Cisco Unified Communications Manager with Cisco Unified Border Element [IOS 15.4(2)T1] using SIP

Transcription

1 Application Note Cox SIP Trunking: Cisco Unified Communications Manager with Cisco Unified Border Element [IOS 15.4(2)T1] using SIP September 22, 2014 Page 1 of 73

2 Table of Contents Introduction... 4 Network Topology... 5 System Components... 7 Hardware Components... 7 Software Requirements... 7 Features Supported... 7 Features Not Supported... 7 Caveats... 7 Configuration... 8 Configuring the Cisco Unified Border Element... 8 Network interface... 8 Global CUBE settings... 9 Media Passing through CUBE (media flow-through vs. media flow-around) Codecs Dial peer Call flow Configuring the Cisco Unified Communications Manager Cisco CallManager service parameter Offnet calls via Cox SIP Trunk Dialplan Acronyms Important Information Appendix A: Test Results Table of Figures Figure 1 Network Topology... 5 Figure 2: CUBE High Availability... 6 Figure 3: Outbound Voice Call Figure 4: Outboud Fax Call Figure 5 Service Parameter Figure 6 Service Parameter cont Figure 7 Service Parameter cont 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 Service Parameter cont Figure 17 Service Parameter cont Figure 18 Service Parameter cont Figure 19 Service Parameter cont Figure 20 Service Parameter cont Figure 21 Service Parameter cont Figure 22 SIP Trunk Security Profile Figure 23 SIP Profile Figure 24 SIP Profile Cont Figure 25: SIP Profile cont Figure 26 SIP Trunks List Figure 27 SIP Trunk to CUBE Figure 28 SIP Trunk to CUBE Cont Figure 29 SIP Trunk to CUBE Cont Figure 30 SIP Trunk to CUBE Cont Figure 31 Route Pattern Figure 32 Route Pattern for Voice Page 2 of 73

3 Figure 33 Route Pattern for Voice Figure 34 Route Pattern Unity Connection Figure 35 Route Pattern Unity Connection cont Figure 36 Route Pattern for Fax Figure 37 Route Pattern for Fax cont Page 3 of 73

4 Introduction Service Providers today, such as Cox, 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. Cox 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 Cox Session Border Controller(EdgeMarc), Cisco Unified Border Element (CUBE) can be used. The Cisco Unified Border Element provides demarcation, security, interworking and session control services for Cisco Unified Communications Manager connected to Cox IP network. This document assumes the reader is knowledgeable with the terminology and configuration of CUCM (Cisco Unified Communications Manager). Only configuration settings specifically required for Cox 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 (CUBE) for connectivity to Cox SIP trunking service. The deployment model covered in this application note is CPE (Cisco UCM ) to PSTN (Cox). Testing was performed in accordance to Cox generic SIP trunking test methodology and among features verified were basic calls, DTMF transport, Music on Hold, Semi-attendant and attendant transfers, call forward, conferences, hunt groups, call pickup, call park, and interoperability with Cisco Unity Connection The CUCM configuration detailed in this document is based on a lab environment with a simple dial-plan used to ensure proper interoperability between Cox 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 Cox SIP trunking network. This application note does not cover the use of calling search spaces (CSS) or partitions on Cisco Unified Communications Manager. To understand and learn how to apply CSS and partitions refer to the cisco.com link below: Page 4 of 73

5 Network Topology Basic Call Setup Figure 1 Network Topology Page 5 of 73

6 Figure 2: CUBE High Availability Page 6 of 73

7 System Components Hardware Components Cisco UCM and Unity Connection run on VMware ISR G router ISR 2951 router IP phones 9971(SIP), 8961(SIP) and 7975(SCCP)( please consult Features not supported for restrictions) Cisco Voice Gateways 224 Software Requirements Cisco Unified Communications Manager IOS 15.4(2)T1 for Cisco Unified Border Element IOS 15.1(3)T3 for VG224 Voice Gateways Cisco Unity Connection Features Supported Incoming and outgoing off-net calls using G711Ulaw (Cox only offer G711Ulaw) with 20ms packetization Call hold Call transfer (Semi-attendant and Attendant) Call conference Call forward (all, busy, no answer) Call park Hunt groups Calling line (number) identification presentation (CLIP) Calling line (number) identification restriction (CLIR) DTMF (RFC2833) Media flow-through on CUBE Fax(both G.711 and T.38) Account Code Authorization Code Features Not Supported Caveats Outbound SIP REFER with Replaces. Cisco UCM does not currently support generation of an outbound SIP REFER with Replaces message, Cisco IP phones used in this test do not support Blind transfer, only Semi-attendant and Attendant transfer were tested Cox Lab test network is closed network, certain test calls (international, 911, 411, etc.) get the announcement inside Cox lab network instead real destinations. Page 7 of 73

8 Configuration Configuring the Cisco Unified Border Element 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 LAN and WAN. interface GigabitEthernet0/0 description LAN to CUCM Clust33 ip address standby delay minimum 30 reload 60 standby version 2 standby 6 ip standby 6 priority 50 standby 6 preempt delay minimum 10 standby 6 track 1 decrement 20 standby 6 track 2 decrement 20 duplex auto speed auto interface GigabitEthernet0/1 ip address standby delay minimum 30 reload 60 standby version 2 standby 0 ip standby 0 priority 50 standby 0 preempt delay minimum 10 standby 0 name SB standby 0 track 1 decrement 20 standby 0 track 2 decrement 20 duplex auto speed auto Page 8 of 73

9 Global CUBE settings In order to enable CUBE IP2IP gateway functionality, following command has to be entered: voice service voip ip address trusted list ipv ipv ipv address-hiding mode border-element allow-connections sip to sip redundancy no supplementary-service sip refer fax protocol pass-through g711ulaw sip bind control source-interface GigabitEthernet0/1 bind media source-interface GigabitEthernet0/1 session refresh asserted-id pai privacy pstn early-offer forced midcall-signaling passthru sip-profiles 1 Explanation Command allow-connections sip to sip fax protocol asserted-id early-offer forced Description Allow IP2IP connections between two SIP call legs Specifies the fax protocol Specifies the type of privacy header in the outgoing SIP requests and response messages Enables SIP Delayed-Offer to Early-Offer globally Page 9 of 73

10 midcall-signaling passthru sip-profiles 1 Passes SIP messages from one IP leg to another IP leg Apply sip profile at global level Media Passing through CUBE (media flow-through vs. media flow-around) Default CUBE configuration enables CUBE to work in flow-through mode(this test use Flow-through mode). In order to enable flow-around mode, please perform the following actions: voice service voip media flow-around Codecs Cox offer only G.711ulaw codec for voice call, it allows codecs other than G.711ulaw but will only accept G.711ulaw. For customers using G.711 alaw codec: voice class codec 1 codec preference 1 g711ulaw Dial peer CUCM uses dial-peer to route the call based on the digit to route the call accordingly. incoming voice call to CUCM dial-peer voice 1 voip destination-pattern session protocol sipv2 session target ipv4: voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0 voice-class sip bind media source-interface GigabitEthernet0/0 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad Page 10 of 73

11 for outgoing calls to Cox ESBC dial-peer voice 10 voip destination-pattern 1[2-9]..[2-9]... session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 11 voip translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]..[2-9]... session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad Page 11 of 73

12 dial-peer voice 13 voip description **0 or 00 calls to Local or International Operator** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 0T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs3 signaling no vad Call flow In the sample configuration presented here, CUCM is provisioned with four-digit directory numbers corresponding to the last four DID digits. No digit manipulation is performed on the CUBE. For incoming PSTN calls, the CUBE presents the full ten-digit DID number to CUCM. The CUCM Translation Pattern strips all but the last four digits and routes the call based on those digits. Voice calls are routed to IP phones; fax calls are routed via a 4-digit route pattern to a SIP trunk that terminates on the VG224 CPE callers make outbound PSTN calls by dialing a 9 prefix followed by the destination number. For outbound fax calls from the analog fax endpoint, VG224 sends to Cisco UCM the DID with leading access code 9. A 9.@ route pattern strips the prefix and routes the call with the remaining digits via a SIP trunk terminating on the CUBE for Voice call or outbound Fax. Page 12 of 73

13 Figure 3: Outbound Voice Call Figure 4: Outboud Fax Call Configuration example The following configuration snippet contains a sample configuration of Cisco Unified Border Element with all parameters mentioned previously. Active CUBE: User Access Verification Username: cisco Password: COX_CUBE1#show run Building configuration... Current configuration : 8577 bytes Last configuration change at 14:16:03 UTC Fri Sep by cisco version 15.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption Page 13 of 73

14 service sequence-numbers hostname COX_CUBE1 boot-start-marker boot system flash c2951-universalk9-mz.spa t1.bin boot-end-marker logging buffered no logging rate-limit no logging console no logging monitor enable secret 4 skpgcy/xpea3wk8xoeswo7ugfanvwzxdeyxwhudjelk ipc zone default association 1 no shutdown protocol sctp local-port 5000 local-ip remote-port 5000 remote-ip no aaa new-model no ip domain lookup ip name-server ip cef no ipv6 cef Page 14 of 73

15 multilink bundle-name authenticated password encryption aes voice-card 0 dsp services dspfarm voice service voip ip address trusted list ipv ipv ipv address-hiding mode border-element allow-connections sip to sip redundancy no supplementary-service sip refer fax protocol pass-through g711ulaw modem passthrough nse codec g711ulaw sip bind control source-interface GigabitEthernet0/1 bind media source-interface GigabitEthernet0/1 session refresh asserted-id pai privacy pstn early-offer forced midcall-signaling passthru sip-profiles 1 Page 15 of 73

16 voice class codec 1 codec preference 1 g711ulaw voice class sip-profiles 1 request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request INVITE sip-header Contact modify ">" ";user=phone>" request REINVITE sip-header Contact modify ">" ";user=phone>" request INVITE sip-header To modify ">" ";user=phone>" request REINVITE sip-header To modify ">" ";user=phone>" request INVITE sip-header From modify ">" ";user=phone>" request REINVITE sip-header From modify ">" ";user=phone>" license udi pid CISCO2951/K9 sn FTX1509AJJV hw-module pvdm 0/0 username CISCO privilege 15 password 0 tekv1z10n redundancy inter-device scheme standby SB redundancy track 1 interface GigabitEthernet0/1 line-protocol track 2 interface GigabitEthernet0/0 line-protocol interface Embedded-Service-Engine0/0 Page 16 of 73

17 no ip address shutdown interface GigabitEthernet0/0 description LAN to CUCM Clust33 ip address standby delay minimum 30 reload 60 standby version 2 standby 6 ip standby 6 priority 60 standby 6 preempt delay minimum 10 standby 6 track 1 decrement 20 standby 6 track 2 decrement 20 duplex auto speed auto interface GigabitEthernet0/1 description WAN to edgemarc ip address standby delay minimum 30 reload 60 standby version 2 standby 0 ip standby 0 priority 60 standby 0 preempt delay minimum 10 standby 0 name SB standby 0 track 1 decrement 20 standby 0 track 2 decrement 20 duplex auto speed auto Page 17 of 73

18 interface GigabitEthernet0/2 no ip address shutdown duplex auto speed auto ip forward-protocol nd no ip http server no ip http secure-server ip route ip route nls resp-timeout 1 cpd cr-id 1 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 1 voip destination-pattern Page 18 of 73

19 session protocol sipv2 session target ipv4: voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0 voice-class sip bind media source-interface GigabitEthernet0/0 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 10 voip destination-pattern 1[2-9]..[2-9]... session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 11 voip translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]..[2-9]... session protocol sipv2 Page 19 of 73

20 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 13 voip description **0 or 00 calls to Local or International Operator** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 0T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs3 signaling no vad dial-peer voice 12 voip translation-profile outgoing PSTN_Outgoing Page 20 of 73

21 preference 1 destination-pattern 011T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 14 voip description **CCA*North American-10-Digit*Service Numbers** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]11 session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs3 signaling no vad Page 21 of 73

22 dial-peer voice 15 voip description **Dial Carrier Access Code** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]..[2-9]... session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte no voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs3 signaling no vad sip-ua sip-server ipv4: :5060 gatekeeper shutdown line con 0 line aux 0 line 2 no activation-character no exec transport preferred none transport output pad telnet rlogin lapb-ta mop udptn v120 ssh Page 22 of 73

23 stopbits 1 line vty 0 4 exec-timeout 0 0 login local transport input all scheduler allocate end Standby CUBE: User Access Verification Username: cisco Password: COX_CUBE2#show run Building configuration... Current configuration : 9982 bytes No configuration change since last restart version 15.4 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption hostname COX_CUBE2 boot-start-marker boot-end-marker Page 23 of 73

24 aqm-register-fnf logging buffered warnings enable secret 4 tnhtc92dxbhelxjyk8lwjrpv36s2i4ntxrpb4rfmfqy ipc zone default association 1 no shutdown protocol sctp local-port 5000 local-ip remote-port 5000 remote-ip no aaa new-model no ip domain lookup ip name-server ip cef no ipv6 cef multilink bundle-name authenticated password encryption aes crypto pki trustpoint TP-self-signed enrollment selfsigned subject-name cn=ios-self-signed-certificate Page 24 of 73

25 revocation-check none rsakeypair TP-self-signed crypto pki certificate chain TP-self-signed certificate self-signed B A D0609 2A F70D F30 2D F532D 53656C66 2D E65642D D E 170D A17 0D A F302D F532D53 656C662D E 65642D D F 300D0609 2A F70D D AC64 457DC FE0 A1AD6097 4F1358E1 3721B264 13A1D71B D711054C F27B071E 91464C54 EACBD884 DC242E08 1BC34A7E 1FA49C2F 4A130BD1 461AC476 BA1352B7 F54C E43E 1FF4824D 8A75A247 F4AB488A 3F9EFD9C 6CED7728 4CE96D86 B43594A1 6684B A 99F337D9 5C04D4D6 ECD7BA8C 1AEF A F D FF FF 301F D B70ED5 6EF1FA77 9D2F8B0B 644BF4DE BC 27301D D0E B70ED56E F1FA779D 2F8B0B64 4BF4DE BC27 300D0609 2A F70D E882FC60 E29C53FE 5A B1A C03D7 677F2A10 178A4A9A B83448B1 EFBC136A 4080D4FC 493C3CDB 623B6343 A3639AEB 2A7753B8 9DFB4C79 F3BF9E03 A3146AA0 11AA9FC1 9F E4D57CB 78413BD3 10C790EE CBBBE796 A8490BE1 D0524A DC8B 91E6A14C 6FAF8DB F 425B3B8C quit voice-card 0 dsp services dspfarm voice rtp send-recv Page 25 of 73

26 voice service voip ip address trusted list ipv ipv ipv address-hiding mode border-element allow-connections sip to sip redundancy no supplementary-service sip refer fax protocol pass-through g711ulaw modem passthrough nse codec g711ulaw sip bind control source-interface GigabitEthernet0/1 bind media source-interface GigabitEthernet0/1 session refresh asserted-id pai privacy pstn early-offer forced midcall-signaling passthru sip-profiles 1 voice class codec 1 codec preference 1 g711ulaw voice class sip-profiles 1 request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request INVITE sip-header Contact modify ">" ";user=phone>" Page 26 of 73

27 request REINVITE sip-header Contact modify ">" ";user=phone>" request INVITE sip-header To modify ">" ";user=phone>" request REINVITE sip-header To modify ">" ";user=phone>" request INVITE sip-header From modify ">" ";user=phone>" request REINVITE sip-header From modify ">" ";user=phone>" license udi pid C3900-SPE250/K9 sn FOC17426ADY hw-module pvdm 0/0 username cisco privilege 15 password D C40 redundancy inter-device scheme standby SB redundancy track 1 interface GigabitEthernet0/1 line-protocol track 2 interface GigabitEthernet0/0 line-protocol interface GigabitEthernet0/0 description LAN to CUCM Clust33 ip address standby delay minimum 30 reload 60 standby version 2 standby 6 ip standby 6 priority 50 standby 6 preempt delay minimum 10 Page 27 of 73

28 standby 6 track 1 decrement 20 standby 6 track 2 decrement 20 duplex auto speed auto interface GigabitEthernet0/1 ip address standby delay minimum 30 reload 60 standby version 2 standby 0 ip standby 0 priority 50 standby 0 preempt delay minimum 10 standby 0 name SB standby 0 track 1 decrement 20 standby 0 track 2 decrement 20 duplex auto speed auto interface GigabitEthernet0/2 no ip address shutdown duplex auto speed auto interface GigabitEthernet0/3 no ip address shutdown duplex auto speed auto Page 28 of 73

29 ip forward-protocol nd no ip http server no ip http secure-server ip route ip route nls resp-timeout 1 cpd cr-id 1 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 1 voip destination-pattern session protocol sipv2 session target ipv4: voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/0 voice-class sip bind media source-interface GigabitEthernet0/0 Page 29 of 73

30 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 10 voip destination-pattern 1[2-9]..[2-9]... session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 11 voip translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]..[2-9]... session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte Page 30 of 73

31 ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 13 voip description **0 or 00 calls to Local or International Operator** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 0T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs3 signaling no vad dial-peer voice 12 voip translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern 011T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced Page 31 of 73

32 voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs5 media ip qos dscp cs4 signaling no vad dial-peer voice 14 voip description **CCA*North American-10-Digit*Service Numbers** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]11 session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte voice-class sip early-offer forced voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte ip qos dscp cs3 signaling no vad dial-peer voice 15 voip description **Dial Carrier Access Code** translation-profile outgoing PSTN_Outgoing preference 1 destination-pattern [2-9]..[2-9]... session protocol sipv2 Page 32 of 73

33 session target sip-server voice-class codec 1 voice-class sip dtmf-relay force rtp-nte no voice-class sip early-offer forced dtmf-relay rtp-nte ip qos dscp cs3 signaling no vad sip-ua sip-server ipv4: :5060 gatekeeper shutdown line con 0 exec-timeout 0 0 password F41 logging synchronous login local line aux 0 line vty 0 4 exec-timeout 0 0 password F0725 logging synchronous login local transport input telnet ssh line vty 5 15 exec-timeout 0 0 logging synchronous Page 33 of 73

34 login local transport input telnet ssh scheduler allocate ntp server End Page 34 of 73

35 Configuring the Cisco Unified Communications Manager Cisco CallManager service parameter Go to System > Service Parameters.we leave all fields in the service parameter as default values for this test Figure 5 Service Parameter Page 35 of 73

36 Figure 6 Service Parameter cont. Page 36 of 73

37 Figure 7 Service Parameter cont. Page 37 of 73

38 Figure 8 Service Parameter cont. Page 38 of 73

39 Figure 9 Service Parameter cont. Page 39 of 73

40 Figure 10 Service Parameter cont. Page 40 of 73

41 Figure 11 Service Parameter cont. Page 41 of 73

42 Figure 12 Service Parameter cont. Page 42 of 73

43 Figure 13 Service Parameter cont. Page 43 of 73

44 Figure 14 Service Parameter cont. Page 44 of 73

45 Figure 15 Service Parameter cont. Page 45 of 73

46 Figure 16 Service Parameter cont. Page 46 of 73

47 Figure 17 Service Parameter cont. Page 47 of 73

48 Figure 18 Service Parameter cont. Page 48 of 73

49 Figure 19 Service Parameter cont. Page 49 of 73

50 Figure 20 Service Parameter cont. Page 50 of 73

51 Figure 21 Service Parameter cont. Page 51 of 73

52 Offnet calls via Cox SIP Trunk Off-net calls are served by SIP trunks configured between CUCM and Cox ESBC. Calls are routed via CUBE. SIP Trunk Security Profile Go to System > Security > SIP Trunk Security Profile and click on Add New. Figure 22 SIP Trunk Security Profile Parameter Value Description Incoming Transport Type TCP + UDP Outgoing Transport Type UDP SIP trunks to Cox ESBC 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. Page 52 of 73

53 SIP Profile SIP Profile will be later associated with the SIP trunk. Navigate to Device > Device Settings > SIP Profile and modify default SIP Profile by clicking on a Copy button in its row. Figure 23 SIP Profile Page 53 of 73

54 Figure 24 SIP Profile Cont. Page 54 of 73

55 Figure 25: SIP Profile cont. Page 55 of 73

56 Parameter Value Description Default MTP Telephony Event Payload Type 101 RFC2833 DTMF payload type Require SDP Inactive Exchange for Mid-Call Media Change SIP Rel1XX Options Checked Send PRACK for 1xx Messages Send SDP with Inactive when call on hold Enable Provisional Acknowlesgements (Reliable 100 messages) Ping Interval for In-service and Partially In-service Trunks (seconds) Ping Interval for Out-of-service Trunks (seconds) 300 OPTIONS message parameters- interval time 5 OPTIONS message parameters- interval time Page 56 of 73

57 SIP Trunk Configuration Create SIP trunks to Cox by navigating to Device > Trunk and clicking Add New button. Same apply to create Sip trunks to Cisco Unity Connection and VG224 Figure 26 SIP Trunks List Page 57 of 73

58 Figure 27 SIP Trunk to CUBE Page 58 of 73

59 Figure 28 SIP Trunk to CUBE Cont. Page 59 of 73

60 Figure 29 SIP Trunk to CUBE Cont. Page 60 of 73

61 Figure 30 SIP Trunk to CUBE Cont. Parameter Value Description Device Name Cox Name for the trunk Device Pool Default 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 Redirecting Diversion Header Delivery - outbound Checked Adding Diversion Header for calls outband from site Destination Address Virtual IP address of the CUBE SIP Trunk Security Profile Non Secure SIP Trunk Profile SIP Trunk Security Profile configured earlier SIP Profile Cox SIP Profile SIP Profile configured earlier DTMF Signaling Method RFC 2833 RFC 2833 is supported for DTMF transport to/from Cox Reset the trunk after the configuration is completed. Apply same procedure to create SIP trunks to Cisco Unity Connection and VG224 Page 61 of 73

62 Dialplan Route Pattern configuration Route patterns are configured as below, Cisco IP phone dial 9+10digits number to access PSTN via CUBE, 9 is removed before send to CUBE; for FAX call, Access Code 9 is used at VG224, 9 is removed at UCM and 10digits number is send to CUBE to Cox network. Incoming fax call to 3710 will send to VG is the Pilot Number for Voice mail to Unity Connection. Navigate to Call Routing > Route/Hunt > Route Pattern and press Add New button to create Route Patterns Figure 31 Route Pattern Page 62 of 73

63 Figure 32 Route Pattern for Voice Page 63 of 73

64 Figure 33 Route Pattern for Voice Page 64 of 73

65 Figure 34 Route Pattern Unity Connection Page 65 of 73

66 Figure 35 Route Pattern Unity Connection cont. Page 66 of 73

67 Figure 36 Route Pattern for Fax Page 67 of 73

68 Figure 37 Route Pattern for Fax cont. Setting Value Description Route Pattern for Voice call and for fax call Specify appropriate Route Pattern Gateway/Route List COX SIP Trunk name configured earlier Require Forced Authorization Code Require Client Matter Code Checked when doing Authorization Code test Check when doing Account Code test Specify if Authorization Code required when make call through this Route Pattern Specify if Account Code required when make call through this Route Pattern Calling Party Transform mask XXXX Specify the Calling Line ID for outgoing call through this Route Pattern Discard Digits PreDot for RP specifies how to modify digit before they are sending to Cox ESBC Page 68 of 73

69 Acronyms Acronym CPE CUBE CUCM 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 69 of 73

70 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 70 of 73

71 Appendix A: Test Results Application Note SP_SIP_master_testp lan_cucm_10.5.xls Page 71 of 73

72 Corporate Headquarters European Headquarters Americas Headquarters Asia Pacific Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA USA Tel: NETS (6387) Fax: Cisco Systems International BV Haarlerbergpark Haarlerbergweg CH Amsterdam The Netherlands www-europe.cisco.com Tel: Fax: Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA USA Tel: Fax: Cisco Systems, Inc. Capital Tower 168 Robinson Road #22-01 to #29-01 Singapore Tel: Fax: Cisco Systems has more than 200 offices in the following countries and regions. Addresses, phone numbers, and fax numbers are listed on the Cisco Web site at Argentina Australia Austria Belgium Brazil Bulgaria Canada Chile China PRC Colombia Costa Rica Croatia Czech Republic Denmark Dubai, UAE Finland France Germany Greece Hong Kong SAR Hungary India Indonesia Ireland Israel Italy Japan Korea Luxembourg Malaysia Mexico The Netherlands New Zealand Norway Peru Philippines Poland Portugal Puerto Rico Romania Russia Saudi Arabia Scotland Singapore Slovakia Slovenia South Africa Spain Sweden Switzerland Taiwan Thailand Turkey Ukraine United Kingdom United States Venezuela Vietnam Zimbabwe 2008 Cisco Systems, Inc. All rights reserved. CCENT, Cisco Lumin, Cisco Nexus, the Cisco logo and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn is a service mark of Cisco Systems, Inc.; and Access Registrar, Aironet, BPX, Catalyst, CCDA, CCDP, CCVP, CCIE, CCIP, CCNA, CCNP, CCSP, Cisco, the Cisco Certified Internetwork Expert logo, Cisco IOS, Cisco Press, Cisco Systems, Cisco Systems Capital, the Cisco Systems logo, Cisco Unity, EtherFast, EtherSwitch, Fast Step, Follow Me Browsing, FormShare, GigaDrive, HomeLink, Internet Quotient, IOS, iphone, iq Expertise, the iq logo, iq Net Readiness Scorecard, iquick Study, LightStream, Linksys, MeetingPlace, MGX, Networking Academy, Network Registrar, Packet, PIX, ProConnect, ScriptShare, SMARTnet, StackWise, The Fastest Way to Increase Your Internet Quotient, and TransPath Page 72 of 73

73 are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries. All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0705R) Printed in the USA Page 73 of 73