Introducing Voice Gateways Part 2

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    Introducing Voice

    Gateways Part 2

    Voice Ports, Signaling, Trunks, and Timers

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    Analog Voice Port Interfaces

    Three types of analog voice interfaces are supported byCisco gateways:

    FXS:

    • connects the router or access server to end-user equipment

    such as telephones, fax machines, or modems

    • supplies ring, voltage, and dial tone to the station

    • includes an RJ-11 connector for basic telephone equipment,key sets, and PBXs

    • stands for Foreign Exchange Station

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    Analog Voice Port Interfaces

    Three types of analog voice interfaces are supported byCisco gateways:

    FXO:

    • used for trunk, or tie-line, connections to a PSTN CO or to a

    PBX that does not support E&M signaling

    • standard RJ-11 modular telephone cable connects the FXOvoice interface card to the PSTN or PBX through a telephonewall outlet

    • stands for Foreign Exchange Office

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    Analog Voice Port Interfaces

    Three types of analog voice interfaces are supported byCisco gateways:

    E&M:

    • trunk circuits that connect telephone switches to one another 

    • do not connect end-user equipment to the network

    • uses special signaling paths that are separate from the audiopath

    • signaling paths are known as the E-lead and M-Lead

    • stands for Ear and Mouth, Earth and Magneto, Receiveand Transmit

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    Analog Signaling

    Signaling techniques can be placed into one of threecategories:

    Supervisory: Involves the detection of changes to the status ofa loop or trunk. When these changes are detected, the

    supervisory circuit generates a predetermined response. Forexample a circuit (loop) can close to connect a call

    Addressing: Involves passing dialed digits (pulsed or tone) to aPBX or CO. These dialed digits provide the switch with aconnection path to another phone

    Informational: Provides audible tones to the user, whichindicate certain conditions such as an incoming call or a busyphone

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    FXS and FXO Supervisory Signaling

    FXS and FXO interfaces indicate on-hook or off-hookstatus and the seizure of telephone lines by one of twoaccess signaling methods: loop-start or ground-start

    The type of access signaling is determined by the type

    of service from the telephone company’s CO.

    Standard home telephone lines use loop-start, butbusiness telephones can order ground-start linesinstead

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    Loop-Start

    Loop-start is the more common of the access signalingtechniques

    When a handset is picked up (the telephone goes off-hook) this action closes the 48V circuit that draws

    current from the telephone company CO

    This indicates a change in status, which signals the COto provide a dial tone

     An incoming call is signaled from the CO to the calledhandset by sending a signal in a standard on/offpattern, which causes the telephone to ring

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    Loop-Start

    The loop-start signaling process is as follows:

    1. In the idle state, the telephone has an open two-wire loop(tip and ring lines open). The CO or FXS waits for a closedloop, which allows current to flow. The CO or FXS has aring generator connected to the tip line and –48VDC on thering line

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    Trivia!

    Why are the wires called “ring” and “tip”?

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    Loop-Start

    The loop-start signaling process is as follows:

    2. A telephone set closes the loop between the tip and ringlines. The CO or FXS module detects current flow and thengenerates a dial tone, which is sent to the telephone set.This indicates that the customer can start to dial.

    The CO or FXS module seizes the line of the telephonebeing called causing the phone to ring. The CO or FXSmodule removes this ring after the telephone set closes thecircuit between the tip and ring lines.

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    Loop-Start

    The loop-start signaling process is as follows:

    3. The telephone set closes the circuit when the called partypicks up the handset. The CO or FXS module closes thecircuit when it has an available resource to connect to thecalled party.

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    Loop-Start

    Loop-start has two disadvantages:

    There is no way to prevent the CO and the subscriber fromseizing the same line at the same time, known as glare.

    It does not provide switch-side disconnect supervision for FXO

    calls. The CO switch expects an FXO interface on a router tohang up the calls it receives (like a telephone would), howeverthis function is not built into the FXO port for received calls, onlyfor calls originating from the FXO port

    These disadvantages are usually not a problem onresidential telephones, but they become significant withthe higher call volume experienced on businesstelephones.

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    Ground-Start

    Ground-start signaling is primarily used in switch-to-switch connections

    The main difference between ground-start and loop-start signaling is that ground-start requires ground

    detection to occur in both ends of a connection beforethe ring and tip loops can be closed

    Because ground-start signaling uses a request and/orconfirm switch at both ends of the interface it is

    preferable over other signaling methods on high-usagetrunks

    For this reason, ground-start signaling is typically usedon trunk lines between PBXs and in businesses where

    call volume is high

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    Ground-Start

    The ground-start signaling process is as follows:

    1. In the idle state, both the tip and ring lines are disconnectedfrom ground. The PBX/Router constantly monitor the tip linefor ground, and the CO constantly monitors the ring line forground. The battery ( –48VDC) is still connected to the ringline just as in loop-start signaling.

    PBX/Router 

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    Ground-Start

    The ground-start signaling process is as follows:

    2. A PBX/Router grounds the ring line to indicate to the COthat there is an incoming call. The CO senses the ringground and then grounds the tip lead to let the PBX/Routerknow that it is ready to receive the incoming call

    PBX/Router 

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    Ground-Start

    The ground-start signaling process is as follows:

    3. The PBX/Router senses the tip ground and closes the loopbetween the tip and ring lines in response. It also removesthe ring ground

    PBX/Router 

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    Analog Address Signaling

    The dialing phase allows the subscriber to enter aphone number (address) of a telephone at anotherlocation

    The customer enters this number with either a rotary

    phone that generates pulses or a touch-tone (push-button) phone that generates tones

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    Analog Address Signaling

    Telephones use two different types of address signalingto notify the telephone company where a subscribercalls

    Pulse dialing

    DTMF dialing

    These pulses or tones are transmitted to the CO switchacross a two-wire twisted-pair cable (tip and ring lines)

    On the voice gateway, the FXO port sends addresssignaling to the FXS port. This address indicates thefinal destination of a call

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    Pulse Dialing

    Pulses were used by the old rotary phones

     As the rotary disk rotated it opened and closed thecircuit a specified number of times based on how farthe disk was turned

    The exchange equipment counted those circuitinterruptions to determine the called number 

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    DTMF

    Dual Tone Multi-frequency (DTMF) tones are morecommon today

     A specific tone (consisting of two separate frequencies)is assigned to each key on a touch-tone dial pad,

    indicating the digits dialed

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    Informational Signaling

    FXS ports provide informational signaling using callprogress (CP) tones

    These tones are audible and are used by the FXSconnected device to indicate the status of calls

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    E&M Signaling

    E&M is another signaling technique used mainlybetween PBXs or other network-to-network telephonyswitches

    Instead of superimposing both voice and signaling on

    the same wire, E&M uses separate paths, or leads, foreach

    There are six distinct physical configurations for thesignaling part of the interface. They are Types I –V and

    Signaling System Direct Current No. 5 (SSDC5)

    They use different methods to signal on-hook or off-hook status

    Note: Cisco supports types I, II, III, and V

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    E&M Signaling

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    E&M Supervisory Signaling

    On-hook/off-hook status and telephone line seizure isindicated using any of three types of access signaling:

    Immediate-start: Immediate-start is the simplest method ofE&M access signaling.

    The calling side seizes the line by going off-hook on its E lead,waits for a minimum of 150ms, and then sends addressinformation as DTMF digits or as dialed pulses.

    This signaling approach is used for E&M tie trunk interfaces.

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    E&M Supervisory Signaling

    Immediate-start signaling:

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    E&M Supervisory Signaling

    On-hook/off-hook status and telephone line seizure isindicated using any of three types of access signaling:

    Wink-start: Wink-start is the most common E&M signalingmethod and the default for E&M voice ports.

    Wink-start was developed to minimize glare.

    The calling side seizes the line by going off-hook on its E leadthen waiting for a short temporary off-hook pulse, or “wink,” fromthe other end on its M lead before sending address informationas DTMF digits.

    The switch interprets the pulse as an indication to proceed andthen sends the dialed digits as DTMF or dialed pulses.

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    E&M Supervisory Signaling

    Wink-start signaling:

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    E&M Supervisory Signaling

    On-hook/off-hook status and telephone line seizure isindicated using any of three types of access signaling:

    Delay-start: With delay-start signaling the calling station seizesthe line by going off-hook on its E lead.

     After a timed interval, the calling side looks at the status of thecalled side. If the called side is on-hook, the calling side startssending information as DTMF digits.

    Otherwise, the calling side waits until the called side goes on-hook and then starts sending address information.

    This signaling approach is used for E&M tie trunk interfaces

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    E&M Supervisory Signaling

    Delay-start signaling:

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    FXS Voice Port Configuration

    FXS ports look like switches to the edge devices thatare connected to them, therefore configuration of theFXS port should emulate the switch configuration of thelocal PSTN

    In North America, this means using the default settingsmost of the time

    In other countries, this usually means the defaultsettings must be changed

    Let’s look at an example

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    FXS Voice Port Configuration

    Configure a voice gateway to route calls to a POTSphone connected to an FXS port on a remote router inGreat Britain

    The British office is configured to enable ground-start

    signaling on FXS voice port 0/2/0

    The call-progress (CP) tones are set for Great Britain,and the ring cadence is set for pattern 1

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    FXS Voice Port Configuration

    Step 1:

    Enter voice-port configuration mode:

    Router(config)# voice-port slot/port

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    FXS Voice Port Configuration

    Step 2:

    Select the access signaling type to match the telephonyconnection you are making.

    Router(config-voiceport)# signal {loopstart | groundstart}

    NOTE: If you change signal type, you must execute a shutdownand a no shutdown command on the voice port

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    FXS Voice Port Configuration

    Step 3:

    Select the two-letter locale for the voice call progress tones andother locale-specific parameters to be used on this voice port.

    Router(config-voiceport)# cptone locale

    NOTE: You can use the ? command to see a list of these two-

    letter locales

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    FXS Voice Port Configuration

    Step 4:

    Specify a ring pattern. Each pattern specifies a ring-pulse timeand a ringinterval time.

    Router(config-voiceport)# ring cadence { pattern-number | define pulse interval}

    NOTE: The patternXX keyword provides pre-set ring-cadencepatterns. The define keyword allows you to create a custom

    ring cadence.

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    FXS Voice Port Configuration

    Step 5:

     Activate the voice port.

    Router(config-voiceport)# no shutdown

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    FXS Voice Port Configuration

    The completed FXS voice port configuration:

    Router(config)# voice-port 0/2/0

    Router(config-voiceport)# signal groundstart

    Router(config-voiceport)# cptone GB

    Router(config-voiceport)# ring cadence pattern01

    Router(config-voiceport)# no shutdown

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    FXO Voice Port Configuration

     An FXO trunk is one of the simplest analog trunksavailable

    Because DNIS information can be sent outbound only,no DID is possible on FXO ports (however ANI is

    supported for inbound calls)

    Two signaling types exist, loop-start and ground-start,with ground-start being the preferred method

    Let’s look at an example

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    FXO Voice Port Configuration

    Configure a voice gateway to route calls to and fromthe PSTN through an FXO port on the router usingground-start signaling

    You must set up a private line automatic ringdown

    (PLAR) connection for the FXO port to automaticallycall extension 4001 whenever an incoming call isreceived from the PSTN

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    FXO Voice Port Configuration

    Because an FXO trunk does not support DID, two-stage dialing is required for all inbound calls

    If all inbound calls should be routed to a specificextension (for example, a front desk), you can use the

    connection plar opx command

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    FXO Voice Port Configuration

    Step 1:

    Enter voice-port configuration mode.

    Router(config)# voice-port 0/0/0

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    FXO Voice Port Configuration

    Step 2:

    Select the access signaling type to match the telephonyconnection you are making.

    Router(config-voiceport)# signal ground-start

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    FXO Voice Port Configuration

    Step 3:

    Specify a PLAR off-premises extension (OPX) connection.

    Router(config-voiceport)# connection plar opx 4001

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    FXO Voice Port Configuration

    Step 4:

     Activate the voice port.

    Router(config-voiceport)# no shutdown

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    FXO Voice Port Configuration

    Step 5:

    We are going to take the example one step further andconfigure the outbound dial peer, so exit voice-portconfiguration mode.

    Router(config-voiceport)# exit

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    FXO Voice Port Configuration

    Step 6:

    Create a standard dial peer for inbound and outbound PSTNcalls.

    Router(config)# dial-peer voice 90 pots

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    FXO Voice Port Configuration

    Step 7:

    Specify the destination pattern.

    Router(config-dialpeer)# destination-pattern 9T

    Note: The 9T string matches any outbound call starting with thedigit 9 (which is commonly used in enterprise networks to get anoutside line)

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    FXO Voice Port Configuration

    Step 8:

    Specify the voice port associated with this dial peer.

    Router(config-dialpeer)# port 0/0/0

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    FXO Voice Port Configuration

    The completed FXO voice port configuration:

    Router(config)# voice-port 0/0/0

    Router(config-voiceport)# signal groundstart

    Router(config-voiceport)#connection plar opx 4001

    Router(config)# dial-peer voice 90 pots

    Router(config-dialpeer)# destination-pattern 9T

    Router(config-dialpeer)# port 0/0/0

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    E&M Voice Port Configuration

    Configuring an E&M analog trunk is straightforward

    Three key options have to be set:

    • The E&M signaling type

    • Two- or four-wire operation• The E&M type

    These settings are usually dictated by the device youare connecting to (PBX/Switch)

    Let’s look at an example

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    E&M Voice Port Configuration

    Configure a voice gateway to work with an existing PBXsystem

    Use wink-start supervisory signaling

    Use 2-wire operation mode Use Type I E&M signaling

    Use a POTS dial peer for the extensions behind thePBX

     Allow DID support

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    E&M Voice Port Configuration

    Step 1:

    Enter voice port configuration mode.

    Router(config)# voice-port 1/1/1

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    E&M Voice Port Configuration

    Step 2:

    Select the access signaling type to match the telephonyconnection you are making.

    Router(config-voiceport)# signal wink-start

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    E&M Voice Port Configuration

    Step 3:Select a specific cabling scheme for the E&M port.

    Router(config-voiceport)# operation 2-wire

    Note: Changing this setting requires shutting down the voiceport and bringing it back up for the changes to take effect

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    E&M Voice Port Configuration

    Step 4:Specify the type of E&M interface.

    Router(config-voiceport)# type 1

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    E&M Voice Port Configuration

    Step 5: Activate the voice port.

    Router(config-voiceport)# no shutdown

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    E&M Voice Port Configuration

    Step 6:Exit voice-port configuration mode.

    Router(config-voiceport)# exit

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    E&M Voice Port Configuration

    Step 7:Create a dial peer for the POTS phones.

    Router(config)# dial-peer voice 10 pots

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    E&M Voice Port Configuration

    Step 8:Specify the destination pattern for the POTS phones.

    Router(config-dialpeer)# destination-pattern 1...

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    E&M Voice Port Configuration

    Step 9:Specify direct inward dialing.

    Router(config-dialpeer)# direct-inward-dial

    NOTE: The DID configuration is needed when the POTSphones call the IP phones.

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    E&M Voice Port Configuration

    Step 10:Specify digit forwarding for all digits, so that no digits will bestripped as they are forwarded out of the voice port. By default,only digits matched by wildcard characters in the destination- pattern command are forwarded.

    Router(config-dialpeer)# forward-digits all

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    E&M Voice Port Configuration

    Step 11:Specify the voice port associated with this dial peer.

    Router(config-dialpeer)# port 1/1/1

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    E&M Voice Port Configuration

    The complete E&M voice port configuration:

    Router(config)# voice-port 1/1/1

    Router(config-voiceport)# signal wink-start

    Router(config-voiceport)# operation 2-wire

    Router(config-voiceport)# type 1

    Router(config-voiceport)# no shutdown

    Router(config-voiceport)# exit

    Router(config)# dial-peer voice 10 pots

    Router(config-dialpeer)# destination-pattern 1...

    Router(config-dialpeer)# direct-inward-dial

    Router(config-dialpeer)# forward-digits all

    Router(config-dialpeer)# port 1/1/1

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    Trunks

    Trunks interconnect gateways or PBX systems to othergateways, PBX systems, or the PSTN

     A trunk is a single physical interface or a logicalinterface that contains several physical interfaces and

    connects to a single destination (think about trunks inLAN switching)

    This could be a single FXO port that provides a singleline connection between a Cisco gateway and an FXS

    port of a small PBX system Or it could be several T1 interfaces with 24 lines each

    in a Cisco gateway providing PSTN lines to severalhundred subscribers

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    Trunks

    Trunk ports can be analog or digital and use a variety ofsignaling protocols

    Signaling can be done using either the voice channel(in-band) or an extra dedicated channel (out-of-band)

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    Analog Trunks

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    Analog Trunks

    PSTN carriers typically offer analog trunk features thatcan be supported on home phones

     A description of several of these features can be foundon pages 78 and 79 of the textbook

    Some of these features include caller ID, call waiting,last-number redial, etc.

     As shown in the next example, caller ID requiresspecial consideration when used on analog trunks

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    Analog Trunks

    The following diagram shows a small business voicenetwork connected through a gateway to the PSTN

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    Analog Trunks

    The voice network supports both analog and IPphones. The connection to the PSTN is via an FXOport and the analog phone is connected to an FXS porton the gateway

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    Analog Trunks

    The issue in this scenario is how the caller ID is passedto call destinations

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    Analog Trunks

    Call 1 is from the analog phone to the IP phone on thesame premises. The FXS port is configured with astation ID and station ID number 

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    Analog Trunks

    When a call is placed from the analog phone to the IPphone on the same premises the caller name andnumber are displayed on the screen

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    Analog Trunks

    Call 2 is placed from the same analog phone but thedestination is off premises on the PSTN

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    Analog Trunks

    The FXO port forwards the station ID information to thePSTN. The CO switch discards this information andreplaces it with information it has configured for thisconnection

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    CAMA Trunks

     A Centralized Automated Message Accounting(CAMA) trunk is a special analog trunk mainly used foremergency call services (911 and E911 services)

    You can use CAMA ports to connect to a Public Safety

    Answering Point (PSAP) for emergency calls

    CAMA trunks are currently used in 80% of E911networks

     Always check local regulations regarding 911 and

    Enhanced 911 (E911) requirements whenimplementing an IP telephony solution

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    CAMA Trunks

    When CAMA trunks are used for E911 services, thecalling number (ANI) is needed at the PSAP for tworeasons

    The calling number is used to reference the Automatic LocationIdentification (ALI) database to find the exact location of the

    caller The calling number is used as a callback number in case thecall is disconnected.

    Calls to emergency services are routed based on thecalling number (not the called number)

    The calling number is checked against a database ofemergency service providers to route the call to theproper PSAP, which dispatches services to the callerlocation

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    CAMA Trunks

    Cisco VIC2-2FXO and VIC2-4FXO cards supportCAMA via software configuration

    When configuring CAMA signaling there are fiveoptions:

    KP-0-NXX-XXXX-ST

    KP-0-NPA-NXX-XXXX-ST

    KP-0-NPA-NXX-XXXX-ST-KP-YYY-YYY-YYYY-ST

    KP-2-ST

    KP-NPD-NXX-XXXX-ST

    Each option transmits different information about thecaller to the PSAP

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    CAMA Configuration Example

    Step 2:Configure a dedicated dial peer to route emergency calls usingthe CAMA trunk when a user dials 911.

    Router(config)# dial-peer voice 911 pots

    Router(config-dialpeer)# destination-pattern 911

    Router(config-dialpeer)# prefix 911

    Router(config-dialpeer)# port 1/1/1

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    CAMA Configuration Example

    Step 3:Configure a dedicated dial peer to route emergency calls usingthe CAMA trunk when a user dials 9911.

    Router(config)# dial-peer voice 9911 pots

    Router(config-dialpeer)# destination-pattern 9911

    Router(config-dialpeer)# prefix 911

    Router(config-dialpeer)# port 1/1/1

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    CAMA Configuration Example

    Step 4:Configure a standard PSTN dial peer for all other outbound(local) PSTN calls.

    Router(config)# dial-peer voice 910 pots

    Router(config-dialpeer)# destination-pattern 9[2-9].........

    Router(config-dialpeer)# port 0/0/0:23

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    Direct Inward Dialing Trunk

    The DID service enables callers to dial an extensiondirectly on a PBX or a VoIP system (e.g. Cisco UnifiedCommunications Manager) without the assistance of anoperator or automated call attendant

    This service makes use of DID trunks, which forwardonly the last three to five digits of a phone number tothe PBX, router, or gateway

    For example, a company has phone extensions 1000 to

    1999. A caller dials 555-1234, and the local COforwards 1234 to the PBX or VoIP system

    The PBX or VoIP system then rings extension 1234

    This entire process is transparent to the caller 

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    Direct Inward Dialing Trunk

    Because an FXS DID trunk can receive only inboundcalls, a combination of FXS-DID, and FXO ports isrequired for inbound and outbound calls

    The following configuration example uses an FXS-DID

    trunk for inbound calls and a standard FXO trunk foroutbound calls

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    Direct Inward Dialing Trunk

    Step 1:Configure the FXS port for DID and wink-start.

    Router(config)# voice-port 0/0/0

    Router(config-voiceport)# signal did wink-start

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    Direct Inward Dialing Trunk

    Step 2:Configure the FXO port for ground-start signaling.

    Router(config)# voice-port 0/1/0

    Router(config-voiceport)# signal groundstart

    Di I d Di li T k

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    Direct Inward Dialing Trunk

    Step 3:Create an inbound dial peer using the FXS DID port. Note thatDID is enabled.

    Router(config)# dial-peer voice 1 potsRouter(config-dialpeer)# incoming called-number .

    Router(config-dialpeer)# direct-inward-dial

    Router(config-dialpeer)# port 0/0/0

    Di t I d Di li T k

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    Direct Inward Dialing Trunk

    Step 4:

    Create a standard outbound dial peer using the FXO port.

    Router(config)# dial-peer voice 910 pots

    Router(config-dialpeer)# destination-pattern 9[2-9].........

    Router(config-dialpeer)# port 0/1/0

    Ti d Ti i

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    Timers and Timing

    You can set a number of timers and timing parametersfor fine-tuning a voice port:

    timeouts initial seconds: This value controls how long thedial tone is presented before the first digit is expected. Thistimer value typically does not need to be changed

    timeouts interdigit seconds: Configures the number of

    seconds for which the system will wait between caller-entereddigits before sending the input to be assessed. Default is 10seconds

    timeouts ringing {seconds | infinity}: Configures thelength of time a caller can continue to let the telephone ringwhen there is no answer. Default is 180 seconds

    Ti d Ti i

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    Timers and Timing

    You can set a number of timers and timing parametersfor fine-tuning a voice port:

    timing digit milliseconds: Configures the DTMF digitsignal duration for a specified voice port. If a user or devicedials too quickly, the digit might not be recognized.

    timing interdigit milliseconds: Configures the DTMF

    interdigit duration for a specified voice port

    timing hookflash-[input | output] milliseconds:

    Configure the maximum duration (in milliseconds) of a

    hookflash indication.

    Changing these timers is often useful for someone witha disability that may require more time to dial digits

    V if i V i P t

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    Verifying Voice Ports

    Step 1:

    Pick up the handset of an attached telephony device and checkfor a dial tone. If there is no dial tone, check the following:

    • Is the plug firmly seated?

    • Is the voice port enabled?• Is the router running the correct version of Cisco IOS to

    recognize the module?

    Step 2:

    If you have a dial tone, dial a DTMF digit. If the dial tone stopswhen you dial a digit, the voice port is probably configuredproperly.

    V if i V i P t

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    Verifying Voice Ports

    Step 3:

    Use the show voice port command to verify that the data

    configured is correct:

    Router# show voice port

    Foreign Exchange Station 0/0/0 Slot is 0, Sub-unit is 0, Port is 0

    Type of VoicePort is FXS VIC2-2FXS

    Operation State is DORMANT

    Administrative State is UP

    No Interface Down Failure

    Description is not set

    Noise Regeneration is enabled

    Non Linear Processing is enabled

    Non Linear Mute is disabled

     

    V if i V i P t

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    Verifying Voice Ports

    Step 4:

    Use the show voice port summary command to make sure

    the port is enabled. If the port is administratively down, use theno shutdown command. If the port was working previously and

    is not working now, it is possible the port is in a hung state. Use

    the shutdown/no shutdown command sequence to reinitializethe port.

    Router# show voice port summary

    IN OUT

    PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==

    0/0/0  —  fxs-ls up dorm on-hook idle y

    0/0/1  —  fxs-ls up dorm on-hook idle y

    V if i V i P t

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    Verifying Voice Ports

    Step 5:

    If you have configured E&M interfaces, make sure the valuesassociated with your specific PBX setup are correct.Specifically, check for two-wire or fourwire, wink-start,immediate-start, or delay-start signaling types, and the E&M

    interface type. These parameters need to match those set onthe PBX for the interface to communicate properly.

    Step 6:

    You must confirm that the voice interface card (VIC) (that is, the

    module in the router that contains the voice ports) is correctlyinstalled. With the device powered down, remove the VIC, andreinsert it to verify the installation. If the device has other slotsavailable, try inserting the VIC into another slot to isolate theproblem.

    V if i V i P t

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    Verifying Voice Ports

    Some handy commands for testing voice ports:

    test voice port identifier inject-tone: Injects a test

    tone into a voice port

    test voice port identifier loopback: Performs a

    loopback test on a voice port

    csim start dial-string : Simulates a call to the specified dialstring (very useful when testing dial plans). This is andunsupported and undocumented command

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