1. Implementing H.323 Gateways
Gateway Deployments
Implementing H.323 Gateways

2. Overview of H.323 Gateways H.323 gateways perform these services:
Translation between audio, video, and data formats
Conversion between call setup signals and procedures
Conversion between communication control signals and procedures

3. H.323 Call Flow PSTN H.323 Gateway Setup Call Proceeding Signaling
Alerting
Connect
RTP Stream
Bearer or Media
Signaling
Release Complete

4. Why H.323? Advantages of H.323 gateways:
Dial plan can be configured directly on the gateway.
Translations can be defined per gateway.
Regional conditions can be met within multisite deployments.
Call routing configuration can be more specific than on Cisco CallManager.
There is no need for extra SRST-related call routing configuration.
There is no dependency on the Cisco Unified CallManager version.
More voice interface types are supported.
ISDN NFAS is supported.
Fax support is advanced.
Call preservation was enhanced for PRI calls since Cisco IOS Software Release 12.4(9)T.

5. Example: Regional Requirements
Calling party: Type: international 1
Calling Mel in the United States.
Calling Max in Germany. 2
Madrid 5
I’ve got an external call. To call back, I need to dial
I’ve got an external call. To call back, I need to dial 4 7
U.S.
Gateway
Germany Gateway
Translate calling and route to destination.
Translate calling and route to destination.
Mel 3 6
Max

6. H.323 Commands (Cont.) Configuring a router interface as a gateway:
router(config-if)#
h323-gateway voip bind srcaddr ip-address
Sets the IP address that is used as H.323 signaling source

7. H.323 Commands (Cont.) Configuring a DTMF relay:
router(config-dial-peer)#
dtmf-relay
Specifies how DTMF tones are forwarded
router(config-dial-peer)#
rtp payload-type type
Identifies the payload type of an RTP packet

8. H.323 Commands (Cont.) Configuring voice classes
router(config)#
voice class codec tag
Enters voice class codec configuration mode and assigns an identification tag
router(config)#
voice class type
Enters voice class configuration mode for special voice classes like H.323 and assigns an identification tag

9. H.323 Commands (Cont.) Configuring codecs
router(config-class)#
codec preference codec-type [bytes payload-size]
Adds a codec to the class and defines its preference
router(config-dial-peer)#
codec codec
Specifies the single codec used
router(config-dial-peer)#
voice-class type tag
Specifies the voice class parameters voice class to be used

10. H.323 Commands (Cont.) Tuning H.323 timers:
router(conf-voi-serv)#
h225 timeout setup seconds
Defines the H.323 setup timeout value
router(config-class)#
h225 timeout tcp establish seconds
Defines the H.323 TCP establish timeout value
router(conf-voi-serv)#
h225 timeout tcp call-idle value minutes
Defines the H.323 TCP call-idle timeout value

11. Integrating H.323 Gateways with Cisco Unified CallManager
Perform at least these steps to integrate an H.323 voice gateway into Cisco Unified CallManager:
Make sure that H.323 is enabled and specify the H.323 source address on the voice gateway.
Configure POTS and VoIP dial peer on the gateway.
Add a new gateway on Cisco Unified CallManager.
Configure the IP address and device pool for the gateway on Cisco Unified CallManager.
Add a new route pattern pointing to the gateway on Cisco Unified CallManager.

12. Integrating H. 323 Gateways with Cisco Unified CallManager: Enable H
voice service voip
h323
ccm-compatible
no shutdown !
interface FastEthernet0/0
ip address
h323-gateway voip bind srcaddr
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

13. Integrating H.323 Gateways with Cisco Unified CallManager: Define the POTS Dial Peer
dial-peer voice 9 pots
destination-pattern 9T
incoming called-number 2...
direct-inward-dial
port 1/0:23
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

14. Integrating H.323 Gateways with Cisco Unified CallManager: Add the First VoIP Dial Peer
voice class h323 1
h225 timeout tcp establish 3 !
dial-peer voice 21 voip
preference 1
destination-pattern 2...
voice-class h323 1
session target ipv4:
incoming called-number 9
codec g711alaw
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

15. Integrating H.323 Gateways with Cisco Unified CallManager: Add the Second VoIP Dial Peer
dial-peer voice 22 voip
preference 2
destination-pattern 2...
voice-class h323 1
session target ipv4:
incoming called-number 9
codec g711alaw
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

16. Integrating H.323 Gateways with Cisco Unified CallManager: Add the Gateway
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

17. Integrating H.323 Gateways with Cisco Unified CallManager: Configure Parameters
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

18. Integrating H.323 Gateways with Cisco Unified CallManager: Define the Route Pattern
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
H.323

19. Verifying an H.323 Integration with Cisco Unified CallManager
Dial-peer show and debug commands:
router#
show dial-peer voice tag
Displays detailed information about the specified dial peer
router#
show dial-peer voice summary
Displays a summary of all active dial peers
router#
debug voip dialpeer
Displays default debug output for all active VoIP dial peers

20. Verifying an H.323 Integration with Cisco Unified CallManager (Cont.)
H.323 debug commands:
router#
debug h225 asn1
Displays ASN.1 content of H.225 messages on the router
router#
debug h225 q931
Displays H.225 Q931 information element details
router#
debug h245 events
Displays H.245 events on the router

21. Summary
H.323 gateways are used to interconnect H.323 VoIP networks to networks that are using different signaling and/or bearer types.
An H.323 call flow consists of signaling and transmission of bearer and media packets.
H.323 allows extremely flexible decentralized call processing and call handling.
When implementing H.323 gateways, also consider DTMF and fax.
To configure an H.323 gateway, you must define at least the H.323 voice service, voice ports, and VoIP and POTS dial peers.

22. Summary (Cont.)
To integrate an H.323 gateway into Cisco Unified CallManager, at least a new gateway and a route pattern.
To verify the Cisco Unified CallManager integration of a gateway, there are useful show and debug commands available on Cisco IOS voice gateways.

24. Implementing MGCP Gateways
Gateway Deployments
Implementing MGCP Gateways

25. Overview of MGCP Gateways
Call processing is done by a call agent such as Cisco Unified CallManager.
MGCP gateways translate voice streams between audio signals and the IP packet network.
Configuration commands for MGCP define this information:
The path between the call agent and the gateway
The type of gateway
The type of calls handled by the gateway

26. Overview of MGCP Gateways (Cont.)
MGCP uses endpoints and connections to construct a call.
Endpoints:
Sources of or destinations for data
Physical or logical locations in a device
Connections:
Point-to-point
Multipoint
MGCP uses UDP for establishing audio connections over IP networks.

27. MGCP Call Flow PSTN MGCP Gateway RQNT RQNT Response NTFY CRCX
Signaling
CRCX Response
MDCX
MDCX Response
RTP Stream
Bearer or Media
NTFY
DLCX
Signaling
DLCX Response

28. Why MGCP? Advantages of MGCP gateways:
Centralized dial plan configuration on the Cisco Unified CallManager
Centralized gateway configuration on the Cisco Unified CallManager
Simple Cisco IOS gateway configuration
Supports QSIG supplementary services with Cisco Unified CallManager

29. Why MGCP? (Cont.) H.323 versus MGCP gateways:
Pros
Dial plan directly on the gateway
Translations defined per gateway
Regional requirements can be met
More specific call routing
Advanced fax support
Centralized dial-plan configuration
Centralized gateway configuration
Simple gateway configuration
Easy implementation
Support of QSIG supplementary services
Cons
Complex configuration
Extra SRST-related call routing configuration

30. MGCP Configuration Considerations
The MGCP gateway downloads the configuration from the Cisco Unified CallManager TFTP server.
Cisco Unified CallManager Administration
Configuration is done via the web-based Cisco Unified CallManager Administration page.
TFTP Server on Cisco Unified CallManager
MGCP Gateway

31. Basic MGCP Configuration
Configuring The MGCP configuration server:
Router(config)#
ccm-manager config server ip-address
Identifies the server from which the MGCP configuration should be downloaded
router(config)#
ccm-manager config
Enables the MGCP configuration download from the configured server

32. Basic MGCP Configuration (Cont.)
ccm-manager config server
ccm-manager config
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server
ccm-manager config !
mgcp
mgcp call-agent service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
no mgcp package-capability res-package
mgcp package-capability sst-package
no mgcp package-capability fxr-package
mgcp package-capability pre-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp rtp payload-type g726r16 static
mgcp profile default
Manually configured
Configuration after TFTP download

33. MGCP Commands Enable MGCP and define the MGCP call agent:
Router(config)#
mgcp
Enables the MGCP protocol
router(config)#
mgcp call-agent ip-address
Specifies the primary Cisco Unified CallManager
router(config)#
ccm-manager mgcp
Enables the MGCP gateway to support Cisco Unified CallManager

34. MGCP Commands (Cont.) Configuring MGCP redundancy:
Router(config)#
ccm-manager redundant-host ip-address
Identifies up to two backup Cisco Unified CallManager servers
router(config)#
ccm-manager switchback mode
Configures switchback mode for returning control to the primary Cisco Unified CallManager
router(config)#
ccm-manager fallback-mgcp
Enables the MGCP fallback feature

35. MGCP Commands (Cont.) Configuring MOH and DTMF: Enables music-on-hold
Router(config)#
ccm-manager music-on-hold
Enables music-on-hold
router(config)#
ccm-manager music-on-hold bind interface
Binds the multicast MOH feature to a designated interface
Router(config)#
mgcp dtmf-relay voip codec {all | low-bit-rate} mode {cisco | nse | out-of-band}
Defines DTMF relay parameters

36. Integrating MGCP Gateways with Cisco Unified CallManager
Perform at least these steps to integrate an MGCP-controlled voice gateway into Cisco Unified CallManager:
Add a new MGCP gateway on Cisco Unified CallManager.
Configure the device hardware on Cisco Unified CallManager.
Configure the ports on Cisco Unified CallManager.
Add a new route pattern pointing to the gateway on Cisco Unified CallManager.
Enable MGCP and define the Cisco Unified CallManager as the call agent on the voice gateway.

37. Integrating MGCP Gateways with Cisco Unified CallManager: Add the Gateway
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
MGCP

38. Integrating MGCP Gateways with Cisco Unified CallManager: Configure the Device
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
MGCP

39. Integrating MGCP Gateways with Cisco Unified CallManager: Configure the Port
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
MGCP

40. Integrating MGCP Gateways with Cisco Unified CallManager: Define the Route Pattern
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
MGCP

41. Integrating MGCP Gateways with Cisco Unified CallManager: Enable MGCP
ccm-manager config server
ccm-manager config
Phone1-1
2001
Phone1-2
2002
PSTN
Router1
MGCP

42. Verifying an MGCP Integration with Cisco Unified CallManager
MGCP show commands:
Router#
show mgcp
Displays the MGCP settings on the gateway
router#
show ccm-manager
Displays the Cisco Unified CallManager failover options
router#
show ccm-manager fallback-mgcp
Displays whether MGCP fallback is enabled and whether it is active or not (on or off)

43. Verifying an MGCP Integration with Cisco Unified CallManager (Cont.)
MGCP show commands (Cont.):
Router#
show mgcp endpoint
Displays endpoints controlled by MGCP
router#
show isdn status
Displays information about the status of ISDN Layers 1 to 3

44. Summary
Within an MGCP environment, call routing is done on Cisco Unified CallManager.
Within an MGCP environment, MGCP signaling is done between Cisco Unified CallManager and the gateway while RTP traffic is transferred between the phone and the gateway.
MGCP simplifies and centralizes configuration.
The MGCP configuration is downloaded from the Cisco Unified CallManager TFTP server by the gateway.
There are several MGCP commands available on Cisco IOS gateways.

45. Summary (Cont.)
To integrate an MGCP gateway into Cisco Unified CallManager, you must configure the device settings on the Cisco Unified CallManager.
You use show commands to view the MGCP status on Cisco IOS gateways.

47. Dial Plans on Cisco IOS Gateways
Implementing Dial Plans

48. PSTN Dial Plan Requirements
PSTN Requirements
Dial Plan Components
Inbound call routing
Call routing and path selection for inbound PSTN dial peer to outbound VoIP or local dial peer
Digit manipulation to transform inbound DNIS to endpoints
Outbound call routing
Call routing and path selection for inbound VoIP or local dial peer to outbound PSTN dial peer
Digit manipulation to transform outbound DNIS to PSTN requirements
Correct ANI presentation
Digit manipulation to transform ANI to meet PSTN requirements

49. Gateway modifies DNIS to 2001 and routes to CallManager.
Inbound PSTN Calls 4
H.323 call setup: DNIS 2001 3
Gateway modifies DNIS to 2001 and routes to CallManager. 5
PSTN
Phone1-1 rings.
H.323 Gateway
DID XXX
Phone1-1
2001
Phone1-2
2002
Call setup from PSTN:
DNIS
User dials 1 2

50. Gateway modifies DNIS and ANI.
Outbound PSTN Calls 2
H.323 call setup: DNIS
ANI 2001 3
Gateway modifies DNIS and ANI. 1
PSTN
User dials
H.323 Gateway
DID: XXX
Phone1-1
2001
Phone1-2
2002
Call setup:
DNIS
ANI´
PSTN phone rings. 5 4

51. ISDN Dial Plan Requirements
ISDN Requirements
Dial Plan Components
Correct PSTN inbound ANI presentation depending on TON
Digit manipulation to transform inbound PSTN ANI according to TON
Correct ISDN numbering plan and TON presentation
Manipulate ISDN numbering plan or TON to meet PSTN/PBX requirements

52. Gateway prepends 91 to ANI due to TON.
Inbound ISDN Calls
San Jose
DID range: XXX CM 3 4
Gateway prepends 91 to ANI due to TON.
H.323 call setup: ANI 5
PSTN
Phone1-1 rings.
ANI
H.323 Gateway
Call setup from PSTN:
ANI
TON National
Phone1-1
2001
Phone1-2
2002
User dials 1 2

53. Digit Manipulation Various mechanisms for digit manipulations:
Simple digit manipulation to strip off or prefix digits for POTS dial peers:
digit-strip
forward-digits
prefix
clid
Number expansion to globally inflate or deflate numbers:
num-exp
Typically used for short dials and site codes
Voice translation rules and profiles:
Powerful and complex digit manipulation using regular expressions

54. Digit Manipulation (Cont.)
POTS
VoIP
Inbound voice port digit manipulation
Global number expansion
Match inbound dial peer
Voice translation profile
CLID
Match outbound dial peer
Voice translation profile
CLID
Inbound
dial peer
match
Outbound
dial peer
match
VoIP
POTS
Global voice translation profiles
Global number expansion
Match inbound dial peer
CLID
Voice translation profile
Match outbound dial peer
Voice translation profile
Digit strip
Prefix
Forward digits
CLID

55. Simple Digit Manipulation for POTS Dial Peers
User dials
dial-peer voice 9 pots
destination 9T
PSTN
Phone1-1
2001
H.323 Gateway
DID: XXX
Command
DNIS
no digit-strip
digit-strip (default)
forward-digits 4
6666
prefix 9 and digit-strip

56. Digit Manipulation with Number Expansion
num-exp
dial-peer voice 9 pots
destination 9T
PSTN
Phone1-1
2001
H.323 Gateway
DID: XXX 1 2 3
User dials 6666.
Number is inflated to and routed to PSTN.
Phone rings.

57. Voice Translation Rules and Profiles
Voice translation rules define up to 15 sub rules to manipulate digits, TONs, and numbering plans.
Voice translation profiles reference up to three rules:
Called: Translation rule for the called number
Calling: Translation rule for the calling number
Redirect-called: Translation rule for the redirect number
Voice translation profiles can be referenced by:
Dial peers, voice ports, any inbound VoIP call, specific range of source IP addresses in VoIP calls, trunk groups, NFAS controllers, or SRST

58. Voice Translation Rules and Profiles (Cont.)
Called
Rule 1 2 3 4 5 6 7 8 9 10
Dial Peer 11 12 13 14 15
Voice Port
VoIP Incoming
Calling
Rule 1 2 3 4 5
Incoming
Source IP Group 6 7 8 9 10
Outgoing 11 12 13 14 15
Trunk Group
NFAS
Redirected
Called
Rule 1 2 3 4 5
SRST 6 7 8 9 10 11 12 13 14 15

59. Voice Translation Rules and Profiles (Cont.)
Char
Description ^
Match the expression at the start of a line. $
Match the expression at the end of the line. /
Delimiter that marks the start and end of both the matching and replacement strings. \
Escape the special meaning of the next character. -
Indicates a range when not in the first or last position. Used with the'[' and ']'.
[list]
Match a single character in a list.
[^list]
Do not match a single character specified in the list. .
Match any single character. *
Repeat the previous regular expression zero or more times. +
Repeat the previous regular expression one or more times. ?
Repeat the previous regular expression zero or one time (use CTRL-V in order to enter in IOS).
\(\)
Groups regular expressions. Use \1-9 to refer to matched groups.
&
Match the substring (matched string). Can also use \0.

60. Voice Translation Rule Search-and-Replace Examples
Input String
Output String
/^9/ //
/^2001/ /3001/
2001
3001
/^[23]…/ /4000/
2025 or 3051
4000
/^2…/ /801&/
/^2…/ /801\0/
/.*/ /91&/ type national national
type national
type national

61. Voice Translation Rule Search-and-Replace Examples (Cont.)/
\(9\)
\([^10].*\) / / \1
1408 \2 /
Input
Output 9 9
1408

62. Voice Translation Profile Example: Search and Replace
PSTN-IN
Called
Rule 1 1
/^ / /2/ 1
/^.*/ /9&/ type subscriber scubscriber 2
/^.*/ /91&/ type national national 3
/^.*/ /9011&/ type international international
Calling
Rule 2
Redirected
Called
voice translation-rule 1
rule 1 /^ / /2/
voice translation-rule 2
rule 1 /^.*/ /9&/ type subscriber subscriber
rule 2 /^.*/ /91&/ type national national
rule 3 /^.*/ /9011&/ type international international
voice translation-profile pstn-in
translate called 1
translate calling 2

63. Cisco Unified CallManager Express with Voice Translation Profiles
Replaces the dial plan pattern and covers inbound and outbound routing of any dial peers.
voice translation-rule 1
rule 1 /^ / /2/
voice translation-profile pstn-in
translate called 1
voice-port 0/0/0:23
translation-profile incoming pstn-in
dial-peer voice 2001 pots
port 1/0/0
destination-pattern 2001
PSTN
CME
DID: XXX
FXS Phone1-1
2001 3 2 1
User dials
Phone rings.
Profile modifies DNIS to 2001.

64. Digit Manipulation Commands
router(config-dial-peer)#
digit-strip
Enables stripping of matched digits (default)
router(config-dial-peer)#
prefix digits
Prefixes the specified digits after the pattern has matched
router(config-dial-peer)#
forward-digits [0-32]|all|extra
Specifies how many digits to forward, right-justified

65. Digit Manipulation Commands (Cont.)
router(config)#
num-exp dialed-digits substitution
Matches on dialed digits and replaces them (wildcards allowed)
router(config)#
voice translation-rule rule-tag
Creates or modifies a voice translation rule with the specified numeric tag
router(cfg-translation-rule)#
rule precedence /match/ /replace/ [type {match-type replace-type} [plan {match-plan replace-plan}]]
Creates a match-and-replace rule

66. Digit Manipulation Commands (Cont.)
router(config)#
voice translation-profile profile-name
Creates or modifies a voice translation profile with the specified name
router(cfg-translation-profile)#
translate {called | calling | redirect-called} translation-rule-number
Binds a translation rule to a profile

67. Digit Manipulation Commands (Cont.)
router(dial-peer)#
translation-profile {incoming | outgoing} name
Binds a translation profile to a dial peer
router#
test voice translation-rule number input-test-string [type match-type [plan match-plan] ]
Tests a translation rule with a user-defined test string

68. Call Routing and Path Selection Commands
router(config-dial-peer)#
destination-pattern [+]string[T]
Defines the destination pattern of a dial peer that will be used during outbound dial-peer matching
router(config-dial-peer)#
incoming called-number [+]string[T]
Specifies the incoming called number that will be used during inbound dial-peer matching
router(config-dial-peer)#
answer-address [+]string[T]
Specifies the incoming calling number will be used during inbound dial-peer matching

69. Call Routing and Path Selection Commands (Cont.)
router(config-dial-peer)#
direct-inward-dial
Uses the inbound POTS DNIS to match the outbound dial peer (usually required for any DID PSTN trunk)
router(config-dial-peer)#
preference [0-9]
Specifies the preference of a dial peer; lower number = high priority default=0
router(config)#
no dial-peer outbound status-check pots
Disables status checking of outbound POTS dial peers during call setup; always includes dial peer in call routing, even if POTS status is down; and useful for some ISDN links

70. Configuring PSTN Dial Plans
Configure digit manipulation for inbound calls.
Define dial peers.
Configure inbound dial-peer matching.
Configure digit manipulation for outbound calls.
Configure digit manipulation for intersite calls.

71. PSTN Dial Plan Scenario
San Jose
Chicago
Users should be able to reach the other site via an extension.
PSTN
CM1:
CM2:
DID: XXXX
DID: XXXX
Router1
H.323
Router3
CME
Phone1-1
2001
Phone1-2
2002
IP WAN
Phone2-1
3001
Phone2-2
3002

72. Step 1: Configure Digit Manipulation for Inbound Calls
Router1
H.323
Router3
CME
voice translation-rule 1
rule 1 /^ / /2/
voice translation-rule 2
rule 1 /^.*/ /9&/ type subscriber subscriber
rule 2 /^.*/ /91&/ type national national
rule 2 /^.*/ /9011&/ type international international
voice translation-profile pstn-in
translate called 1
translate calling 2
voice-port 0/0/0:23
translation-profile incoming pstn-in
voice translation-rule 1
rule 1 /^ / /3/
voice translation-rule 2
rule 1 /^.*/ /9&/ type subscriber subscriber
rule 2 /^.*/ /91&/ type national national
rule 2 /^.*/ /9011&/ type international international
voice translation-profile pstn-in
translate called 1
translate calling 2
voice-port 0/0/0:23
translation-profile incoming pstn-in

73. Step 2: Define Dial Peers
Router1
H.323
Router3
CME
dial-peer voice 9 pots
destination-pattern 9T
direct-inward-dial
port 0/0/0:23
dial-peer voice 20 voip
destination-pattern 2...
session-target ipv4:
codec g711ulaw
dial-peer voice 21 voip
preference 1
session-target ipv4:
dial-peer voice 9 pots
destination-pattern 9T
direct-inward-dial
port 0/0/0:23

74. Step 3: Configure Inbound Dial-Peer Matching
Router1
H.323
Router3
CME
dial-peer voice 9 pots
destination-pattern 9T
incoming called-number 2...
direct-inward-dial
port 0/0/0:23
dial-peer voice 20 voip
answer-address 2...
destination-patter 2...
session-target ipv4:
codec g711ulaw
dial-peer voice 21 voip
preference 1
session-target ipv4:
dial-peer voice 9 pots
destination-pattern 9T
incoming called-number 3...
direct-inward-dial
port 0/0/0:23

75. Step 4: Configure Digit Manipulation for Outbound Calls
Router1
H.323
Router3
CME
voice translation-rule 3
rule 1 /^2/ / /
voice translation-profile pstn-out
translate calling 3
voice-port 0/0/0:23
translation-profile outgoing pstn-out
voice translation-rule 3
rule 1 /^3/ / /
voice translation-profile pstn-out
translate calling 3
voice-port 0/0/0:23
translation-profile outgoing pstn-out

76. Step 5: Configure Digit Manipulation for Intersite Calls
Router1
H.323
Router3
CME
num-exp
num-exp

77. Inbound PSTN Call Flow PSTN 3 2 1 1 2 3 H.323 Gateway DID: 4085552XXX 1
voice-port 0/0/0:23
translation-profile incoming pstn-in
ANI
DNIS
2001
Outgoing
Incoming 2
dial-peer voice 9 pots
destination-pattern 9T
incoming called-number 2...
direct-inward-dial
port 0/0/0:23
ANI
DNIS
2001
Outgoing
Incoming 3
dial-peer voice 20 voip
answer-address 2...
destination-patter 2...
session-target ipv4:
codec g711ulaw
ANI
DNIS
2001
Outgoing
Incoming

78. Outbound PSTN Call Flow1 2 3
PSTN
H.323 Gateway
DID: XXX 1
dial-peer voice 20 voip
answer-address 2...
destination-patter 2...
session-target ipv4:
codec g711ulaw
ANI
DNIS
2001
Outgoing
Incoming 2
dial-peer voice 9 pots
destination-pattern 9T
incoming called-number 2...
direct-inward-dial
port 0/0/0:23
ANI
DNIS
2001
Outgoing
Incoming 3
voice-port 0/0/0:23
translation-profile outgoing pstn-out
ANI
DNIS
2001
Outgoing
Incoming

79. Verifying PSTN Dial Plans
router#
show dialpeer voice summary
Displays a summary of all dial peers
Session Target
Voice Port
Router1#show dial-peer voice summary
dial-peer hunt 0
AD PRE PASS OUT
TAG TYPE MIN OPER PREFIX DEST-PATTERN FER THRU SESS-TARGET STAT PORT
pots up up T up 1/0:23
voip up up syst ipv4:
voip up up syst ipv4:
Administrative and Operational State
Destination Pattern
Preference

80. Verifying PSTN Dial Plans (Cont.)
router#
debug isdn q931
Debugs ISDN Layer 3 information, which includes DNIS and ANI information
router#
debug voip dialpeer
Debugs dial-peer matching
router#
debug voice translation
Debugs voice-translation-rule operation

81. Summary
Interworking with the PSTN requires appropriate call routing and digit manipulation.
Digit manipulation based on a TON and numbering plan is required for ISDN networks.
Voice translation profiles are powerful tools for digit manipulation but sometimes can be replaced by more simple mechanisms.
Voice translation profiles consist of voice translation rules that use regular expressions for match and replace.
Voice translation profiles can be used to work around issues with the dialplan-pattern command.
Voice translation rules are configured to be used within voice translation profiles.

82. Summary (Cont.)
Call routing and path selection are configured using dial peers.
Configuring a PSTN dial plan includes configuration of digit manipulation, inbound dial peers, inbound dial-peer matching and outbound dial-peer matching.
Verify correct dial-peer matching and digit manipulation using show and debug commands.