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Cisco 800 Series Routers Software Configuration Guide
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About This Guide
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Concepts
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Configuring Basic Networks
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Configuring Advanced Networks
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Network Scenarios
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Configuring Remote CAPI
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Configuring Telephone Interfaces
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Router Feature Configurations
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Advanced Router Configuration
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Troubleshooting
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Cisco IOS Basic Skills
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ROM Monitor
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Common Port Assignments
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Provisioning the ISDN Line
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ISDN BRI Cause Values
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Table Of Contents
Configuring Telephone Interfaces
Configuring Physical Characteristics
What You Need to Know About SPIDs
Forwarding Incoming ISDN Voice Calls to Connected Devices
Configuring Advanced Telephone Features
Data over Voice Bearer Service
How to Use Telephones Connected to Cisco 800 Series Routers
Disabling Pound Key End-of-Call Function
POTS Dial Feature (Japan Only)
Activating the POTS Dial Feature
Call Scenarios for the POTS Dial Feature
Cisco 813 Router Enhanced Voice Features
(Japan Only)General Requirements and Restrictions
Requirements for Activating Caller ID Display
Requirements for Activating Call Blocking on Caller ID
Configuring Call Blocking on Caller ID
Requirements for Activating Local Call Waiting
Configuring Local Call Waiting
Requirements for Activating E Ya Yo
Requirements for Activating Voice Warp
Requirements for Activating Voice Select Warp
Requirements for Activating Nariwake
Requirements for Activating Trouble-Call Blocking
Configuring Trouble-Call Blocking
Requirements for Activating I Number
Example of Silent Fax Calls Configuration
Example of Silent Fax Calls Configuration Output
Supplementary Telephone Services for the Net3 Switch
Requirements for Supplementary Telephone Services Support
Configuring Caller ID for the Net3 Switch
Call Forwarding for the Net3 Switch
Configuring the Call Forwarding Method
Configuring the Call Forwarding Service
Cisco 804 and 813 Routers Enhanced Voice Features
Calling Between Telephone Ports
Activating the Calling Between Telephone Ports Feature
Calling Between Telephone Ports Scenarios
Activating the Call Transfer Feature
Call Transfer Feature Scenarios
Volume Adjustment Configuration Example
Volume Adjustment Configuration Output Example
Distinctive Ringing Based on Caller ID
Configuring Distinctive Ringing Based on Caller ID
Distinctive Ringing Configuration Example
Distinctive Ringing Configuration Output Example
Configuring Subaddresses for POTS Ports
Subaddressing Configuration Example
Subaddressing Configuration Output Example
Caller ID on the Cisco 813 Router
Configuring Local Call Forwarding
Local Call Forwarding Scenarios
Local Call Forwarding Configuration Example
Local Call Forwarding Configuration Output Example
PIAFS Configuration Output Example
Configuring Telephone Interfaces
The term telephone port refers to the physical port on the router back panel. The term telephone interface refers to a logical interface that you must configure to make an analog telephone or fax connected to a telephone port work properly.
This chapter describes how to configure standard and advanced features of the those Cisco 800 series routers supporting telephone features (Cisco 803, 804, and 813 routers). These routers support push-button analog telephones only; the Cisco routers do not support rotary telephones. This chapter also describes how to use the connected devices.
Physical Characteristics
This section discusses the following:
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Physical characteristics that you must configure
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Configuring Physical Characteristics
Starting in global configuration mode, use these steps to configure physical characteristics. For information on the commands used in this table, refer to the Cisco IOS documentation set.
Step 1
pots country country
Enter the pots country ? command to get a list of supported countries and the code you must input to indicate a particular country. By specifying a country, you are configuring your telephone to use country-specific default settings for each physical characteristic. If you need to change a country-specific default setting, you can use the optional commands described in this table.
Step 2
pots line-type
{type1 | type2 | type3}Optional. Set the line type. Line type 1 runs at 600 ohms, line type 2 runs at 900 ohms, and line type 3 runs at 300 or 400 ohms. Lines in the U.S. typically run at 600 ohms (line type 1).
Step 3
pots dialing-method
{overlap | enblock}Optional. Set the dialing method. If you select overlap, the router transmits each digit dialed in a separate message. If you select enblock, the router collects all digits dialed and transmits in one message. To interrupt collection and transmission of dial-string digits, enter pound sign (#) or stop dialing digits until a timer runs out.
Step 4
pots disconnect-supervision
{osi | reversal}Optional. Set how the router notifies the connected device when calling party has hung up. Japan typically uses the reversal option. Most other countries use the osi option.
Step 5
pots encoding {alaw | ulaw}
Optional. Set the pulse code modulation (PCM) encoding scheme. Europe typically uses the alaw option. North America typically uses the ulaw option.
Step 6
pots tone-source
{local | remote}Optional. Set who supplies dial, ringback, and busy tones. If you select local, the router supplies the tones. If you select remote, the telephone switch provides the tones. For more information, refer to the "Tones for NET3 Switch" section.
Step 7
pots ringing-freq
{20Hz | 25Hz | 50Hz}Optional. Set the frequency at which telephone ports ring.
Step 8
pots disconnect-time interval
Optional. If a connected device, such as an answering machine, fails to detect that a calling party has hung up, you can adjust the interval at which selected disconnect supervision method is applied. Interval is from 50 to 2000 milliseconds.
Step 9
pots silence-time interval
Optional. If a connected device, such as an answering machine, fails to detect that a calling party has hung up, you can adjust the interval of silence after a hang-up. Interval is from 0 to 10 seconds.
Step 10
pots distinctive-ring-guard-
time millisecondsOptional. Set the delay, in milliseconds (0 to 1000), before a telephone port can be rung after a previous call is disconnected. For more information, refer to the "Distinctive Ringing" section.
Step 11
show pots status
Optional. Display settings of physical characteristics as well as other information on telephone interfaces.
Tones for NET3 Switch
By default, the Cisco 800 series routers are configured so that the telephone switch supplies tones, such as dial, ringback, and busy tones. However, NET3 switches, which are used in Europe, do not provide these tones. You can use the pots tone-source local command from global configuration mode to configure the router instead of the telephone switch to provide these tones.
Note
If the pots dialing-method command is set to enblock, the router provides the internal dial tone.
REN
You can connect multiple devices (analog telephone or fax machine) to a router telephone port. The number of devices that you can connect depends on the following:
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If the REN of each device you plan to connect is one, then you can connect a maximum of five devices to that particular telephone port.
Creating Dial Peers
You can create a dial peer to determine how incoming calls are routed to the telephone ports. You can create a total of six dial peers for the two telephone ports. There are no restrictions on how many dial peers you can create per port; for example, you can create six dial peers for port 1 and zero on port 2.
Starting from global configuration mode, use the steps below to create a dial peer.
Step 1
dial-peer voice tag pots
Set up tag number (1 through 6) for dial peer.
Step 2
destination-pattern ldn
Specify local ISDN directory number assigned to telephone interface. Do not specify an area code.
Step 3
port port-number
Specify number (1 or 2) associated with telephone port.
Step 4
no call-waiting
Optional. Disable call waiting.
Step 5
ring cadence-number
Optional. Set up distinctive ring (0
through 2). For more information, see the "Distinctive Ringing" section.Step 6
show dial-peer voice [tag]
Optional. Display all or a particular dial-peer configuration (1 through 6).
For example, if you have connected one voice device (555-1111) to port 1 and another (555-2222) to port 2, you can create two dial peers. The following output example shows two dial peers:
dial-peer voice 1 pots destination-pattern 5551111 port 1 no call-waiting ring 0dial-peer voice 2 pots destination-pattern 5552222 port 2 no call-waiting ring 0When a caller dials 555-1111, the call is routed to port 1. When a caller dials 555-2222, the call is routed to port 2. If the dial peers are not created, calls to both numbers are routed to port 1.
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What You Need to Know About SPIDs
North America uses SPIDs to identify subscribed services. The SPID format is generally an ISDN telephone number with several numbers added to it, such as 40855511110101. Your ISDN line could be assigned zero, one, or two SPIDs.
You must associate a SPID with an ISDN directory number and a telephone port number by using the isdn spid1 and isdn spid2 commands in global configuration mode and the port command in dial peer configuration mode. Make sure that you specify all the ISDN directory numbers provided by your telephone service provider in the isdn spid1 and isdn spid2 commands. Also make sure that all ISDN directory numbers associated with a SPID are associated with the same telephone port. For information on using the port command while setting up a dial peer, see the "Creating Dial Peers" section.
Forwarding Incoming ISDN Voice Calls to Connected Devices
Starting from global configuration mode, follow these steps:
Note
Configuring Advanced Telephone Features
This section describes advanced telephone features and how to configure them.
ISDN Voice Priority
The ISDN voice priority feature controls the priority of data and voice calls for telephones or fax machines connected to the router telephone ports. If an ISDN circuit endpoint is busy with a data call or calls and either a voice call comes in (incoming) or you attempt to place a voice call (outgoing), the data call is handled per the voice priority setting.
You can configure the router so that data calls are handled in one of the following ways:
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Use the following command to reconfigure the priority.
isdn voice-priority local-directory-number
{in | out} {always | conditional | off}Configure ISDN voice priority for each ISDN directory number.
If you have multiple ISDN directory numbers associated with a SPID, then the outgoing voice priority that you set for any of these directory numbers applies to the other numbers.
For example, if you enter the following command, the outgoing voice priority for all directory numbers specified in the isdn spid1 command is set to conditional:
router(config-if)# isdn spid1 0 4085551111 4085552222 4085553333router(config-if)# isdn voice-priority 5551111 out conditionalTable 6-1 describes the possible data call scenarios, what happens when a voice call comes in, and what happens when you place an outgoing voice call with a particular configuration.
The setting of the pots dialing-method command determines whether you hear a busy signal if a data call cannot be bumped when you are trying to make an outgoing call. If the setting is overlap, you hear a busy signal when you pick up the handset. If the setting is enblock, you hear a dial tone initially, then a busy signal.
Data over Voice Bearer Service
Note
In some tariff areas, voice calls are less expensive than data calls. If this is the case in your tariff area, the Cisco 800 series routers support incoming and outgoing data over voice (DOV) calls. DOV calls are data calls made over the ISDN line using voice bearer capability (VBC).
The router recognizes the difference between a data call and a voice call. Incoming data calls are routed to the LAN over the Ethernet port. If a telephone interface has been configured for DOV, incoming data calls made with VBC are routed to the LAN over the Ethernet port. Figure 6-1 and Table 6-2 illustrate a data call being routed to the LAN.
Incoming voice calls are forwarded to the analog device over the analog telephone port, as shown in Figure 6-2 and Table 6-3.
Figure 6-1 Data Call over VBC Line
Table 6-2 Key for Data Call over VBC Line
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Analog telephone
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ISDN BRI line with VBC
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Central office switch
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Ethernet LAN
Figure 6-2 Voice Call over VBC Line
Table 6-3 Key for Voice Call over VBC Line
1
Analog telephone
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ISDN BRI line with VBC
3
Router
4
Ethernet LAN
Note
Use the following command to configure the router to accept incoming DOV calls:
isdn incoming-voice data 56
Follow these steps to configure the router to place outgoing DOV calls:
Distinctive Ringing
A ringing cadence is a pattern of a ringing and a quiet period. There are two types of ringing cadences: a primary ringing cadence and distinct ringing. The primary cadence is determined by the country where your router is located. In addition to the primary cadence, you can configure up to two distinctive rings on a telephone port.
Because the router associates a distinctive ring with the ISDN directory number assigned to an interface, you must configure a distinctive ring with a dial peer. For information on dial peers and how to configure them, see the "Creating Dial Peers" section.
Note
To configure the ringing cadence, insert the following commands into a dial-peer configuration:
ring cadence-number
where cadence-number can be 0, 1, or 2.
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By default, the ring cadence is set to 0, which means that the interface uses the primary ringing cadence.
You can also insert the following command syntax into a dial-peer configuration:
pots distinctive-ring-guard-time milliseconds
where milliseconds can be a number from 50 to 1000. This command configures the delay, in milliseconds, before a telephone port can be rung after a previous call is disconnected. The default is no delay.
Caller Identification
In addition to an analog telephone or fax machine, North American users can connect a caller ID device to the router telephone ports. This device displays the telephone numbers of incoming callers. The Cisco 800 series routers support the following caller ID devices:
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The Cisco 800 series routers do not support the following devices:
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How to Use Telephones Connected to Cisco 800 Series Routers
This section describes how to make a basic call and how to use the supplementary services that you ordered from your telephone service provider.
Making a Basic Call
To make a basic telephone call, pick up the handset, and dial the number of the desired party.
To make a basic call if your router is connected to a Nippon Telegraph and Telephone (NTT) switch, follow these steps:
Step 1
You must enter each digit within 12 seconds of entering the previous digit. If you wait longer than 12 seconds, an incomplete set of digits is sent to the switch.
Step 2
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Disabling Pound Key End-of-Call Function
You can disable the end-of-call function (initiated by pressing the pound key [#]) by entering the following command on the telephone keypad:
**98#
Note
You can disable this function if a telephone number you are dialing requires the pound key (#) as one of the digits. After entering the **98# command, wait for a dial tone and then enter the digits, including the pound key. To send the digits to the switch, wait 6 seconds without entering any digits.
The end-of-call function automatically resumes for the next call.
Using Supplementary Services
This section describes how to use the following supplementary services:
Call Holding and Retrieving
For this feature to work, you must request it when you order your ISDN line. For information on ordering your ISDN line, see Appendix D, "Provisioning an ISDN Line." However, you do not need to configure the router to make this feature work.
You can put an active voice call on hold, make a second call, and toggle between the two calls. Follow these steps:
Step 1
Step 2
Step 3
If you hang up with a call still on hold, the phone rings to remind you of the outstanding call. Pick up the handset to reconnect to the call.
Call Waiting
For this feature to work, you must request it when you order your ISDN line. For information on ordering your ISDN line, see Appendix D, "Provisioning an ISDN Line."
By default, call waiting is enabled. You can disable it permanently by using the no call-waiting command. (You might want to disable it for fax machines.) Because the router associates call waiting with the ISDN directory number assigned to a telephone interface, you should disable call waiting at the same time that you are configuring a dial peer. For information on dial peers and how to configure them, refer to the "Creating Dial Peers" section.
To disable call waiting on a per-call basis, enter **99# on the telephone keypad.
During an active voice call, a call-waiting tone sounds if another call comes in. Subsequent tones sound at 10-second intervals until the incoming caller hangs up or until you answer the call. During this time, the incoming caller hears ringing.
When you hear the call-waiting tone, you can do one of the following:
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To put the current call on hold and answer the incoming call, quickly press the telephone receiver (flash) button once. Press this button again to go back to the current call.
Three-Way Conference Call
For this feature to work, you must request it when you order your ISDN line. For information on ordering your ISDN line, see Appendix D, "Provisioning an ISDN Line."
If you are connected to a National ISDN-1 (NI1) or a Northern Telecom DMS-100 custom switch, you might need to activate this feature by using the following command syntax:
isdn conference-code range
The range is from 0 to 999. The default code is 60. Your telephone service provider should provide a code when you order this feature; if a code other than 60 is provided, you need to reconfigure the code using the isdn conference-code command.
Otherwise, you do not need to configure the router to make this feature work.
You can talk simultaneously with two other parties. To create a conference call, follow these steps:
Step 1
Step 2
Step 3
Call Transfer
For this feature to work, you must request it when you order your ISDN line. For information on ordering your ISDN line, see Appendix D, "Provisioning an ISDN Line."
If you are connected to a National ISDN-1 (NI1) or a Northern Telecom DMS-100 Custom switch, you might need to activate this feature, using the following command syntax:
isdn transfer-code range
The range is from 0 to 999. The default code is 61. Your telephone service provider should provide a code when you order this feature; if a code other than 61 is provided, you need to reconfigure the code by using the isdn transfer-code command.
Otherwise, you do not need to configure the router to make this feature work.
You can transfer an incoming or outgoing voice call to another party. To transfer a call, do the following:
Note
Step 1
Step 2
Step 3
Hanging up connects the first and second parties. Instead of doing Step 3, you can also create a three-way call conference by quickly pressing the flash button once.
If the call to the second party fails, you can return to the first party by doing one of the following:
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If you hang up, the telephone rings to indicate that the first party is still on hold.
Call Forwarding
The call forwarding feature works for Sweden and Finland only. For this feature to work, you must request it when you order your ISDN line. For information on ordering your ISDN line, see Appendix D, "Provisioning an ISDN Line."
The router supports the following call forwarding features:
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To make sure that the router accepts the activation and deactivation of the call forwarding features using the telephone keypad, use the pots country country command in global configuration mode. The country variable is the country that your router is in. Enter the pots country ? command to get a list of supported countries and the code you must enter to indicate a particular country.
To activate call forwarding unconditional, call forwarding no reply, or call forwarding busy, follow these steps:
Step 1
Step 2
*feature-number *telephone-number-to-forward-to#Your telephone service provider should provide the number for each call forwarding feature. For example, to forward a call to 408-555-2222, enter the following:
*21*4085552222#Step 3
To deactivate call forwarding unconditional, call forwarding no reply, or call forwarding busy, follow these steps:
Step 1
Step 2
#feature-number#Your telephone service provider should provide the number for each call forwarding feature. For example, to deactivate call forwarding, enter the following:
#21#Step 3
Note
POTS Dial Feature (Japan Only)
The Cisco 813 router supports the plain old telephone service (POTS) dial feature for Japanese telephones. This feature can be activated by a dial application on your workstation that dials a telephone number for the POTS port on the Cisco 813 router. The telephone connected to the port can be on- or off-hook when the dial command is issued. If the telephone is on-hook, the router rings the telephone, waits until the telephone is taken off hook, then dials the number requested. If the telephone is off-hook when the command is issued, the router dials the number requested, provided that the telephone is receiving a dial tone.
Activating the POTS Dial Feature
Each time you wish to activate this feature on the router for use by the dial application, enter the following Cisco IOS command in EXEC mode:
test pots port dial number [#]where port is the port number 1 or 2, and number is the telephone number to dial.
Note
The following example shows the POTS dial command:
router# test pots 1 dial 4085551234#Displaying POTS Call State
To show the current state of POTS calls and the most recent event received by the call switching module (CSM), use the show pots csm command in EXEC mode.
show pots csm portwhere port is port number 1 or 2.
Output Example
The following is an example of the show pots csm command screen output:
router# show pots csm 1POTS PORT: 1CSM Finite State Machine:Call 0 - State: idle, Call Id: 0x0Active: noEvent: CSM_EVENT_NONE Cause: 0Call 1 - State: idle, Call Id: 0x0Active: noEvent: CSM_EVENT_NONE Cause: 0Call 2 - State: idle, Call Id: 0x0Active: noEvent: CSM_EVENT_NONE Cause: 0router#Disconnecting a POTS Call
To disconnect a telephone call for the POTS port on the router, use the test pots port disconnect command in EXEC mode:
test pots port disconnect
where port is the port number 1 or 2.
The following example disconnects a telephone call from POTS port 1:
router# test pots 1 disconnectrouter#POTS Debug Command
To display the status of calls made to and from the POTS ports, enter the following command in EXEC mode:
debug pots csmEntering this command activates events by which your dial application can determine the progress of calls to and from the ports.
Debug Message Formats
Debug messages are displayed in one of two formats that are relevant to the POTS dial feature:
hh:mm:ss: CSM_STATE: CSM_EVENT, call id = ??, port = ?or
hh:mm:ss: EVENT_FROM_ISDN:dchan_idb=0x???????, call_id=0x????, ces=? bchan=0x????????, event=0x?, cause=0x??where:
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CSM States
Table 6-4 shows the values for CSM states.
CSM Events
Table 6-5 shows the values for CSM events.
Events
Table 6-6 shows the values for events that are translated into CSM events.
Cause Values
Table 6-7 shows cause values that are assigned only to call-progressing events.
Call Scenarios for the POTS Dial Feature
This section describes three call scenarios and shows examples of the Cisco IOS command output for each scenario. The output examples for the debug and disconnect commands show the sequence of events that occur during a POTS dial call.
Call Scenario 1
In this call scenario, port 1 is on-hook, the application dial is set to call 4085552221, and the far end successfully connects. The following example shows the Cisco IOS command:
router# test pots 1 dial 4085552221#router#The following screen output shows an event indicating that port 1 is being used by the dial application:
01:0, port = 1The following screen output shows events indicating that the CSM is receiving the application digits of the number to dial:
01:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:58:27: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 1The following screen output shows that the telephone connected to port 1 is off hook:
01:58:39: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_OFFHOOK, call id = 0x0, port = 1The following screen output shows a call-proceeding event pair indicating that the router ISDN software has sent the dialed digits to the ISDN switch:
01:58:40: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8004, ces=0x1 bchan=0x0, event=0x3, cause=0x001:58:40: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_ISDN_PROC, call id = 0x8004, port = 1The following screen output shows the call-progressing event pair indicating that the telephone at the far end is ringing:
01:58:40: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8004, ces=0x1 bchan=0xFFFFFFFF, event=0x5, cause=0x001:58:40: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_ISDN_CALL_PROGRESSING, call id = 0x8004, port = 1The following screen output shows a call-connecting event pair indicating that the telephone at the far end has answered:
01:58:48: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8004, ces=0x1 bchan=0xFFFFFFFF, event=0x4, cause=0x001:58:48: CSM_PROC_CONNECTING: CSM_EVENT_ISDN_CONNECTED, call id = 0x8004, port = 1The following screen output shows a call-progressing event pair indicating that the telephone at the far end has hung up, and the calling telephone is receiving an in-band tone from the ISDN switch:
01:58:55: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8004, ces=0x1 01:58:55: CSM_PROC_CONNECTED: CSM_EVENT_ISDN_CALL_PROGRESSING, call id = 0x8004, port = 1The following screen output shows that the telephone connected to port 1 has hung up:
01:58:57: CSM_PROC_CONNECTED: CSM_EVENT_VDEV_ONHOOK, call id = 0x8004, port = 1The following screen output shows an event pair indicating that the call has been terminated:
01:58:57: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8004, ces=0x1 bchan=0xFFFFFFFF, event=0x0, cause=0x001:58:57: CSM_PROC_NEAR_END_DISCONNECT: CSM_EVENT_ISDN_DISCONNECTED, call id = 0x8004, port = 1813_local#Call Scenario 2
In this scenario, port 1 is on-hook, the application dial is set to call 4085552221, and the destination number is busy. The following example shows the Cisco IOS command:
router# test pots 1 dial 4085552221#router#The following screen output shows that your dial application is using port 1:
01:59:42: CSM_PROC_IDLE: CSM_EVENT_VDEV_APPLICATION_CALL, call id = 0x0, port = 1The following screen output shows the events indicating that the CSM is receiving the application digits of the number to call:
01:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 101:59:42: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 1The following screen output shows an event indicating that the telephone connected to port 1 is off-hook:
01:59:52: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_OFFHOOK, call id = 0x0, port = 1The following screen output shows a call-proceeding event pair indicating that the telephone at the far end is busy:
01:59:52: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8005, ces=0x1 bchan=0x0, event=0x3, cause=0x1101:59:52: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_ISDN_PROC, call id = 0x8005, port = 1The following screen output shows a call-progressing event pair indicating that the calling telephone is receiving an in-band busy tone from the ISDN switch:
01:59:58: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8005, ces=0x1 bchan=0xFFFFFFFF, event=0x5, cause=0x001:59:58: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_ISDN_CALL_PROGRESSING, call id = 0x8005, port = 1The following screen output shows an event indicating that the calling telephone has hung up:
02:00:05: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_VDEV_ONHOOK, call id = 0x8005, port = 1The following screen output shows an event pair indicating that the call has terminated:
02:00:05: EVENT_FROM_ISDN:dchan_idb=0x280AF38, call_id=0x8005, ces=0x1 bchan=0xFFFFFFFF, event=0x0, cause=0x002:00:05: CSM_PROC_NEAR_END_DISCONNECT: CSM_EVENT_ISDN_DISCONNECTED, call id = 0x8005, port = 1Call Scenario 3
In this call scenario, port 1 is on-hook, the application dial is set to call 4086661112, the far end successfully connects, and the command test pots disconnect terminates the call.
router# debug pots csmrouter# test pots 1 dial 4086661112router#The following screen output follows the same sequence of events as shown in Call Scenario 1:
1d03h: CSM_PROC_IDLE: CSM_EVENT_VDEV_APPLICATION_CALL, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_DIGIT, call id = 0x0, port = 11d03h: CSM_PROC_APPLIC_DIALING: CSM_EVENT_VDEV_OFFHOOK, call id = 0x0, port = 11d03h: EVENT_FROM_ISDN:dchan_idb=0x2821F38, call_id=0x8039, ces=0x1bchan=0x0, event=0x3, cause=0x01d03h: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_ISDN_PROC, call id = 0x8039, port = 11d03h: EVENT_FROM_ISDN:dchan_idb=0x2821F38, call_id=0x8039, ces=0x1bchan=0xFFFFFFFF, event=0x5, cause=0x01d03h: CSM_PROC_ENBLOC_DIALING: CSM_EVENT_ISDN_CALL_PROGRESSING, call id = 0x8039, port = 1router# test pots 1 disconnectThe test pots disconnect command disconnects the call before you have to put the telephone back on hook.
1d03h: CSM_PROC_CONNECTING: CSM_EVENT_VDEV_APPLICATION_HANGUP_CALL, call id = 0x8039, port = 11d03h: EVENT_FROM_ISDN:dchan_idb=0x2821F38, call_id=0x8039, ces=0x1bchan=0xFFFFFFFF, event=0x0, cause=0x01d03h: CSM_PROC_DISCONNECTING: CSM_EVENT_ISDN_DISCONNECTED, call id = 0x8039, port = 11d03h: CSM_PROC_DISCONNECTING: CSM_EVENT_TIMEOUT, call id = 0x8039, port = 1Cisco 813 Router Enhanced Voice Features
(Japan Only)The Cisco 813 router supports the enhanced voice features in addition to the standard voice features of the Ciso 800 series routers. The enhanced voice features were developed to work with the INS-NET-64 switch used by Nippon Telephone and Telegraph (NTT).
Support for these features is limited to Japanese telephones, with the exception of the call blocking on caller ID feature. For more information about each feature, see the following topics:
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General Requirements and Restrictions
The following is a list of requirements for activating the enhanced voice features on the Cisco 813 router:
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Caller ID Display
This feature displays the caller ID information provided by the INS-NET-64 switch on analog telephones connected to the PHONE 1 or 2 port of the Cisco 813 router.
Requirements for Activating Caller ID Display
The following is a list of requirements for activating this feature:
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Note
Configuring Caller ID Display
By default, this feature is disabled. To configure this feature, use the Cisco IOS caller-id enable command in the dial-peer configuration command mode.
caller-id enableno caller-id enableCall Blocking on Caller ID
This feature can reject an incoming voice call based on the caller ID information presented to the Cisco 813 router from the INS-NET-64 switch. This feature can block calls for up to ten caller IDs for each local directory number (LDN).
Requirements for Activating Call Blocking on Caller ID
The following requirements must be met before activating this feature:
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Configuring Call Blocking on Caller ID
By default, this feature is disabled. To configure this feature, use the Cisco IOS block-caller command in the dial-peer configuration command mode.
block-caller numberno block-callerwhere number is the telephone number to block. You can use a period (.) as a wildcard to substitute for one or more numbers to block. For example, to block all numbers ending in the number 5, you can enter the following:
block-caller .5You can enter up to ten caller ID numbers for each LDN. However, you cannot exceed the maximum of ten numbers. You must remove one or more numbers before you can add any new numbers to block.
If no caller ID numbers are specified for a particular LDN, all voice calls to that LDN are accepted.
Example of Caller ID Blocking Configuration
The following example configures the router to block calls from the caller whose caller ID number is 4085551234:
router(config)# pots country jprouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# block-caller 4085551234Example of Caller ID Blocking Output
To display caller IDs entered for call blocking, use the show run command. The following is an example of caller ID configuration output:
!dial-peer voice 1 potsno forward-to-unused-portcall waitingring 0registered-caller ring 1port 1block-caller 4085551234block-caller 4085552345Local Call Waiting
This feature notifies you of an incoming call while you are connected to an external telephone call (by issuing a call waiting tone). You can choose to place the first call on hold by pressing flash, connect to the second call, then return to the first call after finishing with the second.
Local call waiting on the Cisco 813 router differs from standard ISDN call waiting in that this enhanced voice feature does not require a subscription to call waiting from the service provider. This feature uses both B channels of the ISDN line, enabling local call waiting support on the router rather than from the service provider.
This feature is not supported if any of the interactive voice response (IVR) features (such as voice warp, voice select warp, and Nariwake) are in use.
Requirements for Activating Local Call Waiting
The following requirements must be met before activating this feature:
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Configuring Local Call Waiting
To configure this feature, use the Cisco IOS pots call-waiting command:
pots call-waiting [local|remote]The call waiting defaults to remote if this feature is not configured. In that case, the call holding pattern follows the settings of the service provider rather than those of the router.
To display the call waiting setting, use the show run or show pots status command.
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Example of Local Call Waiting Configuration
The following example configures the call waiting style to follow the local call holding pattern that is set on the router:
router(config)# pots country jprouter(config)# pots call-waiting localE Ya Yo
This feature conceals the caller ID of the outgoing call from the receiving device.
Requirements for Activating E Ya Yo
The following requirements must be met before activating this feature:
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Configuring E Ya Yo
According to the NTT specification, dialing the prefix 184 followed by the destination device number will render your caller ID invisible to the receiving party.
Voice Warp
The voice warp feature on the INS-NET-64 switch forwards all incoming calls for a terminal device to another device. Voice warp registration, activation, and deactivation requests are sent to the switch for each LDN. The Cisco 813 router supports the registration, activation, and deactivation requests for any device attached to the PHONE 1 or 2 port. The forwarding function itself is performed by the INS-NET-64 switch.
During the registration phase of the device, you can:
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During the activation phase of this feature, you determine whether calls are redirected all the time or only if the receiving device is busy or does not answer within the no-answer time period specified during registration.
This feature can be deactivated after its registration and activation phases.
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Requirements for Activating Voice Warp
The following requirements must be met before activating this feature:
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Configuring Voice Warp
This feature is configured using the interface on the telephone as specified in the NTT user manual.
To hear the voice warp registration details of a device, use the keypad dialing sequence specified in the NTT user manual. Information is transmitted only by voice.
Voice Select Warp
This feature is an enhanced version of the voice warp feature. You can create a list of incoming caller IDs that is used in call redirection, either by redirecting incoming calls only from matching caller IDs, or by redirecting all calls except those from matching caller IDs.
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Requirements for Activating Voice Select Warp
The following requirements must be met before activating this feature:
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Configuring Voice Select Warp
This feature is configured using the interface on the telephone as specified in the NTT user manual.
To get voice warp registration details of a device, use the keypad dialing sequence specified in the NTT user manual. Information is transmitted only by voice.
Nariwake
Nariwake checks for caller IDs that you register for each LDN and presents a distinctive ring to the telephone port receiving the incoming call if a match is detected. The Cisco 813 router provides three different ring cadences that you can set for calls from registered and unregistered callers. The number of caller IDs you can register for each LDN at one time is defined by the INS-NET-64 switch and not by the router.
You can register this feature with the list of caller IDs for each LDN, cancel the registration for the LDN, or get registration information from the INS-NET-64 switch.
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Requirements for Activating Nariwake
The following requirements must be met before activating this feature:
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Configuring Nariwake
To configure the ring cadence for this feature, use the registered-caller ring command in the dial-peer configuration mode:
registered-caller ring cadencewhere cadence is a value of 0, 1, or 2. The default ring cadence for registered callers is 1 and for unregistered callers is 0.
The on/off periods of ring 0 (normal ringing signals) and ring 1 (ringing signals for the Nariwake service) are defined in the NTT user manual.
Note
If more than one dial-peer is configured with destination-pattern not-provided, the router uses only the first dial-peer for the incoming calls.To hear the caller ID registration details, use the keypad dialing sequence specified in the NTT user manual. Information is transmitted only by voice.
Example of Nariwake Configuration
The following example sets the ring cadence for registered callers to 2.
router(config)# pots country jprouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# registered-caller ring 2Add the destination-pattern not-provided command if you also subscribe to the I Number and dial-in services.
router(config-dial-peer)# destination-pattern not-providedExample of Nariwake Configuration Output
To display the Nariwake ring cadence setting, use the show run command. The following is an example of screen output for Nariwake configuration:
dial-peer voice 1 potsno forward-to-unused-portcall waitingring 0registered-caller ring 2port 1destination-pattern not-providedblock-caller 4085552222block-caller 4085553333Trouble-Call Blocking
The trouble-call blocking feature causes all future incoming calls from a particular telephone number to be rejected by the network if the recipient activates this feature after the initial call. As the recipient of the call, you are not required to specify the telephone number of the caller and will not be notified of subsequent connection attempts from that telephone number. When this feature is activated, the caller will hear a standard telephone announcement and a disconnect message. For information about the announcement or message, see your NTT user manual.
The number of callers that you can block is defined by the service provider at the time the service is provisioned. If you request an additional telephone number to block after having reached the limit, the oldest number is discarded (unblocked) before the latest telephone number is registered for blocking.
Requirements for Activating Trouble-Call Blocking
The following requirements must be met before activating this feature:
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Configuring Trouble-Call Blocking
You can activate, cancel, or request confirmation of the results of your trouble-call blocking by using the keypad dialing sequence specified in the NTT user manual.
Note
You can disable this feature for only the last registered number or for all numbers registered for blocking. You will be notified over the telephone whether or not the cancellation is successful.
You can request to hear the results of the trouble-call blocking. You will hear the number of attempted calls that were blocked for the past two months.
I Number
This feature supports the use of multiple terminal devices with one subscriber line. The telephone numbers of the subscriber line and router ports are assigned by the service provider. Calls coming into any of the assigned numbers will route through the same subscriber line to the terminal device attached to the target port.
Requirements for Activating I Number
The following requirements must be met before activating this feature:
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Configuring I Number
To configure this feature, perform the following steps:
Step 1
isdn i-number number ldnwhere number is a value from 1 to 3 (based on NTT specifications) and ldn is your local directory number configured under the dial-peer. The number variable maps the I number to one of the LDNs.
Step 2
destination-pattern ldn
Example of I Number Configuration
The following example shows screen output for two LDNs configured under interface BRI0:
router(config)# interface bri0router(config-if)# isdn i-number 1 5551234router(config-if)# isdn i-number 2 5556789router(config-if)# exitrouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# destination-pattern 5551234router(config-dial-peer)# exitrouter(config)# dial-peer voice 2 potsrouter(config-dial-peer)# destination-pattern 5556789Silent Fax Calls
The silent fax calls feature enables you to configure your router port to send a silent fax tone instead of a ring alert, which is recognized by fax machines with silent fax recognition capability (Smart Fax type 2). With the silent fax feature, the fax machine does not ring but the fax call get connected. If a phone is connected instead of a fax machine, the phone will not ring.
Configuring Silent Fax Calls
To configure your telephone port as a silent fax type 2, use the Cisco IOS silent-fax command in dial-peer configuration mode:
silent-faxno silent-faxBy default, this feature is disabled.
Example of Silent Fax Calls Configuration
The following is an example of a silent fax call configuration:
router# configure terminalrouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# silent-faxExample of Silent Fax Calls Configuration Output
The following is an example of the silent fax configuration output:
dial-peer voice 1 potscaller-idno forward-to-unused-portcall-waitingring 0no silent-faxregistered-caller ring 1port 1volume 3destination-pattern 7773000!dial-peer voice 2 potscaller-idno forward-to-unused-portcall-waitingring 0no silent-faxregistered-caller ring 1port 2volume 3destination-pattern 7773100!Supplementary Telephone Services for the Net3 Switch
The Cisco 800 series routers now support the following plain old telephone service (POTS) features for the European Telecommunications Standards Institute (ETSI) Net3 switch type:
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Requirements for Supplementary Telephone Services Support
You must subscribe to the following Net3 switch services for these supplementary telephone services to work:
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Configuring Caller ID for the Net3 Switch
To enable caller ID on the Net3 switch, configure the country type by using the Cisco IOS pots country command in global configuration mode:
pots country {dk|fi}
Note
To verify whether caller ID is enabled, use the show pots status command. The following is an example of the output from that command:
router# show pots statusPOTS Global Configuration:Country:DenmarkDialing Method:Overlap, Tone Source:Local, CallerId Support:YES----------------------Out Going Hunt:DisabledCall Forwarding for the Net3 Switch
The following types of call forwarding services (for voice calls only) are supported on the Net3 switch:
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You can select one or more call forwarding services at a time. However, CFU has the highest precedence, CFB the next highest, and CFNR the lowest. The default setting is that no forwarding type is selected.
Note
To enable and configure this feature, follow these steps:
Step 1
Step 2
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Configuring the Call Forwarding Method
You can select the method by which the call forwarding feature is controlled:
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To enable the call forwarding method, use the Cisco IOS pots forwarding-method command in global configuration mode:
pots forwarding-method {functional | keypad}no pots forwarding-method
Note
The following example configures the call forwarding feature to give control to the router:
router# configure terminalrouter(config)# pots forwarding-method functionalConfiguring the Call Forwarding Service
Table 6-8 shows the DTMF keypad command sequence that you enter to configure the call forwarding service.
You should hear a dial tone after you enter the DTMF commands if the call forwarding service is successfully configured. If you hear a busy signal, the command is invalid or the switch does not support that service.
Displaying POTS Status
Use the show pots status command to display details of the call forwarding type. This status is not stored when you reboot. The following is an example of the screen output:
router# show pots statusPOTS Global Configuration:Country:DenmarkDialing Method:Overlap, Tone Source:Local, CallerId Support:YESOut Going Hunt:DisabledForwarding Method:functional method-------------------------------------Call Forwarding status:The Forwarding Method Enabled is CFUThe forwarded to Address is :33236877The served user Number(s) are :33795742The Forwarding Method Enabled is CFBThe forwarded to Address is :33236877The served user Number(s) are :ALL -> Will work for all numbers allocated to the terminal.Configuring CLIR
Configure CLIR by following these steps:
Step 1
Step 2
Step 3
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Debug POTS Commands
Use the following commands to debug problems with caller ID configuration:
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Use the following commands for problems configuring other supplementary telephone features:
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For more information about using debug commands, refer to the Cisco IOS documentation.
Cisco 804 and 813 Routers Enhanced Voice Features
The Cisco 804 and 813 routers support the following enhanced voice features. For information on each feature, see the following topic:
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Prefix Dialing
Cisco 803 and Cisco 804 routers support prefix dialing. You can add a telephone prefix and create a prefix filter to the dialed number for analog telephone calls. When a telephone number is dialed through the telephone port, the router checks for prefix filters. If the router finds a match, no prefix is added to the dialed number. If no filter match is found, the router adds the user-defined prefix to the called number.
Configuring a Prefix Number
To set a prefix to be added to a telephone number called, use the Cisco IOS pots prefix number command in global configuration mode:
pots prefix number numberno pots prefix numberwhere number is a prefix number from 1 to 5 digits in length. Only one prefix can be configured at a time, and configuring a new number will overwrite the existing one.
The following example sets the prefix number to 12345:
router# configure terminalrouter(config)# pots prefix number 12345Configuring a Prefix Filter
You can configure a prefix filter that is compared to the digits that you dial. If a match occurs, the prefix number is not added to the called number. To create a prefix filter, use the pots prefix filter command in global configuration mode:
pots prefix filter numberno pots prefix filter numberwhere number is a prefix filter from 1 to 8 digits in length. You can define up to ten filters for your router. If you have reached the maximum number of filters defined, no new filter configurations are accepted until you remove at least one existing filter number using the no pots prefix filter number command.
The following are examples of how to set prefix filters:
router# configure terminalrouter(config)# pots prefix filter 192router(config)# pots prefix filter 1router(config)# pots prefix filter 9router(config)# pots prefix filter 0800router(config)# pots prefix filter 08456Calling Between Telephone Ports
The calling between telephone ports voice feature enables a connection between the two telephone ports of your router. This voice call is handled by the router and does not affect any data calls handled on the B channels. However, the following restrictions apply:
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Activating the Calling Between Telephone Ports Feature
To make a call between telephone ports, press **0# on your telephone handset.
Calling Between Telephone Ports Scenarios
Table 6-9 shows scenarios for calling between telephone ports.
Redial
This feature enables you to redial the last number called on either telephone
port 1 or 2. The following conditions apply:•
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Activating the Redial Feature
To redial the last number called, press **4# on your telephone handset.
Redial Feature Scenarios
Table 6-10 shows scenarios for the redial feature.
Call Transfer
The call transfer feature enables you to transfer an external call from one telephone port to the other. Call transfer does not require any subscription from the switch.
Activating the Call Transfer Feature
To transfer an incoming voice call from one port to another, press the flash hook switch, then **0# on the telephone handset.
Call Transfer Feature Scenarios
Table 6-11 shows scenarios for call transfer.
Volume Adjustments
The volume adjustment features enables you to adjust the receiver volume of the POTS ports.
To configure the telephone receiver volume on each port, use the Cisco IOS volume command in the dial-peer configuration mode:
volume numberwhere number is a numeric value from 1 to 5 representing the volume setting
ranging from -12 to 0 decibels (dB). The default setting is 3.
Table 6-12 lists the values and definitions of the number variable.
Table 6-12 Volume Adjustment Number Variable Definitions
1
-12
2
-9
3
-6
4
-3
5
0
Volume Adjustment Configuration Example
The following example configures the volume of the receiver on the router telephone ports 1 and 2:
router# configure terminalrouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# volume 4router(config-dial-peer)# dial-peer voice 2 potsrouter(config-dial-peer)# volume 2Volume Adjustment Configuration Output Example
The following is an example of the volume adjustment configuration output from the show running-config command:
dial-peer voice 1 potsdestination-pattern 5551111port 1no call-waitingring 0volume 4dial-peer voice 2 potsdestination-pattern 5552222port 2no call-waitingring 0volume 2Distinctive Ringing Based on Caller ID
The distinctive ringing feature enables you to configure the ring cadence for incoming calls based on the caller ID. You can choose from three ring cadences to associate with each telephone number and store up to twenty numbers per dial peer. You can configure a total of six dial peers but only one dial peer per port can be active at one time.
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Configuring Distinctive Ringing Based on Caller ID
To enable and configure distinctive ringing based on caller ID, use the following Cisco IOS command in dial-peer configuration mode:
caller number ring cadence
no caller number ring cadence
where number is the caller ID number of the incoming call, and cadence is the setting for ring cadence and duration. By default, this feature is disabled.
If you have configured the maximum number of twenty per dial peer, disable the numbers by using the no caller number ring cadence command.
Table 6-13 shows the available ring cadence settings.
Distinctive Ringing Scenarios
Table 6-14 shows scenarios for distinctive ringing.
Distinctive Ringing Configuration Example
The following is an example of the distinctive ringing configuration:
cisco801# configure terminalEnter configuration commands, one per line. End with CNTL/Z.cisco801#(config)#dial-peer voice 1 potscisco801#(config-dial-peer)#caller-number 11111 ring 1cisco801#(config-dial-peer)#caller-number 22222 ring 2cisco801#(config-dial-peer)#caller-number 33333 ring 1Distinctive Ringing Configuration Output Example
The following is an example of the output for the distinctive ringing feature from the show running-config command:
!
dial-peer voice 1 potsno caller-idno forward-to-unused-portcall-waitingring 0no silent-faxregistered-caller ring 1port 1volume 3caller-number 11111 ring 1caller-number 22222 ring 2caller-number 33333 ring 1!dial-peer voice 2 potsno caller-idno forward-to-unused-portcall-waitingring 0no silent-faxregistered-caller ring 1port 1volume 3caller-number 11111 ring 1caller-number 33333 ring 1caller-number 22222 ring 2Subaddresses for POTS Ports
The subaddressing feature enables you to assign an ISDN subaddress to each POTS port so that an external call can be directly connected to the number dialed.
Configuring Subaddresses for POTS Ports
To configure the subaddress for a POTS port, use the Cisco IOS subaddress command in dial-peer configuration mode:
subaddress numberno subaddress numberwhere number is the subaddress of a POTS port. Only one subaddress can be configured for each port. By default, no subaddresses are configured.
Subaddressing Scenarios
Table 6-15 shows scenarios for subaddresses for a POTS port.
Subaddressing Configuration Example
The following is an example of the subaddresses configuration:
router# configure terminalrouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# destination-pattern 5551111router(config)# dial-peer voice 2 potsrouter(config-dial-peer)# destination-pattern 5552222router(config-dial-peer)# subaddress 10Subaddressing Configuration Output Example
The following is an example of the output for configuring subaddresses of the POTS ports:
dial-peer voice 1 potsdestination-pattern 5551111port 1no call-waitingring 0volume 4caller 1112222 ring 3caller 2223333 ring 1caller 3334444 ring 1subaddress 20dial-peer voice 2 potsdestination-pattern 5552222port 2no call-waitingring 0volume 2caller 1111111 ring 1caller 2223323 ring 2caller 3213213 ring 3caller 8552345 ring 1caller 2223456 ring 2caller 3214567 ring 2subaddress 10Caller ID on the Cisco 813 Router
The correct information is as follows:
By default, the caller ID feature is disabled. To enable this feature, use the Cisco IOS caller-id command in the dial-peer configuration command mode.
caller-idno caller-idDebug POTS Commands
Use the following commands to debug problems with caller ID configuration:
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Use the following commands for problems configuring other supplementary telephone features:
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For more information about using debug commands, refer to the Cisco IOS documentation.
Local Call Forwarding
The local call forwarding feature enables you to forward an incoming voice call to an external telephone number if that call is not answered within a certain number of ring cycles. Highlights of this feature are as follows:
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Configuring Local Call Forwarding
To configure local call forwarding on your router, use the following Cisco IOS command in dial-peer configuration mode:
forward number after number of ringsno forward number after number of ringswhere number is the external telephone number to forward an incoming voice call, and number of rings is the maximum number of ring cycles (from 0 to 7) before the router forwards the call. By default, this feature is disabled.
Local Call Forwarding Scenarios
Table 6-16 shows scenarios for local call forwarding:
Local Call Forwarding Configuration Example
The following is an example of configuring the local call forwarding feature:
router# configure terminalrouter(config)# dial-peer voice 1 potsrouter(config-dial-peer)# forward 8765432 after 0router(config)# dial-peer voice 2 potsrouter(config-dial-peer)# forward 1234567 after 3Local Call Forwarding Configuration Output Example
The following is an example of the output for local call forwarding configuration:
dial-peer voice 1 potsdestination-pattern 5551111port 1no call-waitingring 0volume 4caller 1112222 ring 3caller 2223333 ring 1caller 3334444 ring 1subaddress 20forward 8765432 after 0dial-peer voice 2 potsdestination-pattern 5552222port 2no call-waitingring 0volume 2caller 1111111 ring 1caller 2223323 ring 2caller 3213213 ring 3caller 8552345ring 1caller 2223456 ring 2caller 3214567 ring 2subaddress 10forward 1234567 after 3Support for PIAFS
Personal Handy-Phone System (PHS) Internet Access Forum Standard (PIAFS) is a standard error-correction protocol for cellular data communication that is designed to pass data over the Personal Handy-Phone System (PHS) of cellular system. It also provides transmission control procedures (comparable to OSI reference model layer 2) for high-quality data transmission. Both PIAFS version 2.0 and version 2.1 are supported on the Cisco 800 series routers.
The following common applications are supported using PIAFS in PHS data communications:
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This enables the user to send and receive e-mail messages. E-mail is a basic service of the PHS multimedia communications menu.
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This enable faxing of data stored in a Personal Digital Assistant (PDA).
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Internet access has influenced PHS in that many users want to be able to obtain necessary information in a timely manner when they are outdoors. It is also projected that PHS will be used extensively to form intranets for in-house communications by facilitating the expansion of office LAN access points.
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This service can be realized by transmitting the signals of a digital still camera directly or through the medium of a personal computer. This can be regarded as another variation of data transmission service that can use the PHS for transmission.
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The spread of groupware recently has led to frequent instances where groups share common data bases in carrying out or supporting the execution of collaborative work. There are demands to extend this collaborative environment even to outside locations through the use of mobile communications. This is made possible by the use of PHS data communications.
The Cisco 800 series routers will accept incoming PIAFS calls from a peer supporting PIAFS 2.2 and will behave as speed variable type 2 devices. The Cisco 800 series routers will not request speed change but will respond to the speed change requests from the peer. See Table 6-17 below.
The table indicates that the Cisco 800 series routers will act only as a PIAFS speed variable type 2 device for all the peers supporting PIAFS 2.2.
Configuring PIAFS
This feature is available by default in all images. It is enabled when the ISDN switch type is set to INS (NTT) and PPP encapsulation is configured on the ISDN interface.
PIAFS Scenarios
Table 6-18 shows scenarios for PIAFS. The feature is activated when the ISDN switch type is set to INS(NTT) and PPP encapsulation is configured on the ISDN interface.
PIAFS Status
The status of the PIAFS calls on the router can be checked by using the following command in privileged mode:
show piafs statusPIAFS Configuration Output Example
The following is an example of the output for PIAFS configuration:
Number of active calls = 1Details of connection 0Call Direction is - INCOMINGThe speed is - 32KThe Bchan assigned for this call is - B1 CHANV42 Negotiated - YESV42 ParametersDirection - BOTHNo of code words - 4096Max string length - 250First PPP Frame Detected - YESPiafs main FSM state - PIAFS_DATA
