Cisco Unified Communications Manager with Cisco Unified Border Element [CUBE IOS-XE 3.15] on ISR 4K using SIP

Transcription

1 Verizon SIP Trunking: Cisco Unified Communications Manager with Cisco Unified Border Element [CUBE IOS-XE 3.15] on ISR 4K using SIP JULY 24, 2015 Page 1 of 63

2 Table of Contents Introduction... 3 Network Topology... 4 System Components... 5 Hardware Requirements... 5 Software Requirements... 6 Features... 6 Features Supported... 6 Features Not Supported... 6 Caveats... 6 Configuration... 6 Configuring Cisco Unified Border Element... 6 Network Interface... 7 Global Cisco UBE settings... 7 Media Passing through Cisco UBE (media flow-through vs. media flow-around)... 9 Codecs... 9 Dial peer... 9 Configuration example Configuring Cisco Unified Communications Manager... Error Bookmark not defined. Cisco UCM Version Offnet calls via Verizon SIP Trunk Dialplan Acronyms Important Information Table of Figures Figure 1 Network Topology... 5 Figure 2 Cisco UCM version Figure 3 Service Parameter Figure 4 Service parameter(cont...) Figure 5 Service parameter(cont...) Figure 6 Service parameter(cont...) Figure 7 Service parameter(cont...) Page 2 of 63

3 Figure 8 Service parameter(cont...) Figure 9 Service parameter(cont...) Figure 10 Service parameter(cont...) Figure 11 Service parameter(cont...) Figure 12 Service parameter(cont...) Figure 13 Service parameter(cont...) Figure 14 Service parameter(cont...) Figure 15 Service parameter(cont...) Figure 16 SIP Trunk Security Profile Figure 17 SIP Profile Figure 18 SIP Profile (cont ) Figure 19 SIP Profile (cont ) Figure 20 SIP Profile (cont ) Figure 21 SIP Trunk list Figure 22 SIP Trunk to CUBE Figure 23 SIP Trunk to CUBE (cont ) Figure 24 SIP Trunk to CUBE (cont ) Figure 25 SIP Trunk to CUBE (cont ) Figure 26 Route Pattern list Figure 27 Route Pattern for Voice Figure 28 Route Pattern for Voice(cont ) Introduction Page 3 of 63

4 Service Providers today, such as Verizon, are offering alternative methods to connect to the PSTN via their IP network. Most of these services utilize SIP as the primary signaling method and centralized IP to TDM POP gateways to provide on-net and off-net services. Verizon is a service provider offering that allows connection to the PSTN and may offer the end customer a viable alternative to traditional PSTN connectivity. A demarcation device between these services and customer owned services is recommended. As an intermediary device between Cisco Unified Communications Manager and Verizon network, Cisco Unified Border Element (Cisco UBE) [IOS- XE Release 3.15, CUBE ] can be used. The Cisco Unified Border Element provides demarcation, security, interworking and session control services for Cisco Unified Communications Manager connected to Verizon IP network. This document assumes the reader is knowledgeable with the terminology and configuration of Cisco UCM (Cisco Unified Communications Manager). Only configuration settings specifically required for Verizon interoperability are presented. Feature configuration and most importantly the dial plan are customer specific and need individual approach. This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) and Cisco Unified Border Element (Cisco UBE) on ISR 4K for connectivity to Verizon SIP trunking service. The deployment model covered in this application note is CPE (Cisco UCM ) to PSTN (Verizon). Testing was performed in accordance to Verizon generic SIP trunking test methodology and among features verified were basic calls, DTMF transport, Music on Hold (MOH), unattended and attended transfers, call forward, conferences and interoperability with Cisco Unity Connection (CUC) The Cisco UCM configuration detailed in this document is based on a lab environment with a simple dial-plan used to ensure proper interoperability between Verizon SIP network and Cisco Unified Communications. The configuration described in this document details the important configuration settings to have enabled for interoperability to be successful and care must be taken by the network administrator deploying Cisco UCM to interoperate to Verizon SIP trunking network. This application note does not cover the use of Calling Search Spaces (CSS) or partitions on Cisco UCM. To understand and learn how to apply CSS and partitions refer to the cisco.com link below: Network Topology Page 4 of 63

5 Figure 1 Network Topology Cisco IP Phones 7975 and 7965 phones are the devices primarily used throughout the testing to place or receive calls. VentaFax Soft Client is used to perform all fax related scenarios. The fax client is connected to SIP Gateway via FXS port which in turn communicates with Cisco UCM over SIP. System Components Hardware Requirements Cisco UCS-C240-M3S VMWare host running ESXi 5.5 Standard Page 5 of 63

6 Cisco UCM running on VMware Cisco ISR 4431 Cube Cisco 2851 Fax Gateway IP phones 7975(SCCP) and 7965 (SCCP). Software Requirements Cisco Unified Communications Manager IOS-XE version 15.5(2)S, XE3.15 for ISR 4431 IOS 15.0(1)XA3 for Cisco 2851 Fax Gateway Features Features Supported Incoming and outgoing off-net calls using G729 Call hold Call transfer (unattended and attended) Call forward (all, busy and no answer) Calling Line (number) Identification Presentation (CLIP) Calling Line (number) Identification Restriction (CLIR) DTMF (RFC2833) Media flow-through on Cisco UBE Fax (G.711 and T.38) Features Not Supported Caveats Cisco IP phones used in this test do not support blind transfer. Unattended and attended transfer scenarios were tested Cisco IOS Enhanced Conference Bridge is used to establish conference with PSTN using G729 codec CLID is not updated on PSTN phones for transfer scenarios Testing is done with only one PBX Configuration Configuring Cisco Unified Border Element Page 6 of 63

7 Network Interface Configure Ethernet IP address and sub interface. The IP address and VLAN encapsulation used are for illustration only, the actual IP address can vary. For SIP trunks two IP addresses must be configured - for LAN and WAN. interface GigabitEthernet0/0/0 ip address negotiation auto interface GigabitEthernet0/0/1 no ip address shutdown media-type rj45 negotiation auto interface GigabitEthernet0/0/2 description LAN interface ip address negotiation auto redundancy rii 1 redundancy group 1 ip exclusive interface GigabitEthernet0/0/3 description WAN interface ip address negotiation auto redundancy rii 2 redundancy group 1 ip exclusive Page 7 of 63

8 Global Cisco UBE settings In order to enable Cisco UBE IP2IP gateway functionality, following command has to be entered: voice service voip no ip address trusted authenticate address-hiding mode border-element allow-connections sip to sip redundancy-group 1 no supplementary-service sip handle-replaces fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip session refresh asserted-id pai early-offer forced no silent-discard untrusted midcall-signaling passthru privacy-policy passthru privacy-policy send-always g729 annexb-all Explanation Command allow-connections sip to sip fax protocol Description Allow IP2IP connections between two SIP call legs Specifies the fax protocol Page 8 of 63

9 asserted-id early-offer forced midcall-signaling passthru Specifies the type of privacy header in the outgoing SIP requests and response messages Enables SIP Delayed-Offer to Early-Offer globally Passes SIP messages from one IP leg to another IP leg Media Passing through Cisco UBE (media flow-through vs. media flow-around) Default Cisco UBE configuration enables Cisco UBE to work in flow-through mode (this test use Flowthrough mode). In order to enable flow-around mode, please perform the following actions: voice service voip media flow-around Codecs G729 is used as the preferred codec for this testing. voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw Dial peer Cisco UBE uses dial-peer to route the call accordingly based on the digits. dial-peer voice 100 voip description incoming from VZ session protocol sipv2 incoming called-number voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/3 voice-class sip bind media source-interface GigabitEthernet0/0/3 dtmf-relay rtp-nte fax rate Page 9 of 63

10 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 101 voip description "incoming call from VZ facing IP-PBX" destination-pattern session protocol sipv2 session target ipv4: :5060 session transport udp voice-class codec 1 voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0/2 voice-class sip bind media source-interface GigabitEthernet0/0/2 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 200 voip description outgoing call to VZ facing VZ network huntstop destination-pattern [2-9]T session protocol sipv2 session target ipv4: :5072 voice-class codec 1 Page 10 of 63

11 voice-class sip rel1xx supported "100rel" voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0/3 voice-class sip bind media source-interface GigabitEthernet0/0/3 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 201 voip description incoming from IP-PBX session protocol sipv2 incoming called-number [2-9]T voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/2 voice-class sip bind media source-interface GigabitEthernet0/0/2 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad Configuration example The following configuration snippet contains a sample configuration of Cisco UBE with all parameters mentioned previously. Page 11 of 63

12 Active Cisco UBE: User Access Verification Username: cisco Password: % Password expiration warning Cisco Configuration Professional (Cisco CP) is installed on this device and it provides the default username "cisco" for one-time use. If you have already used the username "cisco" to login to the router and your IOS image supports the "one-time" user option, then this username has already expired. You will not be able to login to the router with this username after you exit this session. It is strongly suggested that you create a new username with a privilege level of 15 using the following command. username <myuser> privilege 15 secret 0 <mypassword> Replace <myuser> and <mypassword> with the username and password you want to use CISCO_4K_ROUTER1#sh running-config Building configuration... Current configuration : 5951 bytes Page 12 of 63

13 Last configuration change at 22:52:47 UTC Thu Jul by cisco version 15.5 service config service timestamps debug datetime msec service timestamps log datetime msec no platform punt-keepalive disable-kernel-core boot-start-marker boot system flash bootflash:/isr4400-universalk s s-std.spa.bin boot-end-marker hostname CISCO_4K_ROUTER1 boot-start-marker boot-end-marker vrf definition Mgmt-intf address-family ipv4 exit-address-family address-family ipv6 exit-address-family vrf definition mgmt-intf logging queue-limit Page 13 of 63

14 logging buffered logging rate-limit enable secret 5 $1$EWC0$6QpPfaUZ/L04rhx3qQlj7/ enable password tekv1z10n aaa new-model aaa session-id common ipc zone default association 1 no shutdown no ip domain lookup ip domain name tekvizion.com subscriber templating multilink bundle-name authenticated cts logging verbose voice service voip no ip address trusted authenticate address-hiding mode border-element allow-connections sip to sip Page 14 of 63

15 redundancy-group 1 no supplementary-service sip handle-replaces fax protocol pass-through g711ulaw sip session refresh asserted-id pai early-offer forced no silent-discard untrusted midcall-signaling passthru privacy-policy passthru privacy-policy send-always g729 annexb-all voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class codec 2 codec preference 1 g711ulaw codec preference 2 g729r8 bytes 30 voice class sip-profiles 1 request REFER sip-header Referred-By remove license udi pid ISR4431/K9 sn FOC18261KJL license boot level appxk9 license boot level uck9 Page 15 of 63

16 spanning-tree extend system-id username cisco privilege 15 password 0 tekv1z10n redundancy mode none application redundancy group 1 name voice-b2bha priority 100 failover threshold 75 timers delay 30 reload 60 control GigabitEthernet0/0/0 protocol 1 data GigabitEthernet0/0/0 track 1 shutdown track 2 shutdown vlan internal allocation policy ascending track 1 interface GigabitEthernet0/0/2 line-protocol track 2 interface GigabitEthernet0/0/3 line-protocol ip tftp source-interface GigabitEthernet0 ip ssh version 1 interface GigabitEthernet0/0/0 ip address Page 16 of 63

17 negotiation auto interface GigabitEthernet0/0/1 no ip address shutdown media-type rj45 negotiation auto interface GigabitEthernet0/0/2 description LAN interface ip address negotiation auto redundancy rii 1 redundancy group 1 ip exclusive interface GigabitEthernet0/0/3 description WAN interface ip address negotiation auto redundancy rii 2 redundancy group 1 ip exclusive interface GigabitEthernet0 vrf forwarding Mgmt-intf no ip address shutdown negotiation auto Page 17 of 63

18 interface Vlan1 no ip address shutdown ip forward-protocol nd no ip http server no ip http secure-server ip route ip route ip route control-plane mgcp behavior rsip-range tgcp-only mgcp behavior comedia-role none mgcp behavior comedia-check-media-src disable mgcp behavior comedia-sdp-force disable mgcp profile default dial-peer voice 100 voip description incoming from VZ session protocol sipv2 incoming called-number Page 18 of 63

19 voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/3 voice-class sip bind media source-interface GigabitEthernet0/0/3 dtmf-relay rtp-nte fax-relay ecm disable fax rate disable fax protocol pass-through g711ulaw no vad dial-peer voice 101 voip description "incoming call from VZ facing IP-PBX" destination-pattern session protocol sipv2 session target ipv4: :5060 session transport udp voice-class codec 1 voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0/2 voice-class sip bind media source-interface GigabitEthernet0/0/2 dtmf-relay rtp-nte fax-relay ecm disable fax rate disable fax protocol pass-through g711ulaw no vad Page 19 of 63

20 dial-peer voice 200 voip description outgoing call to VZ facing VZ network huntstop destination-pattern [2-9]T session protocol sipv2 session target ipv4: :5072 voice-class codec 1 voice-class sip rel1xx supported "100rel" voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0/3 voice-class sip bind media source-interface GigabitEthernet0/0/3 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 201 voip description incoming from IP-PBX session protocol sipv2 incoming called-number [2-9]T voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/2 voice-class sip bind media source-interface GigabitEthernet0/0/2 dtmf-relay rtp-nte fax rate Page 20 of 63

21 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad gateway media-inactivity-criteria all timer receive-rtcp 5 timer receive-rtp sip-ua no remote-party-id retry invite 2 timers expires connection-reuse line con 0 logging synchronous stopbits 1 line aux 0 stopbits 1 line vty 0 session-timeout 90 exec-timeout no activation-character logging synchronous transport preferred ssh Page 21 of 63

22 transport input all stopbits 1 line vty 1 4 exec-timeout logging synchronous transport input all End Standby Cisco UBE: CISCO_4K_ROUTER2#sh running-config Current configuration : 5676 bytes version 15.5 service timestamps debug datetime msec service timestamps log datetime msec no platform punt-keepalive disable-kernel-core hostname CISCO_4K_ROUTER2 boot-start-marker boot system flash bootflash:/isr4400-universalk s s-std.spa.bin boot-end-marker vrf definition Mgmt-intf address-family ipv4 Page 22 of 63

23 exit-address-family address-family ipv6 exit-address-family enable secret 5 $1$zQRB$CCbzfD1aYzk3kPvzAm2KU0 enable password cisco aaa new-model aaa session-id common no ip domain lookup ip domain name tekvizion.com subscriber templating multilink bundle-name authenticated cts logging verbose voice service voip no ip address trusted authenticate address-hiding mode border-element allow-connections sip to sip redundancy-group 1 no supplementary-service sip handle-replaces fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip Page 23 of 63

24 session refresh asserted-id pai early-offer forced no silent-discard untrusted midcall-signaling passthru privacy-policy passthru privacy-policy send-always g729 annexb-all voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class codec 2 codec preference 1 g711ulaw codec preference 2 g729r8 bytes 30 voice class sip-profiles 1 request REFER sip-header Referred-By remove license udi pid ISR4431/K9 sn FOC license boot level appxk9 license boot level uck9 spanning-tree extend system-id username cisco privilege 15 secret 5 $1$AGR7$e7pQx6UI0be3bzRbc0lr81 Page 24 of 63

25 Page 25 of 63

26 redundancy mode none application redundancy group 1 name voice-b2bha priority 100 failover threshold 75 timers delay 30 reload 60 control GigabitEthernet0/0/0 protocol 1 data GigabitEthernet0/0/0 track 1 shutdown track 2 shutdown vlan internal allocation policy ascending track 1 interface GigabitEthernet0/0/2 line-protocol track 2 interface GigabitEthernet0/0/3 line-protocol ip tftp source-interface GigabitEthernet0 interface GigabitEthernet0/0/0 ip address negotiation auto interface GigabitEthernet0/0/1 no ip address shutdown Page 26 of 63

27 negotiation auto interface GigabitEthernet0/0/2 ip address negotiation auto redundancy rii 1 redundancy group 1 ip exclusive interface GigabitEthernet0/0/3 description Wan Interface ip address negotiation auto redundancy rii 2 redundancy group 1 ip exclusive interface GigabitEthernet0 vrf forwarding Mgmt-intf no ip address shutdown negotiation auto interface Vlan1 no ip address shutdown ip forward-protocol nd no ip http server Page 27 of 63

28 no ip http secure-server ip route ip route ip route control-plane mgcp behavior rsip-range tgcp-only mgcp behavior comedia-role none mgcp behavior comedia-check-media-src disable mgcp behavior comedia-sdp-force disable mgcp profile default dial-peer voice 100 voip description incoming from VZ session protocol sipv2 incoming called-number voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/3 voice-class sip bind media source-interface GigabitEthernet0/0/3 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 101 voip Page 28 of 63

29 description "incoming call from VZ facing IP-PBX" destination-pattern session protocol sipv2 session target ipv4: :5060 session transport udp voice-class codec 1 voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0/2 voice-class sip bind media source-interface GigabitEthernet0/0/2 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 200 voip description outgoing call to VZ facing VZ network huntstop destination-pattern [2-9]T session protocol sipv2 session target ipv4: :5072 voice-class codec 1 voice-class sip rel1xx supported "100rel" voice-class sip asserted-id pai voice-class sip privacy-policy passthru voice-class sip early-offer forced Page 29 of 63

30 voice-class sip bind control source-interface GigabitEthernet0/0/3 voice-class sip bind media source-interface GigabitEthernet0/0/3 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad dial-peer voice 201 voip description incoming from IP-PBX session protocol sipv2 incoming called-number [2-9]T voice-class codec 1 voice-class sip bind control source-interface GigabitEthernet0/0/2 voice-class sip bind media source-interface GigabitEthernet0/0/2 dtmf-relay rtp-nte fax rate fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none no vad gateway media-inactivity-criteria all timer receive-rtcp 5 timer receive-rtp sip-ua no remote-party-id retry invite 2 Page 30 of 63

31 timers expires connection-reuse line con 0 stopbits 1 line aux 0 stopbits 1 line vty 0 session-timeout 90 exec-timeout password tekv1z10n no activation-character logging synchronous transport preferred ssh transport input all stopbits 1 line vty 1 4 exec-timeout password tekv1z10n logging synchronous transport input all end Page 31 of 63

32 Page 32 of 63

33 Configuring Cisco Unified Communications Manager Cisco UCM Version Figure 2 Cisco UCM version Cisco Call Manager Service Parameters Page 33 of 63

34 Navigation path: System-> Service Parameters Select Server* = clus28pubsub CUCM Voice/Video (Active) Select Service*= Cisco CallManager (Active) Select Duplex Streaming Enabled* = True Figure 3 Service Parameter Page 34 of 63

35 Figure 4 Service parameter (cont...) Page 35 of 63

36 Figure 5 Service parameter (cont...) Page 36 of 63

37 Figure 6 Service parameter (cont...) Page 37 of 63

38 Figure 7 Service parameter (cont...) Page 38 of 63

39 Figure 8 Service parameter (cont...) Page 39 of 63

40 Figure 9 Service parameter (cont...) Page 40 of 63

41 Figure 10 Service parameter (cont...) Page 41 of 63

42 Figure 11 Service parameter (cont...) Page 42 of 63

43 Figure 12 Service parameter (cont...) Page 43 of 63

44 Figure 13 Service parameter (cont...) Page 44 of 63

45 Figure 14 Service parameter (cont...) Page 45 of 63

46 Figure 15 Service parameter (cont...) Page 46 of 63

47 Offnet calls via Verizon SIP Trunk Off-net calls are served by SIP trunks configured between Cisco UCM and Verizon network and Calls are routed via Cisco UBE. SIP Trunk Security Profile Navigation Path: System -> Security -> SIP Trunk Security Profile Figure 16 SIP Trunk Security Profile Parameter Value Description Page 47 of 63

48 Incoming Transport Type TCP + UDP Outgoing Transport Type UDP SIP trunks to Verizon should use UDP as a transport protocol for SIP. This is configured using SIP Trunk Security profile, which is later assigned to the SIP trunk itself. SIP Profile Configuration SIP Profile will be later associated with the SIP trunk. Navigation Path: Device-> Device Settings-> SIP Profile Figure 17 SIP Profile Page 48 of 63

49 Figure 18 SIP Profile (cont ) Page 49 of 63

50 Figure 19 SIP Profile (cont ) Page 50 of 63

51 Figure 20 SIP Profile (cont ) Parameter Value Description Default MTP Telephony Event Payload Type 101 RFC2833 DTMF payload type SIP Rel1XX Options Send PRACK for 1xx Messages Enable Provisional Acknowledgements (Reliable 100 messages) Ping Interval for In-service and Partially In-service Trunks (seconds) Ping Interval for Out-ofservice Trunks (seconds) 60 OPTIONS message parameters- interval time 120 OPTIONS message parameters- interval time Page 51 of 63

52 SIP Trunk Configuration Create SIP trunks to Cisco UBE. Create SIP trunks similarly to Cisco Unity Connection and Cisco Fax Gateway. Navigation: Device-> Trunk Figure 21 SIP Trunk list Page 52 of 63

53 Figure 22 SIP Trunk to CUBE Page 53 of 63

54 Figure 23 SIP Trunk to CUBE (cont ) Page 54 of 63

55 Figure 24 SIP Trunk to CUBE (cont ) Page 55 of 63

56 Figure 25 SIP Trunk to CUBE (cont ) Page 56 of 63

57 Parameter Value Description Device Name VERIZON Name for the trunk Device Pool G729_pool Default Device Pool is used for this trunk Media Resource Group List MRGL_MTP MRG with resources:ann, CFB, MOH and MTP Significant Digits 4 4 digits Extension for all CPE phones Destination Address Virtual IP address of the Cisco UBE SIP Trunk Security Profile SIP Profile Verizon Non Secure SIP Trunk Profile Standard SIP Profile w/early media disabled SIP Trunk Security Profile configured earlier SIP Profile configured earlier Dialplan Page 57 of 63

58 Route Pattern Configuration Navigation: Call Routing-> Route/Hunt-> Route Pattern Route patterns are configured as below, Cisco IP phone dial digits number to access PSTN via Cisco UBE, 9 is removed before send to Cisco UBE; for FAX call, Access Code 9 is used at Cisco Fax gateway, 9 is removed at Cisco UCM and 10 digits number is send to Cisco UBE to Verizon network. Figure 26 Route Pattern list Page 58 of 63

59 Figure 27 Route Pattern for Voice Page 59 of 63

60 Figure 28 Route Pattern for Voice (cont ) Setting Value Description Route Pattern 9.@ for Voice call, Specify appropriate Route Pattern Gateway/Route List Verizon for Route Pattern 9.@ SIP Trunk name configured earlier Numbering Plan NANP for Route Pattern 9.@ North American Numbering Plan Call Classification Offnet for Route Pattern 9.@ a Restrict the transferring of an external call to an external device Discard Digits PreDot for Route Pattern 9.@ Specifies how to modify digit before they are sending to Verizon network. Page 60 of 63

61 Acronyms Acronym CPE Cisco UBE Cisco UCM MTP POP PSTN ESBC SCCP SIP Definitions Customer Premise Equipment Cisco Unified Border Element Cisco Unified Communications Manager Media Termination Point Point of Presence Public Switched Telephone Network Enterprise Session Border Controller Skinny Client Control Protocol Session Initiation Protocol Page 61 of 63

62 Important Information THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES Page 62 of 63

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