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Implementing H.323 Gateways

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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.

23

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.

46

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 Flow
1 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.

83


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