Curso de Telefonía IP para el MTC. Sesión 5-1 Implementación de Gateways SIP. Mg. Antonio Ocampo Zúñiga

Transcription

1 Curso de Telefonía IP para el MTC Sesión 5-1 Implementación de Gateways SIP Mg. Antonio Ocampo Zúñiga

2 SIP Fundamentals SIP is a simple extensible protocol. SIP is defined in IETF RFC 2543 and RFC SIP creates, modifies, and terminates multimedia sessions with one or more participants. SIP leverages various standards: RTP, RTCP, HTTP, SDP, DNS, SAP, MGCP, and RTSP.

3 SIP Fundamentals Handles the transfer and termination of calls SIP performs addressing by E.164, , or DNS service record. SIP is ASCII text-based for easy implementation and debugging.

4 SIP Fundamentals (Cont.) SIP provides these capabilities: Determines the location of the target endpoint Determines the media capabilities of the target endpoint Determines the availability of the target endpoint Establishes a session between the originating and target endpoints

5 How SIP Works ASCII-based protocol User identified by a unique SIP address sip:userid@gateway.com Users register with registrar server using their assigned SIP addresses When user initiates a call, a SIP request is sent to a SIP server Location of end user can be dynamically registered with the SIP server

6 Why SIP Advantages of SIP gateways: Dial-plan configuration directly on the gateway Translations defined per gateway Advanced support for third-party telephony system integration Interoperability with third-party voice gateways Support of third-party end devices (SIP phones)

7 SIP Architecture SIP Proxy, Registrar, Location, and Redirect Servers SIP SIP User Agents (UAs) SIP RTP SIP SIP Gateway PSTN T1 or PRI Legacy PBX

8 SIP Call Flow SIP Proxy Server SIP Gateway PSTN Invite (SDP) 100 Trying 180 Ringing 200 OK ACK Invite (SDP) 100 Trying 180 Ringing 200 OK ACK Signaling RTP Stream BYE 200 OK Bearer or Media Signaling

9 Direct Call Setup SIP Gateway SIP Gateway IP Calling Party Invite (SDP) 100 Trying 180 Ringing 200 OK Called Party SIP Signaling and SDP (UDP or TCP) ACK RTP Stream BYE 200 OK Bearer or Media (UDP) Signaling

10 Call Setup Using a Proxy Server SIP Gateway Proxy Server SIP Gateway IP Calling Party Invite (SDP) Invite (SDP) Called Party SIP Signaling and SDP (UDP or TCP) 100 Trying 180 Ringing 200 OK 100 Trying 180 Ringing 200 OK ACK ACK Bearer or Media (UDP) BYE RTP Stream BYE 200 OK 200 OK

11 Call Setup Using a Redirect Server SIP Gateway Redirect Server SIP Gateway IP Calling Party Invite Moved Called Party SIP Signaling and SDP (UDP or TCP) Invite Trying Ringing OK ACK Bearer or Media (UDP) RTP Stream BYE 200 OK

12 SIP Addresses Fully qualified domain names E.164 addresses user=phone Mixed addresses sip: ;

13 Address Registration Registrar Server Redirect Server Location Database SIP Proxy (UAS) Register Here I am! SIP UACs SIP UACs SIP Gateway

14 Address Resolution Registrar Server Redirect Server Location Database Where is the name or phone number? SIP Proxy

15 Configuring a SIP Gateway Enable SIP voice services Configure SIP service Transport Bind interface Configure SIP User Agent (UA) Timers Authentication SIP servers

16 Configuring a SIP Gateway Configure dial-peer SIP parameters Session protocol Session target DTMF relay

17 Integrating Gateways with a SIP ITSP Sip2.pucp.edu.pe Configure Gateway to connect to a SIP service provider network and route external calls via that connection. SIP ITSP SIP Gateway router(config)# voice service voip router(conf-voi-serv)# sip router(conf-serv-sip)# session transport udp router(conf-serv-sip)# bind control source-interface Loopback 0 router(conf-serv-sip)# bind media source-interface Loopback 0

18 Integrating Gateways with a SIP ITSP (Cont.) Sip2.pucp.edu.pe SIP ITSP SIP Gateway router(config)# sip-ua router(config-sip-ua)# authentication username JDoe password secret router(config-sip-ua)# registrar dns:sip2.pucp.edu.pe expires 3600 router(config-sip-ua)# sip-server dns:sip2.pucp.edu.pe router(config-sip-ua)# retry invite 2 router(config-sip-ua)# retry response 2 router(config-sip-ua)# retry bye 2 router(config-sip-ua)# retry cancel 2

19 Integrating Gateways with a SIP ITSP (Cont.) Sip2.pucp.edu.pe Gateway SIP Gateway Ext.: 2 SIP ITSP router(config)# dial-peer voice 2000 voip router(config-dial-peer)# destination-pattern 2... router(config-dial-peer)# session protocol sipv2 router(config-dial-peer)# session target sip-server router(config-dial-peer)# dtmf-relay rtp-nte router(config)# dial-peer voice 2001 voip router(config-dial-peer)# destination-pattern 2... router(config-dial-peer)# session protocol sipv2 router(config-dial-peer)# session target ipv4: router(config-dial-peer)# dtmf-relay sip-notify router(config-dial-peer)# preference 1 router(config)# dial-peer voice 90 voip router(config-dial-peer)# destination-pattern 9T router(config-dial-peer)# session target ipv4: router(config-dial-peer)# session protocol sipv2 router(config-dial-peer)# dtmf-relay rtp-nte

20 Verifying SIP Gateways Command show sip-ua service show sip-ua status show sip-ua register status show sip-ua timers show sip-ua connections show sip-ua calls show sip-ua statistics Description Displays the status of the SIP VoIP service. Displays the status of the SIP UA. Displays the status of E.164 numbers that a SIP gateway has registered with an external primary SIP registrar. Displays SIP UA timers. Displays active SIP UA connections. Displays active SIP UA calls. Displays SIP traffic statistics.

21 Verifying SIP Gateways (Cont.) Router# show sip service SIP Service is up Router# show sip-ua status SIP User Agent Status SIP User Agent for UDP : ENABLED SIP User Agent for TCP : ENABLED SIP User Agent bind status(signaling): DISABLED SIP User Agent bind status(media): DISABLED SIP max-forwards : 6 SIP DNS SRV version: 1 (rfc 2052) Redirection (3xx) message handling: ENABLED Router# show sip-ua timers SIP UA Timer Values (millisecs) trying 500, expires , connect 500, disconnect 500 comet 500, prack 500, rel1xx 500, notify 500 refer 500, register 500

22 Verifying SIP Gateways (Cont.) Router# show sip-ua register status Line peer expires(sec) registered no no no no

23 Verifying SIP Gateways (Cont.) router# show sip-ua calls SIP UAC CALL INFO Number of SIP User Agent Client(UAC) calls: 0 SIP UAS CALL INFO Call 1 SIP Call ID : D215F304-7B5A11DC-8005EA1A- 6A8F4AD@ State of the call : STATE_ACTIVE (7) Substate of the call : SUBSTATE_NONE (0) Calling Number : Called Number : 1003 Bit Flags : 0x A 0x x488 CC Call ID : 1 Source IP Address (Sig ): Destn SIP Req Addr:Port : :5060 Destn SIP Resp Addr:Port: :56884 Destination Name :

24 Verifying SIP Gateways (Cont.).. Number of Media Streams : 1 Number of Active Streams: 1 RTP Fork Object : 0x0 Media Stream 1 State of the stream : STREAM_ACTIVE Stream Call ID : 1 Stream Type : voice-only (0) Negotiated Codec : g729r8 (20 bytes) Codec Payload Type : 18 Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 Media Source IP Addr:Port: :18050 Media Dest IP Addr:Port : :16522 Orig Media Dest IP Addr:Port : :0 Number of SIP User Agent Server(UAS) calls: 1

25 Debug Commands Command debug asnl events debug voip ccapi inout debug voip ccapi protoheaders debug ccsip Description Verifies that the SIP subscription server is up. Shows every interaction with the call control API. Displays messages sent between the originating and terminating gateways. For general SIP debugging; for example views direction-attribute settings and port and network address-translation traces.

26 Debug Commands (Cont.) HQ-1# debug ccsip messages SIP Call messages tracing is enabled HQ-1# *Mar 6 14:19:14: Sent: INVITE sip: @ ;user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP :55820 From: " " <sip: @ > To: <sip: @ ;user=phone;phone-context=unknown> Date: Sat, 06 Mar :19:14 GMT Call-ID: ABBAE7AF E2-0-1CD274BC@ Cisco-Guid: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: Contact: <sip: @ :5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 138

27 Debug Commands (Cont.) *Mar 6 14:19:16: Received: SIP/ OK Via: SIP/2.0/UDP :55820 From: " " <sip: @ > To: <sip: @ ;user=phone;phonecontext=unknown>;tag=27dbc6d Date: Mon, 08 Mar :45:12 GMT Call-ID: ABBAE7AF E2-0-1CD274BC@ Timestamp: Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: <sip: @ :5060;user=phone> CSeq: 101 INVITE Content-Type: application/sdp Content-Length: 138 v=0 o=ciscosystemssip-gw-useragent IN IP s=sip Call t=0 0 c=in IP m=audio RTP/AVP 0

28 Debug Commands (Cont.) *Mar 6 14:19:19: Received: BYE sip: @ :5060;user=phone SIP/2.0 Via: SIP/2.0/UDP :53600 From: <sip: @ ;user=phone;phonecontext=unknown>;tag=27dbc6d To: " " <sip: @ > Date: Mon, 08 Mar :45:14 GMT Call-ID: ABBAE7AF E2-0-1CD274BC@ User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Max-Forwards: 6 Timestamp: CSeq: 101 BYE Content-Length: 0

29 Summary SIP is defined by IETF RFC 2543 and RFC 3261 and allows easy integration with thirdparty VoIP networks. There are five advantages to using SIP gateways as voice gateways. SIP is modeled on the interworking of UAs and network servers. A SIP call flow consists of signaling and transmitting bearer and media packets.

30 Summary (Cont.) A SIP address consists of an optional user ID, a host description, and optional parameters to qualify the address more precisely. SIP call-setup models include direct, proxy server, and redirection.

31 Curso de Telefonía IP para el MTC Sesión 5-2 Planes de marcación Mg. Antonio Ocampo Zúñiga

32 Defining Dial Plans A dial plan defines how calls are interconnected: Endpoint addressing: Internal destination accessibility can be provided by assigning directory numbers to all endpoints. Call routing and path selection: Different paths can be selected to reach the same destination. Digit manipulation: Digits can be manipulated prior to or after a routing decision has been made.

33 Defining Dial Plans Calling privileges: Different groups of devices can be assigned to different classes of service, by granting or denying access to certain destinations or resources. Call coverage: Special groups of devices can be created to handle incoming calls for a certain service according to different rules, avoiding dropped calls.

34 Defining Dial Plans (Cont.) Dial Plan Component Gateway Communications Manager Endpoint addressing Call routing and path selection Digit manipulation Calling privileges Call coverage POTS* dial peers for FXS ports and ephone-dn Dial peers Voice translation profiles prefix, digit-strip, forward-digits, and num-exp Class of restriction (COR) and COR lists Dial peers, hunt groups, and call applications Directory number Route patterns, route groups, route lists, translation patterns, partitions, and calling search spaces Translation patterns, route patterns, and route lists Partitions, calling search spaces, and FAC*s Line groups, hunt lists, and hunt pilots *POTS = Plain old telephone service *FXS = Foreign Exchange Station *FACs = Forced Authorization Codes

35 Defining Dial Plans (Cont.) Lima DID: XXX Trujillo DID: XXX IP WAN Router1 Router2 Phone Phone PSTN Phone Phone

36 Endpoint Addressing Directory numbers are assigned to endpoints (e.g., phones) Internal extensions are mapped to inbound PSTN calls Often dependant on range of DID numbers Auto-attendant can be used for non-did numbers

37 Endpoint Addressing The biggest challenge: creating an endpoint addressing scheme in multisite environments Primarily a Communications Manager issue Gateways simply route the call to the call agent Exception: Gateway routes calls during SRST mode

38 Call Routing and Path Selection Route the call depending on the dialed number. Select the appropriate path. Handled by dial peers on gateways: Inbound and outbound dial-peer matching determines the routing. Complexity depends on design.

39 Digit Manipulation Digit manipulation is closely connected with call routing and path selection. Inbound calls: Called number needs to match internally used patterns. Calling number should be presented as a dialable number. Outbound calls: Called number needs to satisfy internal and PSTN requirements. Calling number needs to be dialable. Special consideration needs to given to emergency calls. Various commands are available for digit manipulation: digit-strip, prefix, forward-digits, num-exp, voice translation-profile, and clid

40 Calling Privileges Defines the destinations a user is allowed to call Often used to control telephony charges: Blocks costly service numbers Restricts international calls Often called Class of Service in PBX systems Not the same as Layer 2 CoS

41 Call Coverage Call coverage ensures that all incoming calls are answered: For individuals: Call forwarding if original called phone is not answering For user groups with pilot numbers: Hunt through multiple phones Hunt through multiple user groups

42 Scalable Dial Plans North American Numbering Plan (NANP) - N X X N X X - X X X X 10-Digit Dial Plan User dials Area Code Local Exchange (CO) Subscriber Local PSTN Remote PSTN Dial plans contain specific dialing patterns for a user who wants to reach a particular telephone number.

43 Scalable Dial Plans (Cont.) Site E Site F Site A Digits Centrex IP 10 Digits Site B 4 Digits Site D 10 Digits 10 Digits 4 Digits Site C Digits

44 Attributes of a Scalable Dial Plan Dial plan logic distribution Hierarchical numbering plan summarization Simplicity in provisioning Reduction in postdial delay Availability and fault tolerance Conformance to public standards

45 PSTN Dial Plan Requirements PSTN Requirements Inbound call routing Outbound call routing Correct ANI presentation Dial Plan Components 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 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 Digit manipulation to transform ANI to meet PSTN requirements

46 Inbound PSTN Calls 3 Gateway modifies DNIS to 2001 and routes to voice port. Phone1-1 rings. 4 Gateway DID XXX 2 PSTN Phone Phone Call setup from PSTN: DNIS User dials

47 Outbound PSTN Calls H.323 call setup: DNIS ANI 2001 Gateway modifies DNIS and ANI. 2 1 User dials PSTN Gateway DID: XXXX Phone Phone Q.931 call setup: DNIS ANI PSTN phone rings.

48 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 and PBX requirements

49 4 Inbound ISDN Calls Lima DID range: XXX H.323 call setup: ANI Gateway prepends 91 to ANI due to TON. 3 Phone1-1 rings. ANI PSTN Gateway Phone Phone Call setup from PSTN: ANI TON National 2 User dials

50 Configuring PSTN Dial Plans 1. Configure digit manipulation for inbound and outbound PSTN calls. 2. Configure digit manipulation for intersite calls. 3. Configure inbound and outbound dial-peer matching.

51 PSTN Dial Plan Scenario Lima PSTN Users should be able to reach the other site via an extension. Trujillo DID: XXX DID: XXX Router1 Router3 Phone Phone IP WAN Phone Phone

52 Digit Manipulation for Inbound Calls Lima Trujillo PSTN Router1 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 Router3 voice translation-rule 1 rule 1 /^ / /3/ voice translation-profile pstn-in translate called 1 voice-port 0/0/0:23 translation-profile incoming pstn-in

53 Digit Manipulation for Outbound Calls Lima Trujillo PSTN Router1 voice translation-rule 2 rule 1 /^2/ / / voice translation-profile pstn-out translate calling 2 voice-port 0/0/0:23 translation-profile outgoing pstn-out Router3 voice translation-rule 2 rule 1 /^3/ / / voice translation-profile pstn-out translate calling 2 voice-port 0/0/0:23 translation-profile outgoing pstn-out

54 Global Digit Manipulation for Intersite Calls Lima Trujillo PSTN Router1 Router3 num-exp num-exp

55 Outbound Dial Peer Matching Lima Trujillo PSTN Router1 Router3 dial-peer voice 910 pots destination-pattern 9[2-9]..[2-9]... direct-inward-dial port 0/0/0:23 dial-peer voice 910 pots destination-pattern 9[2-9]..[2-9]... direct-inward-dial port 0/0/0:23

56 Inbound Dial Peer Matching Lima Trujillo PSTN Router1 Router3 dial-peer voice 910 pots destination-pattern 9[2-9].. [2-9]... incoming called-number 2... direct-inward-dial port 0/0/0:23 dial-peer voice 910 pots destination-pattern 9[2-9].. [2-9]... incoming called-number 3... direct-inward-dial port 0/0/0:23

57 Inbound PSTN Call Flow /0/1 DID: XXX 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 PSTN Incoming Outgoing DNIS ANI dial-peer voice 910 pots destination-pattern 9[2-9]..[2-9]... incoming called-number 2... direct-inward-dial port 0/0/0:23 dial-peer voice 2001 pots destination-pattern 2001 port 0/0/1 Incoming Outgoing DNIS ANI Incoming Outgoing DNIS ANI

58 Outbound PSTN Call Flow 2001 DID: XXX PSTN dial-peer voice 2001 pots destination-pattern 2001 port 0/0/1 Incoming Outgoing DNIS ANI dial-peer voice 910 pots destination-pattern 9[2-9]..[2-9]... incoming called-number 2... direct-inward-dial port 0/0/0:23 voice translation-rule 2 rule 1 /^2/ / / voice translation-profile pstn-out translate calling 2 voice-port 0/0/0:23 translation-profile outgoing pstn-out Incoming Outgoing DNIS ANI Incoming Outgoing DNIS ANI

59 Verifying PSTN Dial Plans Command show dial-peer voice <number> show dial-peer voice summary show dialplan number dialstring [carrier identifier] [fax huntstop voice] [timeout] Description Displays information for a specific voice dial peer. Displays a short summary of each voice dial peer. Displays which outgoing dial peer is reached when a particular telephone number is dialed.

60 Verifying PSTN Dial Plans (Cont.) router# show dial-peer 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 9 pots up up 9T 0 up 1/0:23 20 voip up up syst ipv4: voip up up syst ipv4: Administrative and Operational States Destination Pattern Preference

61 Verifying PSTN Dial Plans (Cont.) router# show dialplan number 1001 VoiceEncapPeer20001 peer type = voice, information type = voice, description = `', tag = 20001, destination-pattern = `1001$', answer-address = `', preference=0, CLID Restriction = None CLID Network Number = `' CLID Second Number sent CLID Override RDNIS = disabled, source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 20001, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited, DTMF Relay = disabled, URI classes: Destination = huntstop = enabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement

62 Verifying PSTN Dial Plans (Cont.) Translation profile (Incoming): Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' advertise 0x40 capacity_update_timer 25 addrfamily 4 oldaddrfamily 4 type = pots, prefix = `', forward-digits 0 session-target = `', voice-port = `50/0/11', direct-inward-dial = disabled, digit_strip = enabled, register E.164 number with H323 GK and/or SIP Registrar = TRUE fax rate = system, payload size = 20 bytes supported-language = '' Time elapsed since last clearing of voice call statistics never Connect Time = 0, Charged Units = 0, Successful Calls = 2, Failed Calls = 0, Incomplete Calls = 0 Accepted Calls = 0, Refused Calls = 0, Last Disconnect Cause is "10 ", ast Disconnect Text is "normal call clearing (16)", Last Setup Time = Matched: 1001 Digits: 5 Target:

63 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

64 debug isdn q931 Router# debug isdn q931 RX <- SETUP pd = 8 callref = 0x06 Bearer Capability i = 0x8890 Channel ID i = 0x89 Calling Party Number i = 0x0083, TX -> CONNECT pd = 8 callref = 0x86 RX <- CONNECT_ACK pd = 8 callref = 0x06

65 debug isdn q931 (Cont.) Router# debug isdn q931 TX -> SETUP pd = 8 callref = 0x04 Bearer Capability i = 0x8890 Channel ID i = 0x83 Called Party Number i = 0x80, RX <- CALL_PROC pd = 8 callref = 0x84 Channel ID i = 0x89 RX <- CONNECT pd = 8 callref = 0x84 TX -> CONNECT_ACK pd = 8 callref = 0x04... Success rate is 0 percent (0/5)

66 debug voip dialpeer *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer: Result=Success(0); Incoming Dial-peer=1 Is Matched *Apr 18 21:07:35.291: //- 1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore: Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=83103 *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype: Is Incoming=TRUE, Number Expansion=FALSE *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchCore: Dial String=83103, Expanded String=83103, Calling Number= Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_FAX *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchCore: Result=-1 *Apr 18 21:07:35.291: //- 1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore: Match Rule=DP_MATCH_ANSWER; Calling Number= *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype: Is Incoming=TRUE, Number Expansion=FALSE

67 debug voip dialpeer (Cont.) *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchCore: Dial String=, Expanded String=, Calling Number= T Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_FAX *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchCore: Result=-1 *Apr 18 21:07:35.291: //- 1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore: Match Rule=DP_MATCH_ORIGINATE; Calling Number= *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype: Is Incoming=TRUE, Number Expansion=FALSE *Apr 18 21:07:35.291: //-1/xxxxxxxxxxxx/DPM/dpMatchCore: Dial String=, Expanded String=, Calling Number= T Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_FAX *Apr 18 21:07:35.295: //-1/xxxxxxxxxxxx/DPM/dpMatchCore: Result=-1

68 debug voice translation Router# debug voice translation 00:51:56:regxrule_get_profile_from_trunkgroup:Voice port 0x64143DA8 does not belong to any trunk group 00:51:56:regxrule_get_profile_from_trunkgroup:Voice port 0x64143DA8 does not belong to any trunk group 00:51:56:regxrule_stack_pop_RegXruleNumInfo:stack=0x63DECAF4; count=1 00:51:56:regxrule_stack_push_RegXruleNumInfo:stack=0x63DECAF4; count=0 This output shows the details of the original number following "regxrule_profile_translate". 00:51:56:regxrule_profile_translate:number= type=unknown plan=unknown numbertype=calling Following "regxrule_profile_match", the output shows that rule 1 in the translation rule 1001 was a match and the details of the SED substitution are shown. 00:51:56:regxrule_profile_match:Matched with rule 1 in ruleset :51:56:regxrule_profile_match:Matched with rule 1 in ruleset :51:56:sed_subst:Successful substitution; pattern= matchpattern=^.* replacepattern= replaced pattern= :51:56:regxrule_subst_num_type:Match Type = none, Replace Type = none Input Type = unknown 00:51:56:regxrule_subst_num_plan:Match Plan = none, Replace Plan = none Input Plan = unknown Then the output shows the details of the translated number following "regxrule_profile_translate". 00:51:56:regxrule_profile_translate:xlt_number= xlt_type=unknown xlt_plan=unknown 00:51:56:regxrule_profile_translate:number= type=unknown plan=unknown

69 debug voice translation (Cont.) numbertype=redirect-called 00:51:56:regxrule_get_RegXrule:Invalid translation ruleset tag=0 In this example, because there was no called number or redirect number translation that was configured on the translation profile, corresponding errors were generated with a message that no match was found. 00:51:56:regxrule_profile_match:Error:ruleset for redirect-called number not found 00:51:56:regxrule_profile_translate:No match:number= type=unknown plan=unknown 00:51:56:regxrule_profile_translate:number= type=unknown plan=unknown numbertype=called 00:51:56:regxrule_get_RegXrule:Invalid translation ruleset tag=0 00:51:56:regxrule_profile_match:Error:ruleset for called number not found 00:51:56:regxrule_profile_translate:No match:number= type=unknown plan=unknown 00:51:56:regxrule_stack_push_RegXruleNumInfo:stack=0x63DECAF4; count=1 Following "regxrule_dp_translate", the output indicates that there is no translation profile for outgoing direction, and then it prints the numbers sent to the outgoing service provider interface (SPI). 00:51:56:regxrule_dp_translate:No profile found in peer for outgoing direction 00:51:56:regxrule_dp_translate:calling_number= calling_octet=0x0 called_number= called_octet=0x80 redirect_number= redirect_type= redirect_plan= :51:56:regxrule_stack_pop_RegXruleNumInfo:stack=0x63DECAF4; count=2 00:51:56:regxrule_stack_push_RegXruleNumInfo:stack=0x63DECAF4; count=1

70 Summary A dial plan defines how calls are interconnected and routed. Endpoint addressing assigns directory numbers to endpoints. Call routing and path selection define where a call is routed to and usually depends on the called party number. The digit manipulation feature ensures that numbers are presented in the correct format and is closely connected to call routing.

71 Summary Calling privileges define the destinations a user may dial. Call coverage ensures that incoming calls are not lost. Several factors must be considered when designing a scalable dial plan. Interworking with the PSTN requires appropriate call routing and digit manipulation.

72 Summary Digit manipulation based on a TON and numbering plan is required for ISDN networks. 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.