Application Note. Note: Testing was conducted in Verizon lab. May 10, Initial Version Table of Contents

Transcription

1 Application Note Verizon IP Trunking and IP Contact Center Services: Connecting Cisco Unified Communications Manager with Cisco Unified Border Element 8.5 (Enterprise Edition) May 10, Initial Version Table of Contents Introduction... 2 Verizon IP Trunking Overview... 2 Verizon IPCC Overview... 2 Network Topology... 3 System Components... 3 Hardware Components... 3 Software Requirements... 3 Sample Bill of Materials... 4 Features and Known Limitations... 4 Features Supported (IP Trunking)... 4 Known Limitations (IP Trunking)... 4 Features Supported (IPCC)... 5 Known Limitations (IPCC)... 5 Cisco UBE Features Roadmap... 5 CISCO UCM 8.X SIP Trunk Deployment Considerations... 6 Call Flow Overview... 6 Outbound Call Flows... 6 Inbound Call Flows... 7 Failover... 7 Known Issues... 7 Inbound Call Issues... 7 New Security Operation in Cisco IOS T... 9 Redirected Dialed Number Identification Service and Diversion Header... 9 RDNIS Configuration in Cisco Unified Communications Manager Administration CISCO UCM Administrator>Device>Device Settings>SIP Profile CISCO UCM Administrator>Device>Device Settings>SIP Profile Communications Manager Configuration Media Resource Group List Media Resource Group CODEC Selection using Device Pools and Regions Clusterwide Parameters (System- Location and Region) List of Device Pools and the associated Regions List of Phones and ATA Devices SIP Trunk Configuration Route Group Configuration Route List for Voice Route List Details for Voice Route List for FAX Route List Details for FAX Route Plan report for Voice and FAX Offnet calls CISCO UBE Example Configuration (North America) Page 1 of 52

2 EMEA Configuration EMEA CISCO UCM Configuration EMEA CISCO UBE dial-peer Configuration IPCC Configuration IPCC CISCO UCM Configuration IPCC CISCO UBE dial-peer Configuration Troubleshooting References Acronyms Important Information Introduction This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 8.0 and Cisco Unified Border Element (Cisco UBE) Enterprise Edition 8.5 for connectivity to Verizon s IP trunking service. The deployment model covered in this application note utilizes Verizon s Private IP (commercial MPLS network) to access Verizon IP Trunking. Supplemental guidelines are also included for using Verizon IP trunking to interface to their IP-based Contact Center Service or IPCC. Please note that in the context of this document, IPCC refers to a cloud-based Contact Center product from Verizon, and should not be confused with a Cisco product. Additional supplemental guidelines are provided for an EMEA configuration. Testing was performed in accordance with the test plans for the Verizon IP trunking (US and EMEA), and IP Contact Center services. All features were verified.although this document does not detail the results of the testing performed it provides the essential configurations required for SIP interoperability with Cisco UCM/Cisco UBE and the Verizon IP Trunking and IPCC services. Verizon IP Trunking Overview Verizon IP trunking services simplify management of your network and can help drive operational efficiencies. They do this by consolidating your voice services onto a SIP-based VoIP network, thereby optimizing your data IP network, and controlling costs associated with maintaining traditional TDM local lines, trunks, and dedicated PRI circuits. Verizon also offer a native IP Trunking option that provides a SIP trunk directly to your IPPBX, and an IP Integrated Access option that leverages a gateway device so you can interface with legacy Key or PBX systems. And, Verizon s latest Burstable Enterprise Shared Trunking (BEST) feature enhancement allows you to share all your voice trunking resources across your enterprise and lets you use idle trunk capacity in one location to accommodate a traffic increase in another location. BEST helps control costs, as fewer concurrent calls need to be purchased at each location and resources can be shared to provide time of day benefits and peak usage management. Verizon IPCC Overview Verizon VoIP Inbound is a component of the IP Contact Center (IPCC) portfolio of internetworking services, which tightly couples signaling and functionality from the Advanced Toll Free and IP networks to deliver the intelligent routing and call treatment required by contact centers. The IPCC services are network-based and include IP Interactive Voice Response (IVR) in addition to VoIP Inbound. VoIP Inbound is standards-compliant and provides single-call service that allows PSTN-originated Toll Free calls to seamlessly terminate and transfer to a SIP or TDM endpoints, without call re-originations that tie up CPE port capacity. VoIP Inbound includes advanced toll free features Page 2 of 52

3 including automatic ISDN User Part and SIP Error overflow for reliable termination to SIP or TDM devices anywhere; and, when combined with IP IVR, supports customer-driven pre/post call routing and/or call treatment and queuing for customers using Cisco ICM Network Topology Figure 1. Typical Reference Network System Components Hardware Components CISCO UBE IOS version T2. Primary and Secondary CISCO UBE routers are used for high availability. Cisco Unified Border Element is an integrated Cisco IOS Software application that runs on various hardware platforms, for more details: Packet Voice Data Module (PVDM). You will need to install DSP modules on a supported ISR platform if you require MTP, Transcoding or Conference Bridge resources. These DSP resources are co-resident on the CISCO UBE routers in our lab configuration. CISCO UCM cluster with (2) Cisco MCS 7800 Series servers (Cisco Unified Communications Manager) Cisco Unified IP Phones Analog Telephony Adapter for FAX, modem, or analog phones Ethernet Switch WAN router used to terminate the Verizon MPLS network Software Requirements Cisco Unified Call Manager Cisco Unified Border Element CISCO UBE running IOS version T2 Page 3 of 52

4 Sample Bill of Materials Product Description Quantity MCS7835I3-K9-CMD1 Unified CM I3 Appliance 2 CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 4 C2921-VSEC-CUBE/K9 C2921 VSEC CUBE Bundle, PVDM3-32, UC SEC Lic, FL-CUBEE-25 2 S29UK9-1512T Cisco IOS UNIVERSAL 2 CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 2 WS-C2960G-24TC-L Catalyst /100/1000, 4 T/SFP LAN Base Image 2 CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 2 CUCM-USR-LIC Top Level Sku For User License 1 LIC-CUCM-BASIC License - 1 Basic User 50 UCM CUCM VG202 Cisco VG202 Analog Voice Gateway 1 CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 1 SVGXAISK M Cisco Voice Gateway 20x Series ADVANCED IP SERVICES 1 CP-7962G Cisco Unified IP Phone SW-CCM-UL-7962 CUCM 3.x or 4.x RTU lic. for single IP Phone CP-7965G Cisco Unified IP Phone 7965, Gig Ethernet, Color 3 SW-CCM-UL-7965 CUCM 3.x or 4.x RTU lic. for single IP Phone Features and Known Limitations Features Supported (IP Trunking) For a full list of supported SIP features please refer to the Verizon Business Retail VoIP Network Interface Specification (for non-registering devices) document. All Tests were performed according to the Verizon Business Retail VoIP Interoperability Test Plan and the EMEA Retail - Test Plan documents. These documents may be obtained by contacting your Verizon Business Account Representative. Known Limitations (IP Trunking) DTMF as RFC2833 NTE (named telephone events) when a compressed audio codec is used. RFC2833 is not currently supported when using CTI Route-Points on CISCO UCM 8.0. An MTP resource is required to enable DTMF relay for any calls that utilize a CTI Route-point. CISCO UBE performs Delayed-Offer to Early-Offer interworking of the initial SIP INIVTE from CISCO UCM. The Cisco UBE device receives the invite with no SDP then forwards the invite to the SIP network with SDP included. T.38 Fax relay is not supported by Verizon IP Trunking Service at this time If you have a Cisco Fax Server or other T.38 Fax device, you will need to ensure that design considerations have been made to support this outside of the Verizon IP Trunking service. (i.e. T1 PRI) Page 4 of 52

5 Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces messaging. CISCO UCM 8.0 can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. This is especially true for mid-call codec negotiation, for any calls that require changing of the initial negotiated codec the CISCO UBE device will insert a transcoder resource in order to avoid a codec mis-match between the SIP provider and the CUCM end-points. The CISCO UBE device must have transcoder resources configured on the Cisco UBE device and registered with CUCM to support the mid-call codec change features on Cisco UBE. This feature allows for dissimilar Voice Class Codec configurations on the incoming and outgoing dial peers. In order to comply with Verizon s requirement of supporting g711u as a secondary codec for all calls and for performing Mid-Call Codec Negotiation, Cisco has provided an acceptable solution of providing this feature via the Cisco UBE by performing transcoding from Call Manager to SIP Trunk out to the network Features Supported (IPCC) For a full list of supported SIP features please refer to the Verizon Business IP Contact Center (IPCC) Trunk Interface Network Interface Specification document. All Tests were performed according to the Verizon Business IPCC Interoperability Lab Test Plan These documents may be obtained by contacting your Verizon Business Account Representative. Known Limitations (IPCC) The IPCC service does not currently support SIP Diversion Headers The IPCC service does not support FAX Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces messaging. CISCO UCM 8.0 can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. This is especially true for mid-call codec negotiation, for any calls that require changing of the initial negotiated codec the CISCO UBE device will insert a transcoder resource in order to avoid a codec mis-match between the SIP provider and the CUCM end-points. The CISCO UBE device must have transcoder resources configured on the Cisco UBE device and registered with CUCM to support the mid-call codec change features on Cisco UBE. This feature allows for dissimilar Voice Class Codec configurations on the incoming and outgoing dial peers. Cisco UBE Features Roadmap This roadmap lists the features documented in the Cisco Unified Border Element Configuration guide and maps them to the chapters in which they appear. Also listed here is the Cisco IOS software release that introduced support for a given feature in a given Cisco IOS software release train. er.html Page 5 of 52

6 CISCO UCM 8.X SIP Trunk Deployment Considerations There are several design considerations to be taken into account when deploying SIP trunks. The following URL describes those design considerations. Call Flow Overview Outbound Call Flows The same SIP trunks are utilized between CISCO UCM to CISCO UBE for both Voice and FAX off-net calls. However, the call type (i.e., Voice vs. FAX) must be differentiated to ensure the desired codec is used. This delineation is achieved by performing digit manipulation at the Route List prior to the call being delivered to the Route Group. Each type of device (i.e., IP Phones vs. analog devices for FAX) will have separate Route-Patterns that belong to their respective partition. The route patterns will then route the call to the specified Route List. The Route List is used to distinguish a Voice call from a FAX call by manipulating the called party numbers. A voice call is forwarded with a leading 9. FAX calls will strip the leading 9 and prepend the called party number with an 8. After the digit manipulation, the Route List then forwards the call to the Route Group, which routes the call to the SIP trunks. The SIP trunks are the same for ALL calls from CISCO UCM to CISCO UBE (see example call flows below). The CISCO UBE will then forward the 10 digit user ID (DID) to the SIP Provider to allow the appropriate call routing Outbound calls can either be sent to the SIP Trunks in a Top-Down or Round-Robin method. Regardless of the method used, if when the call gets routed to the CISCO UBE and the CISCO UBE is not able to complete the call, the call is then routed to the next SIP Trunk or CISCO UBE in the Route-group. This provides redundancy for outbound calls by using multiple CISCO UBE devices connecting the VZ VoIP network. Example call flow for Voice Calls (G.729) Route Pattern 9@ For Voice Calls Route List Route Group SIP Trunk CUBE CUBE 1 VZ VoIP CUCM Cluster No digits stripped on Voice calls in CUCM CUBE CUBE 2 9 is stripped in CUBE for Voice calls Example call flow for FAX Calls (G.711ulaw) Page 6 of 52

7 A Route Pattern For FAX Calls Route List Route Group SIP Trunk CUBE CUBE 1 VZ VoIP CUCM Cluster 9 is stripped on FAX calls in CUCM and replaced with 8 CUBE CUBE 2 8 is stripped in CUBE for FAX calls Inbound Call Flows Inbound calls are received from either the IP Trunking or IPCC services. These services provide a 10 digit DID for domestic customers and a variable length DID (10, 11,12, or 13 dependent upon country) for EMEA customers for delivery of the SIP call. The IP PBX (Cisco UCM) is then responsible for routing this call to the appropriate IP Phone or analog device. Failover The VoIP Network sends periodic SIP options messages as a keepalive mechanism to determine the state of the CISCO UBE devices. If the primary CISCO UBE does not respond to these options messages, the calls are then routed to the Secondary CISCO UBE router. The CISCO UBE will respond to the SIP options pings by default. NO additional configuration is necessary. The VoIP network will also re-route any calls to the secondary CISCO UBE if it receives a temporary call setup failure SIP message from the primary CISCO UBE. (Example: 503 or 404 messages) To allow failover for inbound calls when the primary CISCO UBE device is unable to contact the CISCO UCM cluster. In the CISCO UBE: Configure voice-class sip options-keepalive on all dial-peers connecting to the CISCO UCM cluster. Change the PSTN cause code mapping under the SIP-UA configuration "set pstn-cause 1 sip-status 503" Without this configuration the incoming call setup from the VZ IP trunking service may time-out and the call would be cancelled before trying the secondary CISCO UBE device. Known Issues Inbound Call Issues When an inbound (from PSTN to Customer IP PBX) call to a DID that terminates on the SIP trunk is not defined/registered on the IP-PBX, the IP-PBX should respond with a 40X error message. Page 7 of 52

8 There are configurations on the Cisco UBE device that can cause this type of call failure to result in a call loop. This is where the call setup is routed between the VZ VoIP network and the Cisco UBE device continually until it exceeds a timeout threshold. An Example of this scenario is when the outbound dial-peer on the CISCO UBE is configured with a destination-pattern of.t, which is used as a gateway of last resort for all calls. When the Cisco UCM responds with a 40X error message the CISCO UBE will hunt for the next available dial-peer to route the call through. Example: dial-peer Voice 100 voip description OUTBOUND G729 Voice SIP calls to VzB translation-profile outgoing DIGITSTRIP-9 destination-pattern.t **This will match any combination of dialed digits and is not the recommended configuration for matching outbound calls. It is recommended to prohibit the matching of assigned DIDs on a dial-peer that is used to route calls towards the VoIP network. Voice-class codec 1 session protocol sipv2 session target sip-server dtmf-relay rtp-nte ip qos dscp af32 signaling no vad If the dial-plan requires the use of the above configuration it will become necessary to configure the CISCO UCM facing dial-peers with the huntstop feature to prevent inbound calls from trying to route back to the Verizon VoIP network. Example: dial-peer Voice 102 voip description To/From CISCO UCM subscriber for Voice preference 2 **The preferred dial-peer with a session target of the subscriber CISCO UCM(huntstop is not applied here). destination-pattern [1-5]... voice-class sip options-keepalive Voice-class codec 1 session protocol sipv2 session target ipv4: incoming called-number 9T FAX rate disable no vad dial-peer Voice 103 voip description To/From CISCO UCM publisher for Voice preference 5 **The preferred dial-peer with a session target of the subscriber CISCO UCM (huntstop is applied) huntstop destination-pattern 1... Page 8 of 52

9 voice-class sip options-keepalive Voice-class codec 1 session protocol sipv2 session target ipv4: incoming called-number 9T dtmf-relay rtp-nte no vad New Security Operation in Cisco IOS T To help mitigate toll fraud opportunities, as of T CISCO IOS no longer allows connections from "unknown" sources to connect by default. Only sources on the IP Trust List are allowed (by default) and all other calls are rejected. IP addresses defined in the "session target ipv4:" commands on dial-peers are automatically included in the IP Trust List. Additional valid source IP addresses can be added manually to the Trust List if needed by using the following CLI: voice service voip ip address trusted list ipv While it is recommended to use the increased security operation available in T, pre t IOS operation can be restored by using the CLI: no ip address trusted authenticate Redirected Dialed Number Identification Service and Diversion Header Starting with CISCO UCM Release 6.1(4) adds the Redirected Dialed Number Identification Service (RDNIS) and diversion header capability for certain calls that use the Cisco Unified Mobility Mobile Connect feature. The RDNIS/diversion header for Mobile Connect enhances this Cisco Unified Mobility feature to include the RDNIS or diversion header information on the forked call to the mobile device. Service providers and customers use the RDNIS for correct billing of end users who make Cisco Unified Mobility Mobile Connect calls. For Mobile Connect calls, the Service Providers use the RDNIS/diversion header to authorize and allow calls to originate from the enterprise, even if the caller ID does not belong to the enterprise Direct Inward Dial (DID) range. Example Use Case Consider a user that has the following setup: Desk phone number specifies Enterprise number specifies Page 9 of 52

10 Remote destination number specifies User gets a call on desk phone number ( ) that causes the remote destination ( ) to ring as well. If the user gets a call from a nonenterprise number ( ) on the enterprise number ( ), the user desk phone ( ) rings, and the call gets extended to the remote destination ( ) as well. Prior to the implementation of the RDNIS/diversion header capability, the fields populated as follows: Calling Party Number (From header in case of SIP): Called Party Number (To header in case of SIP): After implementation of the RDNIS/diversion header capability, the Calling Party Number and Called Party Number fields populate as before, but the following additional field gets populated as specified: Redirect Party Number (Diversion Header in case of SIP): Thus, the RDNIS/diversion header specifies the enterprise number that is associated with the remote destination. RDNIS Configuration in Cisco Unified Communications Manager Administration To enable the RDNIS/diversion header capability for Mobile Connect calls, ensure the following configuration takes place in Cisco Unified Communications Manager Administration: All gateways and trunks must specify that the Redirecting Number IE Delivery Outbound check box gets checked. In Cisco Unified Communications Manager Administration, you can find this check box by following the following menu paths: For H.323 and MGCP gateways, execute Device > Gateway and find the gateway that you need to configure. In the Call Routing Information - Outbound calls pane, ensure that the Redirecting Number IE Delivery - Outbound check box gets checked. For T1/E1 gateways, check the Redirecting Number IE Delivery - Outbound check box in the PRI Protocol Type Information pane. For SIP trunks, execute Device > Trunk and find the SIP trunk that you need to configure. In the Outbound Calls pane, ensure that the Redirecting Diversion Header Delivery - Outbound check box gets checked Page 10 of 52

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12 Early-Media Cut-thru: Enable PRACK on CISCO UCM Early media refers to media (e.g., audio and video) that is exchanged before the called-party accepts a particular session. Typical examples of early media generated by the called-party are ringing tone and announcements (e.g., queuing status). Early media generated by the caller typically consists of voice commands or dual tone multi-frequency (DTMF) tones to drive interactive voice response (IVR) systems. Enabling PRACK is required in order to allow early media between CISCO UCM and CISCO UBE. PRACK- Provisional Acknowledgement to a Session not yet established Purpose is to acknowledge progress information on a requested process The INVITE Includes a Require header stipulating the User Agent Client (UAC) wants a reliable provisional response SIP Rel1XX Enabled: This parameter determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably to the remote SIP endpoint. If this parameter is disabled, CISCO CallManager does not acknowledge or confirm 18X messages. Valid values specify True (acknowledge 18X messages with PRACK) or False (do not acknowledge 18X messages with PRACK). The SIP REL1XX parameter is located in the SIP Profile. Once the SIP Profile has been changed to support PRACK for all messages, the profile will then need to be applied to the appropriate SIP Trunk device. CISCO UCM Administrator>Device>Device Settings>SIP Profile Change the SIP Rel1XX Options from default value of disabled to enabled for all 1xx messages Page 12 of 52

13 No changes are required on the CISCO UBE. The CISCO UBE supports PRACK and Early Media by default. Known Issue on CUCM 8.0 with PRACK enabled: Semi-attended call transfers over SIP Trunk results in one-way audio with Prack enabled. Current workaround is to disable Prack on the SIP Trunk interface in CUCM 8.0. Page 13 of 52

14 CISCO UCM Administrator>Device>Device Settings>SIP Profile If Early-media is required as mentioned previously in this document, then PRACK will need to be enabled and the end-users will need to ensure they use fully attended transfer method to transfer calls. Page 14 of 52

15 Communications Manager Configuration Media Resource Group List List of Media Resource Groups configured for the SIP Trunk MRGL Media Resource Group Configured Conference Bridge resource associated with DSP resources configured on CISCO UBE Page 15 of 52

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17 CODEC Selection using Device Pools and Regions All Voice calls through the SIP trunk should use G.729 and FAX devices should use G.711. Note in the configuration below, there are two regions. Calls between the Default and SIP Trunk Offnet region will use G.729 and calls between Default and SIP Trunk Offnet use G.711. Applying this configuration to our testbed, the SIP trunk is placed in a Device Pool with the SIP Trunk Offnet region, and phone devices should be placed in a Device Pool that with the Default region. Devices used for analog FAX should use a Device Pool with the SIP Trunk Offnet region. Devices that belong to the same region are configured to use the G.711 codec Page 17 of 52

18 With CISCO UCM 8.0 the system defaults for Intra-Region codec preference is to use the highest quality audio codec. By default this is G722 or G711. The system default for Inter-Region codec preference is G729. The above region configuration is used to ensure that these codecs will be used if the system defaults are changed. Page 18 of 52

19 Clusterwide Parameters (System- Location and Region) Page 19 of 52

20 List of Device Pools and the associated Regions List of Phones and ATA Devices Configured Device Pools will determine the codec used by each endpoint for Offnet SIP calls Page 20 of 52

21 SIP Trunk Configuration The SIP Trunk Offnet Device Pool is configured for codec negotiation and the SIP_Trunk_MRGL is selected for Conference Bridge resources. MTP required Not Selected Page 21 of 52

22 Route Group Configuration Both SIP Trunks are members of the same Route Group Page 22 of 52

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24 Route List for Voice The previously defined ROUTE GROUP is selected in the Voice Route List Route List Details for Voice No Digits are discarded for off-net Voice calls. The leading 9 is preserved when the call is forwarded to the CISCO UBE, this allows the CISCO UBE to differentiate the call as Voice and use the corresponding G.729 CODEC. Page 24 of 52

25 Route List for FAX The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List) Page 25 of 52

26 Route List Details for FAX The 9 is stripped from the called party number and replaced with an 8. Page 26 of 52

27 This dial plan configuration ensures that the user only needs to dial a 9 for Voice and FAX off-net calls. Route Plan report for Voice and FAX Offnet calls The configured partition on each endpoint will determine how the Offnet SIP calls get routed and allows for a leading 9 to be dialed regardless of type of device. Phone or FAX devices will be able to use the same dial-plan from the user perspective. Page 27 of 52

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29 CISCO UBE Example Configuration (North America) Configuration of Cisco Unified Border Element (CISCO UBE) IOS version T2 Critical commands are marked in Bold with footnotes at bottom of each page version 15.1 service timestamps debug datetime msec localtime service timestamps log datetime msec no service password-encryption service sequence-numbers hostname CUBE boot-start-marker boot-end-marker logging buffered no logging rate-limit no logging console no aaa new-model no ipv6 cef ip source-route ip cef ip dhcp pool IPPHONES 1 network option 150 ip default-router ip domain name pipiptrunksit2.gsiv.com 2 ip name-server ip name-server multilink bundle-name authenticated crypto pki token default removal timeout 0 voice-card 0 dspfarm 1 (Optional ) DHCP Service: automatically assign IP address and TFTP server (option 150) configuration to IP Phones 2 DNS Domain name for SIP Realm and name server list for DNS resolution Page 29 of 52

30 dsp services dspfarm voice service pots voice service voip ip address trusted list 3 ipv ipv ipv ipv address-hiding allow-connections sip to sip 4 fax protocol none sip early-offer forced 5 midcall-signaling passthru 6 voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw voice class codec 2 codec preference 1 g711ulaw voice cause-code voice translation-rule 5 rule 1 /^91\(.*\)/ /+1\1/ rule 2 /^2/ / / rule 3 /^1/ / / rule 4 /^4/ / / rule 5 /^9\(.*\)/ /\1/ voice translation-rule 10 rule 2 /^9\(.*\)/ /\1/ voice translation-rule 11 rule 2 /^8\(.*\)/ /\1/ 3 Only sources on the IP Trust List are allowed (by default) and all other calls are rejected. 4 Allow SIP to SIP call Processing 5 Use this command to forcefully configure a Cisco Unified Border Element to send a SIP invite with SDP on the Out-Leg (OL), Delayed-Offer to Early-Offer for SIP calls. This is applied to all voip dial-peers. 6 Enables support for SIP Supplementary Services Page 30 of 52

31 voice translation-profile DIGITSTRIP8 7 translate called 11 translate redirect-target 5 translate redirect-called 5 voice translation-profile DIGITSTRIP9 8 translate calling 5 translate called 10 translate redirect-target 5 translate redirect-called 5 license udi pid CISCO2911/K9 hw-module pvdm 0/0 redundancy translation-rule 711 interface GigabitEthernet0/0 description connection to Vz IP Network ip address duplex auto speed auto interface GigabitEthernet0/1 description connection to CUCM LAN ip address duplex auto speed auto interface GigabitEthernet0/2 ip address dhcp shutdown duplex auto speed auto 7 Strip the leading 8 from outgoing called number, also performs digit manipulation for transferred calls. 8 Strip the leading 9 from outgoing called number, also performs digit manipulation for transferred calls and calling party number. Page 31 of 52

32 ip forward-protocol nd no ip http server no ip http secure-server ip route control-plane call treatment on 9 call threshold global cpu-avg low 68 high 75 call threshold global total-mem low 75 high 85 call threshold global total-calls low 20 high 40 voice-port 0/0/0 10 input gain -6 output attenuation 4 no non-linear no vad playout-delay maximum 120 playout-delay nominal 15 playout-delay minimum low timeouts interdigit 2 timing digit 300 station-id number 2168 caller-id enable voice-port 0/0/1 input gain -6 output attenuation 4 no non-linear no vad playout-delay maximum 120 playout-delay nominal 15 playout-delay minimum low timeouts interdigit 2 timing digit 300 station-id number caller-id enable voice-port 0/0/2 voice-port 0/0/3 9 Global Call Admission Control based on Resource utilization 10 Optional FXS port for FAX devices connected directly to the CISCO UBE Page 32 of 52

33 mgcp fax t38 ecm sccp local GigabitEthernet0/1 sccp ccm identifier 2 priority 2 version 7.0 sccp ccm identifier 1 priority 1 version 7.0 sccp sccp ccm group 10 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 12 register CONF001 associate profile 10 register XCODE001 dspfarm profile 10 transcode 11 codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 12 associate application SCCP dspfarm profile 12 conference 12 description conference bridge codec g711ulaw codec g729ar8 maximum sessions 12 associate application SCCP dial-peer voice 1 pots service session destination-pattern 2168 incoming called-number 2168 port 0/0/0 dial-peer voice 100 voip description OUTBOUND to VzB translation-profile outgoing DIGITSTRIP9 13 destination-pattern 9T 14 session protocol sipv2 session target sip-server 11 DSP Resources for Transcoding registered with CISCO UCM cluster 12 DSP resources for Conferencing registered with CISCO UCM cluster 13 Strip the leading 9 from outgoing called number 14 Match on outbound calls from CISCO UCM with leading 9 Page 33 of 52

34 voice-class codec 1 offer-all 15 voice-class sip asserted-id pai dtmf-relay rtp-nte 16 ip qos dscp af32 signaling no vad dial-peer voice 101 voip description INBOUND from VzB session protocol sipv2 session target sip-server incoming called-number [1-5] voice-class codec 1 offer-all dtmf-relay rtp-nte no vad dial-peer voice 102 voip description OUTBOUND FAX to VzB translation-profile outgoing DIGITSTRIP8 18 destination-pattern 8T no modem passthrough session protocol sipv2 session target sip-server voice-class codec 1 offer-all voice-class sip asserted-id pai voice-class sip privacy disable fax rate ip qos dscp af32 signaling no vad dial-peer voice 103 voip description INBOUND FAX dial peer from VzB translation-profile outgoing DIGITSTRIP8 session protocol sipv2 session target sip-server incoming called-number Sends a list of all available CODECs to the SIP Network. The offer-all keyword sends all available codecs without filtering based on list configured in the associated voice-class 16 Forwards DTMF tones by using RTP with the Named Telephone Event (NTE) payload type. 17 Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is required to set the DTMF, FAX, and CODEC parameters for the In-Leg of the VoIP call. 18 Strip the leading 8 from outgoing called number Page 34 of 52

35 voice-class codec 1 offer-all ip qos dscp af32 signaling no vad dial-peer voice 200 voip description connection to CM3 preference 5 destination-pattern [1-5] session protocol sipv2 session target ipv4: incoming called-number 9T 21 voice-class codec 1 voice-class sip options-keepalive 22 dtmf-relay rtp-nte fax rate no vad dial-peer voice 201 voip description connection to CM4 preference 2 destination-pattern [1-5]... session protocol sipv2 session target ipv4: incoming called-number 9T voice-class codec 1 voice-class sip options-keepalive dtmf-relay rtp-nte 19 Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is required to set the DTMF, FAX, and CODEC parameters for the In-Leg of the VoIP call. 20 Enables CISCO UBE to set configuration parameters and call routing for incoming calls destined to CISCO UCM endpoints. This will match the incoming called party information. 21 Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is required to set the DTMF, FAX, and CODEC parameters for the In-Leg of the VoIP call. 22 Enables monitoring of dial-peer targets using Out of Dialog Options PING messages. Used here to monitor the status of the CISCO UCM SIP interface. Page 35 of 52

36 fax rate no vad dial-peer voice 2 pots service session destination-pattern port 0/0/1 dial-peer voice 1167 voip description INBOUND FAX dial peer from VzB to local FXS session protocol sipv2 session target sip-server incoming called-number voice-class codec 1 offer-all ip qos dscp af32 signaling no vad dial-peer voice 300 voip description FAX dial peer from/to CUCM preference 10 destination-pattern session protocol sipv2 session target ipv4: incoming called-number 8T voice-class codec 2 24 ip qos dscp af32 signaling no vad dial-peer voice 301 voip description FAX dial peer from/to CUCM preference 1 destination-pattern 1174 session protocol sipv2 session target ipv4: incoming called-number 8T 25 voice-class codec 2 ip qos dscp af32 signaling no vad sip-ua set pstn-cause 1 sip-status 503 set pstn-cause 3 sip-status Enables CISCO UBE to set configuration parameters and call routing for incoming calls destined to CISCO UCM endpoints. This will match the incoming called party information. 24 CODEC is set on dial-peer to force use of g711ulaw for FAX calls. 25 Match on outbound calls from CISCO UCM with leading 8 Page 36 of 52

37 retry invite 2 retry bye 2 retry cancel 2 timers trying 550 sip-server dns:pcclv1n0022.pipiptrunksit2.gsiv.com 27 g729-annexb override gatekeeper shutdown line con 0 stopbits 1 line aux 0 stopbits 1 line vty 0 4 exec-timeout 0 0 privilege level 15 logging synchronous login local transport input ssh line vty 5 15 exec-timeout 0 0 privilege level 15 password password logging synchronous login local transport input ssh exception data-corruption buffer truncate scheduler allocate ntp master 3 ntp peer ntp peer end 26 Overrides the default value of the SIP status code to correspond with the PSTN cause code. 27 SIP Proxy FQDN name for outbound SIP calls to the IP Trunking service Page 37 of 52

38 EMEA Configuration EMEA CISCO UCM Configuration The following steps are required to enable localised Network tones and User Interface: 1. Download necessary localisation files from (requires valid CCO account) 2. Install localisation software on every Communications Manager in the cluster. Page 38 of 52

39 This does require a restart to enable the localisation file after installation. Page 39 of 52

40 3. Using the CISCO UCM Administration website either change the locale information at the device pool level or at the Phone device level. Example shows change to Network Locale on the Phone configuration page: User Locale changes the User interface only and is controlled independently of the network tones. Page 40 of 52

41 4. All EMEA Phones should be setup in similar regions as the North America phone configuration to ensure G729 is the preferred CODEC. 5. Next create a variable-length Route-Pattern with # as terminating digit. Example: 9.011# The previously configured Voice Route List is utilized for this route-pattern in order to allow the complete calling number to be sent to CISCO UBE. Page 41 of 52

42 Page 42 of 52

43 EMEA CISCO UBE dial-peer Configuration The CISCO UBE configuration for EMEA is very similar to the US (Domestic) IP Trunking configuration. dial-peer Voice 100 voip description OUTBOUND G729 Voice SIP calls to VzB translation-profile outgoing DIGITSTRIP9 destination-pattern 9T voice-class codec 1 offer-all session protocol sipv2 session target sip-server dtmf-relay rtp-nte ip qos dscp af32 signaling no vad dial-peer Voice 101 voip description INBOUND Voice SIP calls from VzB EMEA voice-class codec 1 offer-all session protocol sipv2 session target sip-server incoming called-number [1-5]... dtmf-relay rtp-nte no vad dial-peer Voice 102 voip description To/From CISCO UCM subscriber for Voice preference 2 destination-pattern [1-5]... voice-class sip options-keepalive voice-class codec 1 session protocol sipv2 session target ipv4: incoming called-number 9T FAX rate disable no vad dial-peer Voice 103 voip description To/From CISCO UCM publisher for Voice preference 5 destination-pattern 1... voice-class sip options-keepalive voice-class codec 1 session protocol sipv2 session target ipv4: incoming called-number 9T dtmf-relay rtp-nte no vad Page 43 of 52

44 IPCC Configuration IPCC CISCO UCM Configuration The CISCO UCM Configuration changes required for IPCC services to work properly are: Verify all IPCC end-points (Phones and Gateways) are in the same Region to allow negotiation of the G.711ulaw codec. Disable diversion-header support on the SIP Trunk device configuration.***need to check this graphic Page 44 of 52

45 For out-bound IPCC calls a XXXX Route-pattern must be configured in the Communications Manager. Page 45 of 52

46 IPCC CISCO UBE dial-peer Configuration The CISCO UBE dial-peers must be configured to negotiate only the G.711 codec for all IPCC inbound calls. This requires specific incoming called numbers for IP Toll-Free calls. Example: User calls and IPCC routes the call to 1212 with the following dial-peer configured on the CISCO UBE router. In this example the IPCC network is only sending the last 4 digits of the called number. dial-peer voice 800 voip description OUTBOUND to VzB IP Toll Free translation-profile outgoing DIGITSTRIP9 destination-pattern T codec g711ulaw session protocol sipv2 session target dns:rchtcsd05011.vzbi.com dtmf-relay rtp-nte ip qos dscp af32 signaling no vad dial-peer voice 801 voip description G.711 INBOUND from VzB IP Toll Free codec g711ulaw session protocol sipv2 session target sip-server incoming called-number 1212 dtmf-relay rtp-nte no vad dial-peer voice 802 voip description G.711 To/From CISCO UCM subscriber IP Toll Free preference 2 destination-pattern 1212 voice-class sip options-keepalive codec g711ulaw session protocol sipv2 session target ipv4: dtmf-relay rtp-nte no vad dial-peer voice 803 voip Page 46 of 52

47 description G.711 To/From CISCO UCM publisher IP Toll Free preference 5 destination-pattern 1212 voice-class sip options-keepalive voice-class codec 2 voice-class sip early-offer forced session protocol sipv2 session target ipv4: dtmf-relay rtp-nte no vad Troubleshooting Always capture logs by enabling logging buffer: logging buffered Remember to disable the console logging: no logging console Add sequence numbering for debugs: service sequence-number Debug Commands debug ccsip all debug voip ccapi inout debug voip dialpeer inout debug transcoding debug dspfarm all The following table lists key "show" commands giving output that enables you to monitor Cisco UBE health, traffic and activity. Key "show" Commands on Cisco UBE Page 47 of 52

48 Category Command Information Provided Configuration show version Displays the version of the image on the router show flash: show ip interface brief show startup-config show running-config show debug show voice iec desc <> show logging Displays information about flash: file system Displays brief summary of IP status and configuration Displays the startup configuration on the router Displays the present/running con configuration on the router Displays the debugs currently enabled Displays definition of an Internal Error Code Displays the contents of logging buffers Traffic show call active voice Displays complete details of an active call like media settings, call statistics, SRTP on/off, etc. Router Health show call active voice brief show call active voice compact show voip rtp connections show call history voice show processes cpu sorted <1min/5min/5sec>" show processes cpu sorted history show memory processor show process memory <> show memory debug leaks show alignment Displays a brief version of active voice calls, e.g. transmitted and received packets and duration of call Displays a compact version of active voice calls Displays active RTP connections Displays calls stored in the history table for voice Displays sorted output based on percentage of CPU utilization Displays CPU history information in a graph format Displays memory statistics Displays memory per process Runs the memory leak detector Displays alignment data and spurious memory references CAC show call threshold config Displays configured resource information Page 48 of 52

49 show call treatment config show call treatment stats Displays call admission control information Displays call treatment statistics SIP show sip-ua connections udp brief Displays summary of SIP UDP connection information show sip-ua connections udp detail show sip-ua connections tcp brief show sip-ua connections tcp detail show sip-ua register status Displays details of SIP UDP connection information Displays summary of SIP TCP connection information Displays details of SIP TCP connection information Displays SIP registration status Transcoding and DSPs show diag Displays diagnostic and hardware information for port adapters and modules DTMF Relay show sdspfarm units show sccp connection show sccp show dspfarm dsp active show call active voice inc CoderTypeRate=" show call active voice comp show call active voice inc tx_dtmfrelay Displays transcoder registration status Displays the active SCCP connections Displays SCCP protocol information Displays the active DSPs Displays call connectivity, codec and the media type information Displays codec information for transcoding calls Displays the DTMF-relay used for the call Page 49 of 52

50 References Cisco UBE on Cisco.com CISCO UCM 8x SIP Trunk Documentation: - wp Cisco UBE PBX / SP Interoperability Verizon Business IP Trunking Services Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP) Redirected Dialed Number Identification Service and Diversion Header Cisco Unified Border Element (CUBE) Management and Manageability Specification Acronyms Acronym SIP SCCP TDM CISCO UCM CISCO UBE PRACK TUI Definition Session Initiation Protocol Skinny Client Control Protocol Time Division Multiplexing Cisco Unified Communications Manager Cisco Unified Border Element Provisional Response Acknowledgement Telephony User Interface Page 50 of 52

51 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. 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, Cisco TelePresence, the Cisco logo and the Cisco Square Bridge logo are trademarks of Cisco Systems, Inc.; Ciso Store and Changing the Way We Work, Live, Play, and Learn are service marks 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, Page 51 of 52

52 HomeLink, Internet Quotient, IOS, iphone, iq Expertise, the iq logo, iq Net Readiness Scorecard, iquick Study, LightStream, Linksys, MeetingPlace, MeetingPlace Chime Sound, MGX, Networking Academy, Network Registrar, Packet, PIX, ProConnect, ScriptShare, SMARTnet, StackWise, The Fastest Way to Increase Your Internet Quotient, and TransPath 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 52 of 52