12-3 Dialing


  • Dial plans are used to describe the digits that must be dialed to reach a certain destination.

  • Dial peers are configured to implement the dial plan. A local dial peer is used to route and terminate a call to a remote dial peer.

  • The path that the call takes through the network is determined by the underlying network configuration, involving IP routing protocols and physical connectivity, including Frame Relay and ATM.

  • An end-to-end call is broken into call legs logical connections between any two telephony devices. Each call leg must have a dial peer associated with it.

  • POTS dial peer is a logical definition on a router that connects to a traditional telephony network on a voice port (PSTN, PBX, telephone, or fax).

  • Voice-network dial peer is a logical definition on a router that points to a remote dial peer. These dial peers include Voice over IP (VoIP) using an IP address, Voice over Frame Relay (VoFR) using a DLCI, Voice over ATM (VoATM) using an ATM VC, and Multimedia Mail over IP (MMoIP) using the e-mail address of an SMTP server.

  • Dial peers are associated with a dial pattern. Incoming calls must match the dial pattern of a dial peer. For outgoing calls, a router collects digits from the caller and sends them to the associated voice port or remote dial peer.

  • Digit manipulation can be used on a string of dialed digits to strip off digits, add prefix digits, translate digits, and expand digits into other digit strings.

  • Dial numbers can be communicated through telephony signaling mechanisms:

    • Automatic Number Identification (ANI) identifies the calling party's telephone number.

    • Dial Number Information Service (DNIS) identifies the called party's telephone number.

Configuration

  1. Map out your telephony network.

    1. Use a diagram to show all voice-capable routers, telephony devices, and connections to the PSTN.

    2. Divide the diagram into call legs: Identify every router voice port and logical connections to voice-network dial peers (VoIP, VoFR, and VoATM).

    3. For each voice-capable router, make a table identifying the dial peers, dial patterns, and call leg destinations. Table 12-4 shows a sample table for this purpose.

      The Dial Peer Number uniquely identifies each dial peer that is defined on a router. The Local Type is either POTS (using a voice port) or VoIP (using a remote session target). Destination Pattern represents a shorthand notation of the digits required to reach the destination. This can be an entire ten-digit phone number to reach a single specific device, or it can include wildcard characters so that an entire remote site of multiple users can be reached.

      Table 12-5 shows the destination pattern wildcard characters that can be used.

      The Local Extension column in Table 12-4 indicates a reduced set of digits that can be used to reach the destination. Normally, when a router matches a destination pattern string, the matching digits are discarded and the destination is sent only the remaining digits. The Prefix to Add column lists any digits that might need to be prefixed to the remaining digits to complete the call. Call Destination lists the endpoint where the router attempts to terminate the call. This can be either a voice port (a local voice port interface) or a session target (the IP address of a remote VoIP peer, the DLCI of a remote VoFR peer, or the VC of a remote VoATM peer).

Table 12-4. Identifying the Dial Peers, Dial Patterns, and Call Leg Destination
Dial Peer Number Local Type Destination Pattern Local Extension Prefix to Add Call Destination
Voice Port SessionTarget
             
             
             
Table 12-5. Destination Pattern Wildcards
Wildcard Description
. Matches a single digit.
[ ] Matches a range of digits, as in [19], or several ranges, as in [15,89].
() Groups digits into a pattern that can be matched with ?, %, or +.
? Matches the preceding digit zero or one time.
% Matches the preceding digit zero or more times.
+ Matches the preceding digit one or more times.
T Indicates a variable-length string of digits. The string is terminated when the interdigit timeout expires or the caller enters the # ( pound ) digit.
  1. (Optional) Use number expansion:

     (global)  num-exp   extension-number expanded-number  

    The dial string extension-number is converted into the string expanded-number before being used with a dial peer. These strings consist of the digits 0 to 9 and can use a period (.) as a wildcard digit. The digits matched with periods in the extension-number are carried over into the corresponding period in the expanded-number. In this fashion, dial strings can be stripped of digits, keeping a shortened extension number, or can be expanded into a longer number.

  2. Configure dial peers.

    1. Identify the dial peer:

       (global)  dial-peer voice   number  {  pots   voip   vofr   voatm  } 

      The dial peer is given a unique number (1 to 2,147,483,647). Specify the peer type as pots (Plain Old Telephone System; analog or digital voice port), voip (Voice over IP), vofr (Voice over Frame Relay), or voatm (Voice over ATM).

    2. Specify the destination pattern to reach the dial peer:

       (dial-peer)  destination-pattern  [+]  string  [T] 

      The string is the pattern of digits or wildcard characters required to reach the destination. Digits can be 0 through 9, the letters A through D, the star (*) or pound (#) keys, or a comma (,) to indicate a pause between digits. The + can be used to indicate that a standard E.164 address follows . The optional T can be used to indicate that the string is of variable length. The string is terminated when the interdigit timeout expires or when the caller presses the # key.

    3. Specify the call destination.

      • POTS destination:

         (dial-peer)  port   location  
      • The location field indicates the physical location of the voice port on the router. This can be slot / subunit / port for an analog voice port, slot / port: ds0- group -no for a channelized T1/E1, or slot: D for the D-channel of a PRI (also slot: 23 for a T1 PRI and slot: 15 for an E1 PRI).

      • VoIP destination:

         (dial-peer)  session target  {  ipv4:   destination-address   dns:  [$s$.  $d$.   $e$.   $u$.  ]  host-name   ras   settlement  } 
      • The VoIP session terminates at IP address destination-address. The target can also be specified using a DNS host name entry, host-name. The host name can also contain wildcard translations, to be replaced by these strings: $s$. (the source number), $d$. (the destination number), $e$. (the digits of the called number are reversed and separated by dots), and $u$. (the unmatched portion of the destination pattern). There should be no spaces between dns:, the wildcard pattern, and the host-name strings.

      • VoFR destination:

         (dial-peer)  session target   interface dlci  [  cid  ] 
      • The session terminates on the Frame Relay interface, using DLCI number dlci. The optional cid field (4 to 255) identifies the DLCI subchannel to use for calls over an FRF.11 trunk. You should use values between 6 and 63 for voice traffic.

      • VoATM destination:

         (dial-peer)  session target   interface   pvc  {  name   vpi/vci   vci  } 
      • The session terminates over the ATM PVC on interface. The PVC is specified by name, by the VPI/VCI pair vpi/vci, or by the vci alone.

    4. (Optional) Select a Codec type for the dial peer.

      • Use a specific Codec type:

         (dial-peer)  codec   type  [  bytes   payload_size  ] 

        The Codec type can be set to a value from the Codec Type column of Table 12-6. You can specify the number of payload bytes in each voice frame with the bytes keyword and payload_size to override the defaults shown in the table. The table also lists the range of payload sizes for each Codec type, as a range, followed by the multiple that the size must be, and the default size .

        Generally, the G.711 Codecs offer the best voice quality at the cost of higher bandwidth. The G.729 Codecs offer a good voice quality at a low bandwidth, usually deployed over serial WAN links.

      • (Optional) Create a list of Codecs in order of preference.

        Create a voice class for the list:

         (global)  voice class codec   tag  

        The voice class is given an arbitrary number tag (1 to 10000).

        Assign a preference to one or more Codecs that might be used:

         (voice-class)  codec preference   priority codec-type  [  bytes   size  ] 

        The codec-type is a name from the first column of Table 12-6, assigned with a priority (1 to 14; lower is preferable). The payload size can be specified as size in case the same Codec type is defined with more than one possible payload size.

        Apply the Codec preference list to a dial peer:

         (dial-peer)  voice-class codec   tag  
    5. Configure parameters for all dial peer types.

      • (Optional) Use Voice Activity Detection (VAD) for reduced bandwidth during silence:

         (dial-peer) [  no  ]  vad  

        With VAD (enabled by default), no voice data is transmitted during a silent audio period. An optional comfort noise can be generated instead so that the caller won't think the call has been disconnected during the silent period. VAD can be disabled with the no keyword so that voice data is sent continuously.

      • (Optional) Limit the number of connections to the dial peer:

         (dial-peer)  max-conn   number  

        The maximum number of allowed connections to the dial peer is number (1 to 2147483647; the default is unlimited).

      • (Optional) Match the calling number for inbound calls:

         (dial-peer)  answer-address  [+]  string  [T] 

        Inbound calls can be matched against the calling number (ANI) if desired. The router matches against any configured answer addresses before destination patterns are matched.

      • (POTS only) Use DID to find a destination for inbound calls:

         (dial-peer)  direct-inward-dial  

        The called number (DNIS) can be used on inbound calls to identify the call destination. In this case, the router accepts a DID number and matches against destination patterns to complete the call.

      • (Optional) Identify the called number in a mixed modem/voice router:

         (dial-peer)  incoming called-number   string  

        When the router is handling both modem and voice calls over an interface, the voice port must be identified with the called number.

      • (Optional) Match against an ITU Q.931 numbering type:

         (dial-peer)  numbering-type   type  

        The numbering type can be matched against, in addition to other dial number matching mechanisms. The Q.931 numbering type can be abbreviated (the abbreviated number as supported locally), international (the number to reach a subscriber in another country), national (the number to reach a subscriber in the same country, outside the local network), network (the administrative or service number), subscriber (the number to reach a subscriber in the local network), unknown (unknown to the local network), or reserved (reserved for extension).

      • (Optional) Use digit manipulation to change a dial string.

        (Optional) Add a prefix to an outbound dial peer:

         (dial-peer)  prefix   string  

        A string of digits can be prefixed to the telephone number associated with an outbound call to a dial peer. The string can contain digits 0 to 9 and a comma (,) to indicate a pause.

        (Optional) Don't strip digits after matching:

         (dial-peer)  no digit-strip  

        By default, the router strips off the leftmost digits that match a destination pattern. The remaining digits are forwarded to other telephony devices. If you need to keep and forward the matching digits, use the no digit-strip command.

        (POTS only) Forward a specific number of digits:

         (dial-peer)  forward-digits  {  number   all   extra  } 

        Rather than forwarding the remaining digits after matching, the router can forward a specific number of digits (0 to 32), all digits up to the length of the destination pattern string, or just the extra digits to the right of the matched string (if the dialed string is longer than the destination pattern).

        (Optional) Use digit translation.

      Step 1. Create a translation ruleset:

       (global)  translation-rule   tag  
      A digit translation rule is created with an arbitrary tag (1 to 2147483647).

      Step 2. Define a translation rule:

       (translate)  rule   tag input-matched-pattern substituted-pattern  [  match-type substituted-type  ] 
      A rule is assigned an arbitrary tag (1 to 2147483647) to uniquely identify it within the ruleset. The input-matched-pattern is a string of digits to match against, including regular expression wildcard characters. The substituted-pattern is the string of digits that matches the pattern. The optional match-type and substituted-type fields specify ITU Q.931 numbering types to match and substitute, as defined earlier in the sixth bullet of Step 3e.

      Step 3. (Optional) Apply the ruleset to inbound calls on a voice port:

       (voiceport)  translate  {  called   calling  }  ruleset  
      The translation ruleset numbered ruleset is used on the voice port to translate either the called or calling number.

      Step 4. (Optional) Apply the ruleset to outbound calls on a dial peer:

       (dial-peer)  translate-outgoing  {  called   calling  }  ruleset  

      The translation ruleset numbered ruleset is used on the dial peer to translate either the called or calling number.

      • (Optional) Use a hunt group to rotate between multiple dial peers.

        First, assign a preference value to each dial peer in the hunt group:

         (dial-peer)  preference   value  

        Dial peers with identical matching destination patterns are used in a rotary fashion, according to the preference value (0 to 10; lower is preferable; the default is 0) given to each.

        (Optional) Next, change the order in which a hunt group is searched:

         (global)  dial-peer hunt   hunt-order  

        By default, hunt groups are searched according to the longest match in the destination pattern, an explicit preference value, or a random selection. If desired, the order can be changed according to the hunt-order: (longest match, preference, random; this is the default), 1 (longest match, preference, least-used), 2 (preference, longest match, random), 3 (preference, longest match, least-used), 4 (least-used, longest match, preference), 5 (least-used, preference, longest match), 6 (random), or 7 (least-used).

        (Optional) Stop hunting if a disconnect condition is reached:

         (global)  voice hunt  {  user-busy   invalid-number   unassigned-number  } 

        Hunting through a group of outbound destination dial peers can be stopped if certain conditions occur: user-busy (all ports to the called party are in use), invalid-number (the dialed number is invalid), or unassigned-number (the dialed number is not assigned at the destination router).

        Stop hunting if a call fails on a dial peer (optional):

         (dial-peer)  huntstop  

        If a call fails on a dial peer that is part of a hunt group, hunting will not continue.

      • (Optional) Configure a transmission speed for sending faxes to the dial peer:

         (dial-peer)  fax rate  {  2400   4800   7200   9600   12000   14400   disable   voice  } [  bytes   size  ] 

        Fax transmission occurs at the rate specified (in bps). The disable keyword disables the fax capability. Use the voice keyword to automatically set the fax rate to the highest value that is lower than the voice Codec in use. The fax payload size may be given as size (in bytes).

      • (Optional) Relay DTMF tones from the caller:

         (dial-peer)  dtmf-relay  [  cisco-rtp  ] [  h245-alphanumeric  ]   [  h245-signal  ] 

        -OR-

         (dial-peer)  dtmf-relay  

        When keys are pressed after a call is established, DTMF tones are relayed into the VoIP network using an out-of- band method: cisco-rtp (RTP with a Cisco proprietary payload), h245-alphanumeric (H.245 "alphanumeric" messages), or h245-signal (H.245 "signal" messages). If more than one message type is listed, one of the methods also supported by the far end is used.

        For VoFR, use the dtmf-relay keyword without any options. The DTMF tones are sent as FRF.11 Annex A frames .

    6. Configure additional VoIP parameters.

      • (Optional) Change the jitter buffer playout delay.

        First, choose the type of playout delay:

         (dial-peer)  playout-delay mode  {  adaptive   fixed  } 

        The playout delay can be set according to adaptive (dynamically adjusted during a call; this is the default) or fixed (use a constant playout delay). Normally, the default adaptive method is acceptable.

        Next, specify the playout delay time:

         (dial-peer)  playout-delay  {  nominal   milliseconds   maximum   milliseconds   minimum  {  default   low   high  }} 

        The jitter buffer is configured to use a nominal playout delay (0 to 1500 milliseconds; the default is 200) at the beginning of the call. In adaptive mode, the playout delay can vary from a maximum (40 to 1700 milliseconds; the default is 200) to a minimum of low (10 milliseconds), default (40 milliseconds), or high (80 milliseconds).

        Generally, the default playout delay settings are optimal. However, if you experience voice breakup, you can increase the playout delay. For bursty jitter conditions, increase the minimum playout delay for adaptive mode or the nominal delay for fixed mode. To get an idea of whether you are experiencing jitter problems, use the show call active voice command and look for nonzero values with LostPackets, EarlyPackets, or LatePackets.

      • (Optional) Use a technology prefix for a dial peer:

         (dial-peer)  tech-prefix   number  

        A technology prefix number (a string of 1 to 11 characters, containing 0 to 9, star [*], and pound [#]) can be assigned to a dial peer, which the H.323 gateway forwards to an H.323 gatekeeper. The gatekeeper can then make call decisions based on the technology that the local router is supporting.

    7. Configure additional VoFR or VoATM parameters.

      • (Optional) Choose a session protocol for a call:

         (dial-peer)  session protocol  {  cisco-switched   frf11-trunk  } 

        VoFR can use cisco-switched (proprietary session protocol, the default) or frf11-trunk (FRF.11 session protocol) to transport voice data. Use the default unless you are connecting to a non-Cisco device. If frf11-trunk is used, you must also configure a termination string with the called-number termination-string command. The termination-string is the destination's E.164 number.

        VoATM can use the cisco-switched keyword only by default. To disable it for non-Cisco trunks, precede the command with the no keyword.

      • (Optional) Generate voice packet sequence numbers:

         (dial-peer)  sequence-numbers  

        By default, voice packets are generated for VoFR without sequence numbers. Sequence numbers can be generated to allow the far end to detect out-of-sequence, duplicate, or lost packets.

      • (Optional) Select a signaling type to expect:

         (dial-peer)  signal-type  {  cas   cept   ext-signal   transparent  } 

        A VoFR or VoATM dial peer can expect a specific signaling type from the far end: cas (Channel-Associated Signaling), cept (E1 with cept/MELCAS signaling), ext-signal (external signaling, out-of-band or CCS), or transparent (T1/E1 signaling is passed through without modification or interpretation).

Table 12-6. Codec Types and Characteristics
Codec Type Codec Standard Data Rate (bps) Com- plexity High Com- plexity Medium VoIP Payload Bytes [multiple] (default) VoFR/VoATM Payload Bytes[multiple] (default)
g711alaw G.711 a-Law 64000 X X 80,160 [40](160) 40 to 240 [40](240)
g711ulaw G.711 u-Law 64000 X X 80,160 [40](160) 40 to 240 [40](240)
g723ar53 G.723.1 Annex A 5300 X   20 to 220 [20](20) 20 to 240 [20](20)
g723ar63 G.723.1 Annex A 6300 X   24 to 216 [24](24) 24 to 240 [24](24)
g723r53 G.723.1 5300 X   20 to 220 [20](20) 20 to 240 [20](20)
g723r63 G.723.1 6300 X   24 to 216 [24](24) 24 to 240 [24](24)
g726r16 G.726 16000 X X 20 to 220 [20](40) 10 to 240 [10](60)
g726r24 G.726 24000 X X 30 to 210 [30](60) 15 to 240 [15](90)
g726r32 G.726 32000 X X 40 to 200 [40](80) 20 to 240 [20](120)
g728 G.728 16000 X   10 to 230 [10](40) 10 to 240 [10](60)
g729r8 G.729 8000 X X 10 to 230 [10](20) 10 to 240 [10](30)
g729abr8 G.729 8000     10 to 230 [10](20) 10 to 240 [10](30)
g729ar8 G.729 8000     10 to 230 [10](20) 10 to 240 [10](30)
g729br8 G.729 8000 X X 10 to 230 [10](20) 10 to 240 [10](30)
clear-channel Clear Channel 64000        
Gsmefr Global System for Mobile Communications Enhanced Rate 12200        
Gsmfr Global System for Mobile Communications Full Rate 13200        


Cisco Field Manual[c] Router Configuration
Cisco Field Manual[c] Router Configuration
ISBN: 1587050242
EAN: N/A
Year: 2005
Pages: 185

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