The guide below assumes that WAN and LAN connectivity has already been established.
Hardware –
Smartlink 4021 or Smartlink 4022
Smartnode 4522, 4524, 4526 or 4528 with FXO ports (order code “JO” indicates FXO ports)
PSTN analog connection OR PBX with analog phone port
Analog phone
Physical Connectivity
Analog phone ----SL402x----- Internet/Intranet -----SN452x-------PSTN or PBX
In the SL4020
Under Telephony, SIP –
Set “SIP Registration Server Address:” to the IP address or domain name of the SmartNode 4520
Uncheck “Send Registration Request with Expire Time:”
Save the changes
Under Telephony, Phone 1
Set the phone number to the phone extension (example 111).
Optional - You can set caller id if you wish.
Under “Dial Out Type” , select Hotline from the drop down menu
Under “Hot Line Number”, enter the phone number that has been routed to the FXO port on the SN452x.
Save the changes
Go to System,Reload and restart the SL4020
In the SmartNode 452x....Please use the Smartnode CO configuration information under "Line Extensions" of the SN452x applications notes. You will need to add call routing and mapping table to strip the called number that is passed from the Smartlink.
routing-table called-e164 TAB-IN
route ... dest-interface IF_FXO0 STRIP
mapping-table called-e164 to called-e164 STRIP
map ...% to \1
map default to ''
How many VPN tunnels can be configured on the SL4020
The configurable limit is 8 now. Based on our preliminary study of code and hardware we believe this limit could be extended significantly. It will eventually be limited by memory and network resources.
Telephones
I would like to be able to push one button and immediately get a dial tone from the PSTN so that I can make calls. I DO NOT have SIP server.
Sounds like you want to do something like press "9" for an outside line and then dial number......
This can be accomplished by using a combination of features on the Smartlink phones or gateways along with a Smartnode serving as the gateway to the PSTN. The resulting design results in two calls being made - one SIP call from the Smartlink device to the SmartNode gateway and a second PSTN call from the Smartnode gateway to the destination address on the PSTN. How it works- The SL4050/SL4020 makes a SIP call to the Smartnode when the speed dial button is pushed. The Smartnode accepts the SIP call and presents dial tone from the FXO line. The user enters the number to be dialed which is transmitted as DMTF tones to the Smartnode that passes the tones to the PSTN. The PSTN completes the call to the number dialed.
Since there isn't SIP server, direct IP address or DNS calling can be used to send the SIP call directly to a Smartnode gateway with an PSTN connection.
To make calling easier on on the end user associate a speed dial button, like the telephone keypad number "9", with the IP address or DNS name of Smartnode.
Optionally the SL4050/10 phone has 10 line buttons that can be set as speed dial buttons. All the end user would do is push a line button and the next thing they would hear is a PSTN dial tone pass through by the Smartnode.
Please see the "FXO Interface Configuration" in the Smartnode configuration guide for information on setting up the Smartnode to pass dialtone from the PSTN to a SIP call.
From the SL4050 web configuration interface -
Select Line Key Settings
Pick a free line key and select "one touch dial"
Enter the DNS name or IP address of the Smartnode gateway that has the PSTN FXO connection
Click on "submit" to save the changes
SmartNode VoIP/ToIP
Call Routing
How can I remove or restrict Caller-ID (CLIP)?
There are two possibilities:
1. Set the ISDN Presentation Indicator (PI) to restricted:
172.16.40.125(ctx-cs)[switch]#mapping-table pi to pi MT-PI-TEST
172.16.40.125(map-tab)[MT-PI-T~]#map default to restricted
2. Delete the Calling-Party Nummer using a E.164 mapping table:
172.16.40.125(map-tab)[MT-PI-T~]#ble calling-e164 to calling-e164 MT-CNPN-TEST
172.16.40.125(map-tab)[MT-CNPN~]#map default to ""
Using Timeout and Termination Characters in Call-Routing Tabels
Call-Routing tables offer two possibilities to terminate overlap dialed numbers.
1. A dialling timeout
2. A special termination caracter like # or *
The timout and the caracter can be configured as follows:
For example:
172.16.40.125(ctx-cs)[switch]#routing-table called-e164 RT-CDPN-EX
172.16.40.125(rt-tab)[RT-CDPN~]#route 123T dest-interface Line0
According to this rule the dialed keys '12345#' will be immediatly matched and the number '12345' will be used without waiting for the timeout.
Special Cases:
The Termination Character can also be part of the rule, in which case it will NOT have the effect of cancelling the timeout period.
Examples:
Rule: #21#T
Dialled Keys: #21#1234
Effect: Timeout is aktive, used number: #21#1234
Rule: #21#T
Dialled Keys: #21#1234#
Effect: No Timeout, used number: #21#1234
Note: The first two dialled '#' do not cancell the timeout, they are part of the rule.
For a general digit-collection with timeout or termination caracter without any restrictions use the following rule:
Rule: T
In this case...
Dialled Keys: 1234
Effect: Timeout active, used number: 1234
or...
Dialled Keys: 1234#
Effect: No Timeout, used number: 1234
Do NOT use a rule as follows:
Rule: .*T
In this case...
Dialled Keys: 1234#
Effect: Timeout is STILL active because '#' matches the regular expression '.T', the used number will be: 1234#
Codecs
Why do I hear a crackling noise when using the G.729 codec?
On the SmartNodes (4110, 4520, 463X, 465X 4830, 491X, 492X, 493X, IC-4FXS) is not possible to use two low-bit-rate codecs at the same time on an FXS port. Thus you must choose to use either G.723 or G.729. G.711 is always supported.
Try this:
enable
configure
system
ic voice 0
low-bitrate-codec g729
In your VOIP profiles, we suggest you use either the G.723 codec or the G.729 codec, but not both, and it should match your low-bitrate-codec selection.
Do the SmartNodes support G.729B?
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?
The H.323 standard requires that H.323 endpoints (gateways, terminals etc.) be capable to process G.711. That is why the SmartNode always sends G.711 in the terminal capability set. If a specific codec is to be enforced a codec has to be specified in the interface (H.323) with the keywork exclusive.
How much bandwidth does a VoIP call use?
The required bandwdidth depends on various factors such as:
- Codec
- Codec samle length
- Protocol stack (IP, PPP, Frame-Relay, etc.)
- Tranmission Network (DSL, ATM, etc.)
- Echo Cancellation
As a general rule of thumb the bandwdith for one call in one direction is between 10 and 110 kb/s.
An excellent overview of how these parameters can be tuned and effect the VoIP bandwdith can be found in the following TechNote.
http://www.patton.com/technotes/smartnode_qos.pdf
What are the available Voice Codecs?
Codec
Net Bandwidth per call (kbps)
Used Bandwidth per call (kbps)
Min. Compression Delay (ms)
Usage
G.711 a-law
64
96
10
Uncompressed, best voice quality, European audio-digitizing
G.711 u-law
64
96
10
Uncompressed, best voice quality, American audio-digitizing
G.723.1
6.3
17
30
Good voice quality at lowest bandwidth, like analog phone, acceptable delay
G.729
G.729a
G.729b
8
40
10
Best relationship between voice quality and used bandwidth, low delay. G.729b is the silence suppression scheme for G.729a and is supported by the SmartNode family. For the sake of simplicity SmartWare and SmartNode always use the term G.729a, implicitly meaning G.729a and G.729b.
transparent
64
96
10
Transparent ISDN data, no echo cancellation
What must be observed with codec selection with H.323 FastConnect procedure in case of overlap dialing?
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?
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
How do I debug QoS?
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?
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 can I check if the routing tables are loaded successfully?
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).
What TCP & UDP Service Ports do I need to open to my SmartNode?
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?
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?
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'.
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.
How does a SmartNode work in combination with IP-Phones?
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?
SmartNodes offer a local console interface which can be connected to a separate device, for example analog modem. In this way the SmartNode can be managed through a dial-up connection if the IP Network is down or the SmarNode is mis-configured.
Keypad facility does not work with H.323.
In ISDN supplementary service can be invoked by means of thekeypad protocol. A service can be invoked with the digitsequence *21#. The phone sends these digits as information element 'Keypad Facility' and not as informationelement 'called party number'. As the H.323 protocol does not dispose of a way to transport 'keypad facility'information it gets lost.On the contrary H.323+ (H.323 Annex M3) is a tunneling protocol for the transparent transport of ISDN over H.323and thus inherently supports 'keypad facility'.Note that on some phones it is required to explicitly switch to keypad facility mode in order to send a specificdigit sequence as info element 'keypad facility'.
Is DTMF supported in SmartWare?
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?
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?
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?
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?
Yes, the command 'show port ethernet' shows the current Ethernet configuration along with the MAC addresses.
Hardware Interfaces
Why does Ethernet not work when connected a Laptop with PC-Card Ethernet interface to a SmartNode using a crossover cable?
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.
What is FXS & FXO?
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
Licenses
Why do I need a License Key for Release 3.10?
SmartWare is the embedded software running on the SmartNodes. SmartWare offers a number of feature options such as QSIG, VPN and IP forwarding. Up to SmartWare release 2.20 feature options had to be paid but where not keyed. Starting with release 3.10 a license key has to be installed to enable the feature options.
Note that some product bundles include some of the feature options i.e. SN1200/2VIL/UI includes IP Forwarding. The "I" in the model code stands for the IP forwarding license.
Q. Do I need a License Key for every SmartNode?
Yes the License Keys are specific to the feature option and the serial number of the SmartNode. The keys can not be transferred from one unit to another.
Q. Where can I buy Licenses?
Feature Options can be purchased through the regular SmartNode distribution channels.
Q. Where can I get License Keys for feature options purchased together with SmartWare 2.20?
The License Keys for SmartNodes delivered with SmartWare 2.20 can be requested using the following web form: Liscense Request Form
How do I install a license key?
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
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?
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?
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?
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?
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)?
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
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
How many SIP users can be supported on a SmartNode?
For all intents and purposes a maximum number of 100 "SIP users" can be supported on a SmartNode
Can I bind multiple SIP Gateways to the same IP Interface?
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
How do I send a hook FLASH to a SIP Provider to use services like three-way conferencing?
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?
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?
SmartWare FLASH Codes
-FLASH 0 - keep current, reject incoming
-FLASH 1 - drop current, accept incoming
-FLASH 2 - hold current, accept incoming
To toggle between the active and the held call, press flash-hook, followed by the "2" key.
Additional Call Offering
To enable aditional call offering, configure the fxs port of the SN with the command:
additional-call-offering
1) Press FLASH, then the first call is placed on hold and you will hear a new dial tone.
2) Dial the number of the second call.
3) If you press FLASH, you may change between the two calls.
4) When you hang-up on the phone, the two other parties are connected together. Sorry, three way conferencing is not yet supported.
Upgrading/TFTP
Using Encrypted TFTP
Encrypted
Configuration Download
-
An external encryption tool on the PC is used to encrypt the
configuration file:
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
What happens if the software upgrade on a SmartNode fails?
Each SmartNode is equipped with a bootloader application. If an upgrade fails and no valid firmware is available on the system the SmartNode will start in this bootloader mode. The bootloader will allow you to install a new firmware.
Please refer to the user documentation on how to operate in bootloader mode.
Note that the bootloader can not be replaced.
Where can I get a TFTP Server to load in my configuration or upgrade my SmartNode Software?
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?
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?
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
Can I do encrypted VoIP calls with the SmartNode IPSec?
Yes, with SmartWare software releases dated 3/1/06 and later. For earlier relases, VoIP calls terminated on the SmartNode route the RTP outside the VPN tunnel.
A VPN feature license has to be installed for this feature to work.
Gateways 