Overview

Webex Calling currently supports two versions of Local Gateway:

  • Local Gateway

  • Local Gateway for Webex for Government

  • Before you begin, understand the premises-based Public Switched Telephone Network (PSTN) and Local Gateway (LGW) requirements for Webex Calling. See Cisco Preferred Architecture for Webex Calling for more information.

  • This article assumes that a dedicated Local Gateway platform is in place with no existing voice configuration. If you modify an existing PSTN gateway or CUBE Enterprise deployment to use as the Local Gateway function for Webex Calling, then pay careful attention to the configuration. Ensure that you don't interrupt the existing call flows and functionality because of the changes that you make.


 
The procedures contain links to command reference documentation where you can learn more about the individual command options. All command reference links go to the Webex Managed Gateways Command Reference unless stated otherwise (in which case, the command links go to Cisco IOS Voice Command Reference). You can access all these guides at Cisco Unified Border Element Command References.

For information on the supported third-party SBCs, refer to the respective product reference documentation.

There are two options to configure the Local Gateway for your Webex Calling trunk:

  • Registration-based trunk

  • Certificate-based trunk

Use the task flow either under the Registration-based Local Gateway or Certificate-based Local Gateway to configure Local Gateway for your Webex Calling trunk.

See Get started with Local Gateway for more information on different trunk types. Perform the following steps on the Local Gateway itself, using the Command Line Interface (CLI). We use Session Initiation Protocol (SIP) and Transport Layer Security (TLS) transport to secure the trunk and Secure Real Time Protocol (SRTP) to secure the media between the Local Gateway and Webex Calling.

Local Gateway for Webex for Government doesn’t support the following:

  • STUN/ICE-Lite for media path optimization

  • Fax (T.38)

To configure Local Gateway for your Webex Calling trunk in Webex for Government, use the following option:

  • Certificate-based trunk

Use the task flow under the Certificate-based Local Gateway to configure the Local Gateway for your Webex Calling trunk. For more details on how to configure a certificate-based Local Gateway, see Configure Webex Calling certificate-based trunk.

It’s mandatory to configure FIPS-compliant GCM ciphers to support Local Gateway for Webex for Government. If not, the call setup fails. For configuration details, see Configure Webex Calling certificate-based trunk.


 
Webex for Government doesn’t support registration-based Local Gateway.

This section describes how to configure a Cisco Unified Border Element (CUBE) as a Local Gateway for Webex Calling, using a registering SIP trunk. The first part of this document illustrates how to configure a simple PSTN gateway. In this case, all calls from the PSTN are routed to Webex Calling and all calls from Webex Calling are routed to the PSTN. The image below highlights this solution and the high-level call routing configuration that will be followed.

In this design, the following principal configurations are used:

  • voice class tenants: Used to create trunk specific configurations.

  • voice class uri: Used to classify SIP messages for the selection of an inbound dial-peer.

  • inbound dial-peer: Provides treatment for inbound SIP messages and determines the outbound route with a dial-peer group.

  • dial-peer group: Defines the outbound dial-peers used for onward call routing.

  • outbound dial-peer: Provides treatment for outbound SIP messages and routes them to the required target.

Call routing from/to PSTN to/from Webex Calling configuration solution

When connecting an on-premises Cisco Unified Communications Manager solution with Webex Calling, you can use the simple PSTN gateway configuration as a baseline for building the solution illustrated in the following diagram. In this case, Unified Communications Manager provides centralized routing and treatment of all PSTN and Webex Calling calls.

Throughout this document, the host names, IP addresses, and interfaces illustrated in the following image are used.

Use the configuration guidance in the rest of this document to complete your Local Gateway configuration as follows:

  • Step 1: Configure router baseline connectivity and security

  • Step 2: Configure Webex Calling Trunk

    Depending on your required architecture, follow either:

  • Step 3: Configure Local Gateway with SIP PSTN trunk

  • Step 4: Configure Local Gateway with existing Unified CM environment

    Or:

  • Step 3: Configure Local Gateway with TDM PSTN trunk

Baseline configuration

The first step in preparing your Cisco router as a Local Gateway for Webex Calling is to build a baseline configuration that secures your platform and establishes connectivity.

  • All registration-based Local Gateway deployments require Cisco IOS XE 17.6.1a or later versions. For the recommended versions, see the Cisco Software Research page. Search for the platform and select one of the suggested releases.

    • ISR4000 series routers must be configured with both Unified Communications and Security technology licenses.

    • Catalyst Edge 8000 series routers fitted with voice cards or DSPs require DNA Advantage licensing. Routers without voice cards or DSPs require a minimum of DNA Essentials licensing.

  • Build a baseline configuration for your platform that follows your business policies. In particular, configure the following and verify the working:

    • NTP

    • ACLs

    • User authentication and remote access

    • DNS

    • IP routing

    • IP addresses

  • The network toward Webex Calling must use an IPv4 address.

  • Upload the Cisco root CA bundle to the Local Gateway.

Configuration

1

Ensure that you assign valid and routable IP addresses to any Layer 3 interfaces, for example:


interface GigabitEthernet0/0/0
  description Interface facing PSTN and/or CUCM
  ip address 10.80.13.12 255.255.255.0
!
interface GigabitEthernet0/0/1
  description Interface facing Webex Calling (Private address)
  ip address 192.51.100.1 255.255.255.240

2

Protect registration and STUN credentials on the router using symmetric encryption. Configure the primary encryption key and encryption type as follows:


key config-key password-encrypt YourPassword
password encryption aes

3

Create a placeholder PKI trustpoint.


 
Requires this trustpoint to configure TLS later. For registration-based trunks, this trustpoint doesn't require a certificate - as would be required for a certificate-based trunk.

crypto pki trustpoint EmptyTP 
 revocation-check none
4

Enable TLS1.2 exclusivity and specify the default trustpoint using the following configuration commands. Transport parameters should also be updated to ensure a reliable secure connection for registration:


 
The cn-san-validate server command ensures that the Local Gateway permits a connection if the host name configured in tenant 200 is included in either the CN or SAN fields of the certificate received from the outbound proxy.
  1. Set tcp-retry count to 1000 (5-msec multiples = 5 seconds).

  2. The timer connection establish command allows you to tune how long the LGW waits to set up a connection with a proxy before considering the next available option. The default for this timer is 20 seconds and the minimum 5 seconds. Start with a low value and increase if necessary to accommodate network conditions.


sip-ua
 timers connection establish tls 5
 transport tcp tls v1.2
 crypto signaling default trustpoint EmptyTP cn-san-validate server
 tcp-retry 1000

5

Install the Cisco root CA bundle, which includes the DigiCert CA certificate used by Webex Calling. Use the crypto pki trustpool import clean url command to download the root CA bundle from the specified URL, and to clear the current CA trustpool, then install the new bundle of certificates:


 

If you need to use a proxy for access to the internet using HTTPS, add the following configuration before importing the CA bundle:

ip http client proxy-server yourproxy.com proxy-port 80

ip http client source-interface GigabitEthernet0/0/1 
crypto pki trustpool import clean url https://www.cisco.com/security/pki/trs/ios_core.p7b
1

Create a registration based PSTN trunk for an existing location in Control Hub. Make a note of the trunk information that is provided once the trunk has been created. These details, as highlighted in the following illustration, will be used in the configuration steps in this guide. For more information, see Configure trunks, route groups, and dial plans for Webex Calling.

2

Enter the following commands to configure CUBE as a Webex Calling Local Gateway:

 
voice service voip
 ip address trusted list
  ipv4 x.x.x.x y.y.y.y
 mode border-element
 media statistics
 media bulk-stats 
 allow-connections sip to sip
 no supplementary-service sip refer  
 stun
  stun flowdata agent-id 1 boot-count 4
  stun flowdata shared-secret 0 Password123$
 sip
  asymmetric payload full
  early-offer forced  

Here's an explanation of the fields for the configuration:


ip address trusted list
 ipv4 x.x.x.x y.y.y.y
  • To protect against toll fraud, the trusted address list defines a list of hosts and networks from which the Local Gateway expects legitimate VoIP calls.

  • By default, Local Gateway blocks all incoming VoIP messages from IP addresses not in its trusted list. Statically configured dial-peers with “session target IP” or server group IP addresses are trusted by default, so do not need to be added to the trusted list.

  • When configuring your Local Gateway, add the IP subnets of your regional Webex Calling data center to the list. For more information, see Port Reference Information for Webex Calling. Also, add address ranges for Unified Communications Manager servers (if used) and PSTN trunk gateways.


     

    If your LGW is behind a firewall with restricted cone NAT, you may prefer to disable the IP address trusted list on the Webex Calling facing interface. The firewall already protects you from unsolicited inbound VoIP. Disable action reduces your longer-term configuration overhead, because we cannot guarantee that the addresses of the Webex Calling peers remain fixed, and you must configure your firewall for the peers in any case.

mode border-element

Enables Cisco Unified Border Element (CUBE) features on the platform.

media statistics

Enables media monitoring on the Local Gateway.

media bulk-stats

Enables the control plane to poll the data plane for bulk call statistics.

For more information on these commands, see Media.

allow-connections sip to sip

Enable CUBE basic SIP back-to-back user agent functionality. For more information, see Allow connections.


 

By default, T.38 fax transport is enabled. For more information, see fax protocol t38 (voice-service).

stun

Enables STUN (Session Traversal of UDP through NAT) globally.

  • When you forward a call to a Webex Calling user (for example, both the called and calling parties are Webex Calling subscribers and if you anchor media at the Webex Calling SBC), then the media cannot flow to the Local Gateway as the pinhole isn't open.

  • The STUN bindings feature on the Local Gateway allows locally generated STUN requests to be sent over the negotiated media path. This helps to open the pinhole in the firewall.

For more information, see stun flowdata agent-id and stun flowdata shared-secret.

asymmetric payload full

Configures SIP asymmetric payload support for both DTMF and dynamic codec payloads. For more information on this command, see asymmetric payload.

early-offer forced

Forces the Local Gateway to send SDP information in the initial INVITE message instead of waiting for acknowledgment from the neighboring peer. For more information on this command, see early-offer.

3

Configure voice class codec 100 filter for the trunk. In this example, the same codec filter is used for all trunks. You can configure filters for each trunk for precise control.


voice class codec 100
 codec preference 1 opus
 codec preference 2 g711ulaw
 codec preference 3 g711alaw

Here's an explanation of the fields for the configuration:

voice class codec 100

Used to only allow preferred codecs for calls through SIP trunks. For more information, see voice class codec.


 

Opus codec is supported only for SIP-based PSTN trunks. If the PSTN trunk uses a voice T1/E1 or analog FXO connection, exclude codec preference 1 opus from the voice class codec 100 configuration.

4

Configure voice class stun-usage 100 to enable ICE on the Webex Calling trunk.


voice class stun-usage 100 
 stun usage firewall-traversal flowdata
 stun usage ice lite

Here's an explanation of the fields for the configuration:

stun usage ice lite

Used to enable ICE-Lite for all Webex Calling facing dial-peers to allow media-optimization whenever possible. For more information, see voice class stun usage and stun usage ice lite.


 

You require stun usage of ICE-lite for call flows using media path optimization. To provide media-optimization for a SIP to TDM gateway, configure a loopback dial-peer with ICE-Lite enabled on the IP-IP leg. For further technical details, contact the Account or TAC teams

5

Configure the media encryption policy for Webex traffic.


voice class srtp-crypto 100
 crypto 1 AES_CM_128_HMAC_SHA1_80

Here's an explanation of the fields for the configuration:

voice class srtp-crypto 100

Specifies SHA1_80 as the only SRTP cipher-suite CUBE offers in the SDP in offer and answer messages. Webex Calling only supports SHA1_80. For more information, see voice class srtp-crypto.

6

Configure a pattern to uniquely identify calls to a Local Gateway trunk based on its destination trunk parameter:


voice class uri 100 sip
 pattern dtg=Dallas1463285401_LGU

Here's an explanation of the fields for the configuration:

voice class uri 100 sip

Defines a pattern to match an incoming SIP invite to an incoming trunk dial-peer. When entering this pattern, use dtg= followed by the Trunk OTG/DTG value provided in Control Hub when the trunk was created. For more information, see voice class uri.

7

Configure sip profile 100, which will be used to modify SIP messages before they are sent to Webex Calling.


voice class sip-profiles 100
 rule 10 request ANY sip-header SIP-Req-URI modify "sips:" "sip:"
 rule 20 request ANY sip-header To modify "<sips:" "<sip:"
 rule 30 request ANY sip-header From modify "<sips:" "<sip:"
 rule 40 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>" 
 rule 50 response ANY sip-header To modify "<sips:" "<sip:"
 rule 60 response ANY sip-header From modify "<sips:" "<sip:"
 rule 70 response ANY sip-header Contact modify "<sips:" "<sip:"
 rule 80 request ANY sip-header From modify ">" ";otg=dallas1463285401_lgu>"
 rule 90 request ANY sip-header P-Asserted-Identity modify "sips:" "sip:"

Here's an explanation of the fields for the configuration:

  • rule 10 to 70 and 90

    Ensures that SIP headers used for call signaling use the sip, rather than sips scheme, which is required by Webex proxies. Configuring CUBE to use sips ensures that secure registration is used.

  • rule 80

    Modifies the From header to include the trunk group OTG/DTG identifier from Control Hub to uniquely identify a Local Gateway site within an enterprise.

8

Configure Webex Calling trunk:

  1. Create voice class tenant 100 to define and group configurations required specifically for the Webex Calling trunk. In particular, the trunk registration details provided in Control Hub earlier will be used in this step as detailed below. Dial-peers associated with this tenant later will inherit these configurations.


     

    The following example uses the values illustrated in Step 1 for the purpose of this guide (shown in bold). Replace these with values for your trunk in your configuration.

    
    voice class tenant 100
      registrar dns:98027369.us10.bcld.webex.com scheme sips expires 240 refresh-ratio 50 tcp tls
      credentials number Dallas1171197921_LGU username Dallas1463285401_LGU password 0 9Wt[M6ifY+ realm BroadWorks
      authentication username Dallas1463285401_LGU password 0 9Wt[M6ifY+ realm BroadWorks
      authentication username Dallas1463285401_LGU password 0 9Wt[M6ifY+ realm 98027369.us10.bcld.webex.com
      no remote-party-id
      sip-server dns:98027369.us10.bcld.webex.com
      connection-reuse
      srtp-crypto 100
      session transport tcp tls 
      url sips 
      error-passthru
      asserted-id pai 
      bind control source-interface GigabitEthernet0/0/1
      bind media source-interface GigabitEthernet0/0/1
      no pass-thru content custom-sdp 
      sip-profiles 100 
      outbound-proxy dns:dfw04.sipconnect-us.bcld.webex.com  
      privacy-policy passthru
    

    Here's an explanation of the fields for the configuration:

    voice class tenant 100

    Defines a set of configuration parameters that will be used only for the Webex Calling trunk. For more information, see voice class tenant.

    registrar dns:98027369.us10.bcld.webex.com scheme sips expires 240 refresh-ratio 50 tcp tls

    Registrar server for the Local Gateway with the registration set to refresh every two minutes (50% of 240 seconds). For more information, see registrar.

    Ensure that you use the Register Domain value from Control Hub here.

    credentials number Dallas1171197921_LGU username Dallas1463285401_LGU password 0 9Wt[M6ifY+ realm BroadWorks

    Credentials for trunk registration challenge. For more information, see credentials (SIP UA).

    Ensure that you use the Line/Port host, Authentication Username and Authentication Password values respectively from Control Hub here.

    authentication username Dallas1171197921_LGU password 0 9Wt[M6ifY+ realm BroadWorks
    authentication username Dallas1171197921_LGU password 0 9Wt[M6ifY+ realm 98027369.us10.bcld.webex.com

    Authentication challenge for calls. For more information, see authentication (dial-peer).

    Ensure that you use the Authentication Username, Authentication Password and Registrar Domain values respectively from Control Hub here.

    no remote-party-id

    Disable SIP Remote-Party-ID (RPID) header as Webex Calling supports PAI, which is enabled using CIO asserted-id pai. For more information, see remote-party-id.

    sip-server dns:us25.sipconnect.bcld.webex.com

    Configures the target SIP server for the trunk. Use the edge proxy SRV address provided in Control Hub when you created your trunk.

    connection-reuse

    Uses the same persistent connection for registration and call processing. For more information, see connection-reuse.

    srtp-crypto 100

    Configures the preferred cipher-suites for the SRTP call leg (connection) (specified in step 5). For more information, see voice class srtp-crypto.

    session transport tcp tls

    Sets transport to TLS. For more information, see session-transport.

    url sips

    SRV query must be SIPs as supported by the access SBC; all other messages are changed to SIP by sip-profile 200.

    error-passthru

    Specifies SIP error response pass-thru functionality. For more information, see error-passthru.

    asserted-id pai

    Turns on PAI processing in Local Gateway. For more information, see asserted-id.

    bind control source-interface GigabitEthernet0/0/1

    Configures the source interface and associated IP address for messages sent to WebexCalling. For more information, see bind.

    bind media source-interface GigabitEthernet0/0/1

    Configures the source interface and associated IP address for media sent to WebexCalling. For more information, see bind.

    no pass-thru content custom-sdp

    Default command under tenant. For more information on this command, see pass-thru content.

    sip-profiles 100

    Changes SIPs to SIP and modify Line/Port for INVITE and REGISTER messages as defined in sip-profiles 100. For more information, see voice class sip-profiles.

    outbound-proxy dns:dfw04.sipconnect-us.bcld.webex.com

    Webex Calling access SBC. Insert the Outbound Proxy Address provided in Control Hub when you created your trunk. For more information, see outbound-proxy.

    privacy-policy passthru

    Configures the privacy header policy options for the trunk to pass privacy values from the received message to the next call leg. For more information, see privacy-policy.

  2. Configure the Webex Calling trunk dial-peer.

    
    dial-peer voice 100 voip
     description Inbound/Outbound Webex Calling
     max-conn 250
     destination-pattern BAD.BAD
     session protocol sipv2
     session target sip-server
     incoming uri request 100
     voice-class codec 100
     dtmf-relay rtp-nte
     voice-class stun-usage 100
     no voice-class sip localhost
     voice-class sip tenant 100
     srtp
     no vad
    

    Here's an explanation of the fields for the configuration:

    
    dial-peer voice 100 voip
      description Inbound/Outbound Webex Calling
    

    Defines a VoIP dial-peer with a tag of 100 and gives a meaningful description for ease of management and troubleshooting.

    max-conn 250

    Restricts the number of concurrent inbound and outbound calls between the LGW and Webex Calling. For registration trunks, the maximum value configured should be 250. Usea lower value if that would be more appropriate for your deployment. For more information on concurrent call limits for Local Gateway, refer to the Get started with Local Gateway document.

    destination-pattern BAD.BAD

    A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. Any valid destination pattern may be used in this case.

    session protocol sipv2

    Specifies that dial-peer 100 handles SIP call legs. For more information, see session protocol (dial-peer).

    session target sip-server

    Indicates that the SIP server defined in tenant 100 is inherited and used for the destination for calls from this dial peer.

    incoming uri request 100

    To specify the voice class used to match a VoIP dial peer to the uniform resource identifier (URI) of an incoming call. For more information, see incoming uri.

    voice-class codec 100

    Configures the dial-peer to use the common codec filter list 100. For more information, see voice-class codec.

    voice-class stun-usage 100

    Allows locally generated STUN requests on the Local Gateway to be sent over the negotiated media path. STUN helps to open a firewall pinhole for media traffic.

    no voice-class sip localhost

    Disables substitution of the DNS local host name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages.

    voice-class sip tenant 100

    The dial-peer inherits all parameters configured globally and in tenant 100. Parameters may be overridden at the dial-peer level.

    srtp

    Enables SRTP for the call leg.

    no vad

    Disables voice activity detection.

After you define tenant 100 and configure a SIP VoIP dial-peer, the gateway initiates a TLS connection toward Webex Calling. At this point the access SBC presents its certificate to the Local Gateway. The Local Gateway validates the Webex Calling access SBC certificate using the CA root bundle that was updated earlier. If the certificate is recognised, a persistent TLS session is established between the Local Gateway and Webex Calling access SBC. The Local Gateway is then able to use this secure connection to register with the Webex access SBC. When the registration is challenged for authentication:

  • The username, password, and realm parameters from the credentials configuration is used in the response.

  • The modification rules in sip profile 100 are used to convert SIPS URL back to SIP.

Registration is successful when a 200 OK is received from the access SBC.

Having built a trunk towards Webex Calling above, use the following configuration to create a non-encrypted trunk towards a SIP based PSTN provider:


 

If your Service Provider offers a secure PSTN trunk, you may follow a similar configuration as detailed above for the Webex Calling trunk. Secure to secure call routing is supported by CUBE.


 

To configure TDM interfaces for PSTN call legs on the Cisco TDM-SIP Gateways, see  Configuring ISDN PRI.

1

Configure the following voice class uri to identify inbound calls from the PSTN trunk:


voice class uri 200 sip
  host ipv4:192.168.80.13

Here's an explanation of the fields for the configuration:

voice class uri 200 sip

Defines a pattern to match an incoming SIP invite to an incoming trunk dial-peer. When entering this pattern, use the IP address of you IP PSTN gateway. For more information, see  voice class uri.

2

Configure the following IP PSTN dial-peer:


dial-peer voice 200 voip
 description Inbound/Outbound IP PSTN trunk
 destination-pattern BAD.BAD
 session protocol sipv2
 session target ipv4:192.168.80.13
 incoming uri via 200
 voice-class sip bind control source-interface GigabitEthernet0/0/0 
 voice-class sip bind media source-interface  GigabitEthernet0/0/0 
 voice-class codec 100
 dtmf-relay rtp-nte 
 no vad

Here's an explanation of the fields for the configuration:


dial-peer voice 200 voip
 description Inbound/Outbound IP PSTN trunk

Defines a VoIP dial-peer with a tag of 200 and gives a meaningful description for ease of management and troubleshooting. For more information, see dial-peer voice.

destination-pattern BAD.BAD

A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface).

session protocol sipv2

Specifies that dial-peer 200 handles SIP call legs. For more information, see session protocol (dial peer).

session target ipv4:192.168.80.13

Indicates the destination’s target IPv4 address to send the call leg. The session target here is ITSP’s IP address. For more information, see  session target (VoIP dial peer).

incoming uri via 200

Defines a match criterion for the VIA header with the IP PSTN’s IP address. Matches all incoming IP PSTN call legs on the Local Gateway with dial-peer 200. For more information, see  incoming url.

bind control source-interface GigabitEthernet0/0/0

Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see  bind.

bind media source-interface GigabitEthernet0/0/0

Configures the source interface and associated IP address for media sent to PSTN. For more information, see  bind.

voice-class codec 100

Configures the dial-peer to use the common codec filter list 100. For more information, see voice-class codec.

dtmf-relay rtp-nte

Defines RTP-NTE (RFC2833) as the DTMF capability expected on the call leg. For more information, see DTMF Relay (Voice over IP).

no vad

Disables voice activity detection. For more information, see vad (dial peer).

3

If you are configuring your Local Gateway to only route calls between Webex Calling and the PSTN, add the following call routing configuration. If you are configuring your Local Gateway with a Unified Communications Manager platform, skip to the next section.

  1. Create dial-peer groups to route calls towards Webex Calling or the PSTN. Define DPG 100 with outbound dial-peer 100 toward Webex Calling. DPG 100 is applied to the incoming dial-peer from the PSTN. Similarly, define DPG 200 with outbound dial-peer 200 toward the PSTN. DPG 200 is applied to the incoming dial-peer from Webex.

    
    voice class dpg 100 
     description Route calls to Webex Calling 
     dial-peer 100 
    voice class dpg 200 
     description Route calls to PSTN 
     dial-peer 200

    Here's an explanation of the fields for the configuration:

    dial-peer 100

    Associates an outbound dial-peer with a dial-peer group. For more information, see  voice-class dpg.

  2. Apply dial-peer groups to route calls from Webex to the PSTN and from the PSTN to Webex:

    
    dial-peer voice 100
     destination dpg 200
    dial-peer voice 200
     destination dpg 100 

    Here's an explanation of the fields for the configuration:

    destination dpg 200

    Specifies which dial-peer group, and therefore dial-peer should be used for the outbound treatment for calls presented to this incoming dial-peer.

    This concludes your Local Gateway configuration. Save the configuration and reload the platform if this is the first time CUBE features are configured.

The PSTN-Webex Calling configuration in the previous sections may be modified to include additional trunks to a Cisco Unified Communications Manager (UCM) cluster. In this case, all calls are routed via Unified CM. Calls from UCM on port 5060 are routed to the PSTN and calls from port 5065 are routed to Webex Calling. The following incremental configurations may be added to include this calling scenario.


 

When creating the Webex Calling trunk in Unified CM, ensure that you configure the incoming port in the SIP Trunk Security Profile settings to 5065. This allows incoming messages on port 5065 and populate the VIA header with this value when sending messages to the Local Gateway.

1

Configure the following voice class URIs:

  1. Classifies Unified CM to Webex calls using SIP VIA port:

    
    voice class uri 300 sip
     pattern :5065
    
  2. Classifies Unified CM to PSTN calls using SIP via port:

    
    voice class uri 400 sip
     pattern :192\.168\.80\.6[0-5]:5060
    

    Classify incoming messages from the UCM towards the PSTN trunk using one or more patterns that describe the originating source addresses and port number. Regular expressions may be used to define matching patterns if required.

    In the example above, a regular expression is used to match any IP address in the range 192.168.80.60 to 65 and port number 5060.

2

Configure the following DNS records to specify SRV routing to Unified CM hosts:


 

IOS XE uses these records for locally determining target UCM hosts and ports. With this configuration, it is not required to configure records in your DNS system. If you prefer to use your DNS, then these local configurations are not required.


ip host ucmpub.mydomain.com 192.168.80.60
ip host ucmsub1.mydomain.com 192.168.80.61
ip host ucmsub2.mydomain.com 192.168.80.62
ip host ucmsub3.mydomain.com 192.168.80.63
ip host ucmsub4.mydomain.com 192.168.80.64
ip host ucmsub5.mydomain.com 192.168.80.65
ip host _sip._udp.wxtocucm.io srv 0 1 5065 ucmpub.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub1.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub2.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub3.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub4.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub5.mydomain.com
ip host _sip._udp.pstntocucm.io srv 0 1 5060 ucmpub.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub1.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub2.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub3.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub4.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub5.mydomain.com

Here's an explanation of the fields for the configuration:

The following command creates a DNS SRV resource record. Create a record for each UCM host and trunk:

ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub5.mydomain.com

_sip._udp.pstntocucm.io: SRV resource record name

2: The SRV resource record priority

1: The SRV resource record weight

5060: The port number to use for the target host in this resource record

ucmsub5.mydomain.com: The resource record target host

To resolve the resource record target host names, create local DNS A records. For example:

ip host ucmsub5.mydomain.com 192.168.80.65

ip host: Creates a record in the local IOS XE database.

ucmsub5.mydomain.com: The A record host name.

192.168.80.65: The host IP address.

Create the SRV resource records and A records to reflect your UCM environment and preferred call distribution strategy.

3

Configure the following dial-peers:

  1. Dial-peer for calls between Unified CM and Webex Calling:

    
    dial-peer voice 300 voip
     description UCM-Webex Calling trunk
     destination-pattern BAD.BAD
     session protocol sipv2
     session target dns:wxtocucm.io
     incoming uri via 300
     voice-class codec 100
     voice-class sip bind control source-interface GigabitEthernet 0/0/0
     voice-class sip bind media source-interface GigabitEthernet 0/0/0
     dtmf-relay rtp-nte
     no vad
    

    Here's an explanation of the fields for the configuration:

    
    dial-peer voice 300 voip
     description UCM-Webex Calling trunk

    Defines a VoIP dial-peer with a tag 300 and gives a meaningful description for ease of management and troubleshooting.

    destination-pattern BAD.BAD

    A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. Any valid destination pattern may be used in this case.

    session protocol sipv2

    Specifies that dial-peer 300 handles SIP call legs. For more information, see  session protocol (dial-peer).

    session target dns:wxtocucm.io

    Defines the session target of multiple Unified CM nodes through DNS SRV resolution. In this case, the locally defined SRV record wxtocucm.io is used to direct calls.

    incoming uri via 300

    Uses voice class URI 300 to direct all incoming traffic from Unified CM using source port 5065 to this dial-peer. For more information, see  incoming uri.

    voice-class codec 100

    Indicates codec filter list for calls to and from Unified CM. For more information, see  voice class codec.

    bind control source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see  bind.

    bind media source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for media sent to PSTN. For more information, see  bind.

    dtmf-relay rtp-nte

    Defines RTP-NTE (RFC2833) as the DTMF capability expected on the call leg. For more information, see  DTMF Relay (Voice over IP).

    no vad

    Disables voice activity detection. For more information, see  vad (dial peer).

  2. Dial-peer for calls between Unified CM and the PSTN:

    
    dial-peer voice 400 voip
     description UCM-PSTN trunk
     destination-pattern BAD.BAD
     session protocol sipv2
     session target dns:pstntocucm.io
     incoming uri via 400
     voice-class codec 100 
     voice-class sip bind control source-interface GigabitEthernet 0/0/0
     voice-class sip bind media source-interface GigabitEthernet 0/0/0
     dtmf-relay rtp-nte
     no vad
    

    Here's an explanation of the fields for the configuration:

    
    dial-peer voice 400 voip
     description UCM-PSTN trunk

    Defines a VoIP dial-peer with a tag of 400 and gives a meaningful description for ease of management and troubleshooting.

    destination-pattern BAD.BAD

    A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. Any valid destination pattern may be used in this case.

    session protocol sipv2

    Specifies that dial-peer 400 handles SIP call legs. For more information, see  session protocol (dial-peer).

    session target dns:pstntocucm.io

    Defines the session target of multiple Unified CM nodes through DNS SRV resolution. In this case, the locally defined SRV record pstntocucm.io is used to direct calls.

    incoming uri via 400

    Uses voice class URI 400 to direct all incoming traffic from the specified Unified CM hosts using source port 5060 to this dial-peer. For more information, see  incoming uri.

    voice-class codec 100

    Indicates codec filter list for calls to and from Unified CM. For more information, see  voice class codec.

    bind control source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see  bind.

    bind media source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for media sent to PSTN. For more information, see  bind.

    dtmf-relay rtp-nte

    Defines RTP-NTE (RFC2833) as the DTMF capability expected on the call leg. For more information, see  DTMF Relay (Voice over IP).

    no vad

    Disables voice activity detection. For more information, see  vad (dial peer).

4

Add call routing using the following configurations:

  1. Create dial-peer groups to route calls between Unified CM and Webex Calling. Define DPG 100 with outbound dial-peer 100 towards Webex Calling. DPG 100 is applied to the associated incoming dial-peer from Unified CM. Similarly, define DPG 300 with outbound dial-peer 300 toward Unified CM. DPG 300 is applied to the incoming dial-peer from Webex.

    
    voice class dpg 100
     description Route calls to Webex Calling
     dial-peer 100
    voice class dpg 300
     description Route calls to Unified CM Webex Calling trunk
     dial-peer 300 
  2. Create a dial-peer groups to route calls between Unified CM and the PSTN. Define DPG 200 with outbound dial-peer 200 toward the PSTN. DPG 200 is applied to the associated incoming dial-peer from Unified CM. Similarly, define DPG 400 with outbound dial-peer 400 toward Unified CM. DPG 400 is applied to the incoming dial-peer from the PSTN.

    
    voice class dpg 200
     description Route calls to PSTN
     dial-peer 200
    voice class dpg 400
     description Route calls to Unified CM PSTN trunk
     dial-peer 400

    Here's an explanation of the fields for the configuration:

    dial-peer  100

    Associates an outbound dial-peer with a dial-peer group. For more information, see  voice-class dpg.

  3. Apply dial-peer groups to route calls from Webex to Unified CM and from Unified CM to Webex:

    
    dial-peer voice 100
     destination dpg 300
    dial-peer voice 300
     destination dpg 100

    Here's an explanation of the fields for the configuration:

    destination dpg 300

    Specifies which dial-peer group, and therefore dial-peer should be used for the outbound treatment for calls presented to this incoming dial-peer.

  4. Apply dial-peer groups to route calls from the PSTN to Unified CM and from Unified CM to the PSTN:

    
    dial-peer voice 200
     destination dpg 400
    dial-peer voice 400
     destination dpg 200 

    This concludes your Local Gateway configuration. Save the configuration and reload the platform if this is the first time CUBE features have been configured.

Diagnostic Signatures (DS) proactively detects commonly observed issues in the IOS XE-based Local Gateway and generates email, syslog, or terminal message notification of the event. You can also install the DS to automate diagnostics data collection and transfer-collected data to the Cisco TAC case to accelerate resolution time.

Diagnostic Signatures (DS) are XML files that contain information about problem trigger events and actions to be taken to inform, troubleshoot, and remediate the issue. You can define the problem detection logic using syslog messages, SNMP events and through periodic monitoring of specific show command outputs.

The action types include collecting show command outputs:

  • Generating a consolidated log file

  • Uploading the file to a user-provided network location such as HTTPS, SCP, FTP server.

TAC engineers author the DS files and digitally sign it for integrity protection. Each DS file has a unique numerical ID assigned by the system. Diagnostic Signatures Lookup Tool (DSLT) is a single source to find applicable signatures for monitoring and troubleshooting various problems.

Before you begin:

  • Do not edit the DS file that you download from DSLT. The files that you modify fail installation due to the integrity check error.

  • A Simple Mail Transfer Protocol (SMTP) server you require for the Local Gateway to send out email notifications.

  • Ensure that the Local Gateway is running IOS XE 17.6.1 or higher if you wish to use the secure SMTP server for email notifications.

Prerequisites

Local Gateway running IOS XE 17.6.1a or higher

  1. Diagnostic Signatures is enabled by default.

  2. Configure the secure email server to be used to send proactive notification if the device is running Cisco IOS XE 17.6.1a or higher.

    configure terminal 
    call-home  
    mail-server <username>:<pwd>@<email server> priority 1 secure tls 
    end 

  3. Configure the environment variable ds_email with the email address of the administrator to notify you.

    configure terminal 
    call-home  
    diagnostic-signature 
    environment ds_email <email address> 
    end 

The following shows an example configuration of a Local Gateway running on Cisco IOS XE 17.6.1a or higher to send the proactive notifications to tacfaststart@gmail.com using Gmail as the secure SMTP server:


 

We recommend you to use the Cisco IOS XE Bengaluru 17.6.x or later versions.

call-home  
mail-server tacfaststart:password@smtp.gmail.com priority 1 secure tls 
diagnostic-signature 
environment ds_email "tacfaststart@gmail.com" 

 

A Local Gateway running on Cisco IOS XE Software is not a typical web-based Gmail client that supports OAuth, so we must configure a specific Gmail account setting and provide specific permission to have the email from the device processed correctly:

  1. Go to Manage Google Account > Security and turn on the Less secure app access setting.

  2. Answer “Yes, it was me” when you receive an email from Gmail stating “Google prevented someone from signing into your account using a non-Google app.”

Install diagnostic signatures for proactive monitoring

Monitoring high CPU utilization

This DS tracks CPU utilization for five seconds using the SNMP OID 1.3.6.1.4.1.9.2.1.56. When the utilization reaches 75% or more, it disables all debugs and uninstalls all diagnostic signatures that are installed in the Local Gateway. Use these steps below to install the signature.

  1. Use the show snmp command to enable SNMP. If you do not enable, then configure the snmp-server manager command.

    show snmp 
    %SNMP agent not enabled 
    
    config t 
    snmp-server manager 
    end 
    
    show snmp 
    Chassis: ABCDEFGHIGK 
    149655 SNMP packets input 
        0 Bad SNMP version errors 
        1 Unknown community name 
        0 Illegal operation for community name supplied 
        0 Encoding errors 
        37763 Number of requested variables 
        2 Number of altered variables 
        34560 Get-request PDUs 
        138 Get-next PDUs 
        2 Set-request PDUs 
        0 Input queue packet drops (Maximum queue size 1000) 
    158277 SNMP packets output 
        0 Too big errors (Maximum packet size 1500) 
        20 No such name errors 
        0 Bad values errors 
        0 General errors 
        7998 Response PDUs 
        10280 Trap PDUs 
    Packets currently in SNMP process input queue: 0 
    SNMP global trap: enabled 
    
  2. Download DS 64224 using the following drop-down options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series or Cisco CSR 1000V Series

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Performance

    Problem Type

    High CPU Utilization with Email Notification.

  3. Copy the DS XML file to the Local Gateway flash.

    LocalGateway# copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash: 

    The following example shows copying the file from an FTP server to the Local Gateway.

    copy ftp://user:pwd@192.0.2.12/DS_64224.xml bootflash: 
    Accessing ftp://*:*@ 192.0.2.12/DS_64224.xml...! 
    [OK - 3571/4096 bytes] 
    3571 bytes copied in 0.064 secs (55797 bytes/sec) 
    
  4. Install the DS XML file in the Local Gateway.

    call-home diagnostic-signature load DS_64224.xml 
    Load file DS_64224.xml success 
  5. Use the show call-home diagnostic-signature command to verify that the signature is successfully installed. The status column should have a “registered” value.

    show call-home diagnostic-signature  
    Current diagnostic-signature settings: 
    Diagnostic-signature: enabled 
    Profile: CiscoTAC-1 (status: ACTIVE) 
    Downloading  URL(s):  https://tools.cisco.com/its/service/oddce/services/DDCEService 
    Environment variable: 
    ds_email: username@gmail.com 

    Download DSes:

    DS ID

    DS Name

    Revision

    Status

    Last Update (GMT+00:00)

    64224

    DS_LGW_CPU_MON75

    0.0.10

    Registered

    2020-11-07 22:05:33


     

    When triggered, this signature uninstalls all running DSs including itself. If necessary, reinstall DS 64224 to continue monitoring high CPU utilization on the Local Gateway.

Monitoring SIP trunk registration

This DS checks for unregistration of a Local Gateway SIP Trunk with Webex Calling cloud every 60 seconds. Once the unregistration event is detected, it generates an email and syslog notification and uninstalls itself after two unregistration occurrences. Use the steps below to install the signature:

  1. Download DS 64117 using the following drop-down options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series or Cisco CSR 1000V Series

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    SIP-SIP

    Problem Type

    SIP Trunk Unregistration with Email Notification.

  2. Copy the DS XML file to the Local Gateway.

    copy ftp://username:password@<server name or ip>/DS_64117.xml bootflash: 
  3. Install the DS XML file in the Local Gateway.

    call-home diagnostic-signature load DS_64117.xml 
    Load file DS_64117.xml success 
    LocalGateway#  
  4. Use the show call-home diagnostic-signature command to verify that the signature is successfully installed. The status column must have a “registered” value.

Monitoring abnormal call disconnects

This DS uses SNMP polling every 10 minutes to detect abnormal call disconnect with SIP errors 403, 488 and 503.  If the error count increment is greater than or equal to 5 from the last poll, it generates a syslog and email notification. Please use the steps below to install the signature.

  1. Use the show snmp command to check whether SNMP is enabled. If it is not enabled, configure the snmp-server manager command.

    show snmp 
    %SNMP agent not enabled 
     
    
    config t 
    snmp-server manager 
    end 
    
    show snmp 
    Chassis: ABCDEFGHIGK 
    149655 SNMP packets input 
        0 Bad SNMP version errors 
        1 Unknown community name 
        0 Illegal operation for community name supplied 
        0 Encoding errors 
        37763 Number of requested variables 
        2 Number of altered variables 
        34560 Get-request PDUs 
        138 Get-next PDUs 
        2 Set-request PDUs 
        0 Input queue packet drops (Maximum queue size 1000) 
    158277 SNMP packets output 
        0 Too big errors (Maximum packet size 1500) 
        20 No such name errors 
        0 Bad values errors 
        0 General errors 
        7998 Response PDUs 
        10280 Trap PDUs 
    Packets currently in SNMP process input queue: 0 
    SNMP global trap: enabled 
    
  2. Download DS 65221 using the following options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series or Cisco CSR 1000V Series

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Performance

    Problem Type

    SIP abnormal call disconnect detection with Email and Syslog Notification.

  3. Copy the DS XML file to the Local Gateway.

    copy ftp://username:password@<server name or ip>/DS_65221.xml bootflash:
  4. Install the DS XML file in the Local Gateway.

    call-home diagnostic-signature load DS_65221.xml 
    Load file DS_65221.xml success 
    
  5. Use the show call-home diagnostic-signature command to verify that the signature is successfully installed. The status column must have a “registered” value.

Install diagnostic signatures to troubleshoot a problem

Use Diagnostic Signatures (DS) to resolve issues quickly. Cisco TAC engineers have authored several signatures that enable the necessary debugs that are required to troubleshoot a given problem, detect the problem occurrence, collect the right set of diagnostic data and transfer the data automatically to the Cisco TAC case. Diagnostic Signatures (DS) eliminate the need to manually check for the problem occurrence and makes troubleshooting of intermittent and transient issues a lot easier.

You can use the Diagnostic Signatures Lookup Tool to find the applicable signatures and install them to self-solve a given issue or you can install the signature that is recommended by the TAC engineer as part of the support engagement.

Here is an example of how to find and install a DS to detect the occurrence “%VOICE_IEC-3-GW: CCAPI: Internal Error (call spike threshold): IEC=1.1.181.1.29.0" syslog and automate diagnostic data collection using the following steps:

  1. Configure an additional DS environment variable ds_fsurl_prefix which is the Cisco TAC file server path (cxd.cisco.com) to which the collected diagnostics data are uploaded. The username in the file path is the case number and the password is the file upload token which can be retrieved from Support Case Manager in the following command. The file upload token can be generated in the Attachments section of the Support Case Manager, as needed.

    configure terminal 
    call-home  
    diagnostic-signature 
    LocalGateway(cfg-call-home-diag-sign)environment ds_fsurl_prefix "scp://<case number>:<file upload token>@cxd.cisco.com"  
    end 

    Example:

    call-home  
    diagnostic-signature 
    environment ds_fsurl_prefix " environment ds_fsurl_prefix "scp://612345678:abcdefghijklmnop@cxd.cisco.com"  
  2. Ensure that SNMP is enabled using the show snmp command. If it is not enabled, configure the snmp-server manager command.

    show snmp 
    %SNMP agent not enabled 
     
     
    config t 
    snmp-server manager 
    end 
  3. Ensure to install the High CPU monitoring DS 64224 as a proactive measure to disable all debugs and diagnostics signatures during the time of high CPU utilization. Download DS 64224 using the following options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series or Cisco CSR 1000V Series

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Performance

    Problem Type

    High CPU Utilization with Email Notification.

  4. Download DS 65095 using the following options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series or Cisco CSR 1000V Series

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Syslogs

    Problem Type

    Syslog - %VOICE_IEC-3-GW: CCAPI: Internal Error (Call spike threshold): IEC=1.1.181.1.29.0

  5. Copy the DS XML files to the Local Gateway.

    copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash: 
    copy ftp://username:password@<server name or ip>/DS_65095.xml bootflash: 
  6. Install the High CPU monitoring DS 64224 and then DS 65095 XML file in the Local Gateway.

    call-home diagnostic-signature load DS_64224.xml 
    Load file DS_64224.xml success 
     
    call-home diagnostic-signature load DS_65095.xml 
    Load file DS_65095.xml success 
    
  7. Verify that the signature is successfully installed using the show call-home diagnostic-signature command. The status column must have a “registered” value.

    show call-home diagnostic-signature  
    Current diagnostic-signature settings: 
    Diagnostic-signature: enabled 
    Profile: CiscoTAC-1 (status: ACTIVE) 
    Downloading  URL(s):  https://tools.cisco.com/its/service/oddce/services/DDCEService 
    Environment variable: 
               ds_email: username@gmail.com 
               ds_fsurl_prefix: scp://612345678:abcdefghijklmnop@cxd.cisco.com 

    Downloaded DSes:

    DS ID

    DS Name

    Revision

    Status

    Last Update (GMT+00:00)

    64224

    00:07:45

    DS_LGW_CPU_MON75

    0.0.10

    Registered

    2020-11-08

    65095

    00:12:53

    DS_LGW_IEC_Call_spike_threshold

    0.0.12

    Registered

    2020-11-08

Verify diagnostic signatures execution

In the following command, the “Status” column of the show call-home diagnostic-signature command changes to “running” while the Local Gateway executes the action defined within the signature. The output of show call-home diagnostic-signature statistics is the best way to verify whether a diagnostic signature detects an event of interest and executes the action. The “Triggered/Max/Deinstall” column indicates the number of times the given signature has triggered an event, the maximum number of times it is defined to detect an event and whether the signature deinstalls itself after detecting the maximum number of triggered events.

show call-home diagnostic-signature  
Current diagnostic-signature settings: 
Diagnostic-signature: enabled 
Profile: CiscoTAC-1 (status: ACTIVE) 
Downloading  URL(s):  https://tools.cisco.com/its/service/oddce/services/DDCEService 
Environment variable: 
           ds_email: carunach@cisco.com 
           ds_fsurl_prefix: scp://612345678:abcdefghijklmnop@cxd.cisco.com 

Downloaded DSes:

DS ID

DS Name

Revision

Status

Last Update (GMT+00:00)

64224

DS_LGW_CPU_MON75

0.0.10

Registered

2020-11-08 00:07:45

65095

DS_LGW_IEC_Call_spike_threshold

0.0.12

Running

2020-11-08 00:12:53

show call-home diagnostic-signature statistics

DS ID

DS Name

Triggered/Max/Deinstall

Average Run Time (seconds)

Max Run Time (seconds)

64224

DS_LGW_CPU_MON75

0/0/N

0.000

0.000

65095

DS_LGW_IEC_Call_spike_threshold

1/20/Y

23.053

23.053

The notification email that is sent during diagnostic signature execution contains key information such as issue type, device details, software version, running configuration, and show command outputs that are relevant to troubleshoot the given problem.

Uninstall diagnostic signatures

Use Diagnostic signatures for troubleshooting purposes are typically defined to uninstall after detection of some problem occurrences. If you want to uninstall a signature manually, retrieve the DS ID from the output of the show call-home diagnostic-signature command and run the following command:

call-home diagnostic-signature deinstall <DS ID> 

Example:

call-home diagnostic-signature deinstall 64224 

 

New signatures are added to Diagnostics Signatures Lookup Tool periodically, based on issues that are commonly observed in deployments. TAC currently doesn’t support requests to create new custom signatures.

For better management of Cisco IOS XE Gateways, we recommend that you enroll and manage the gateways through the Control Hub. It is an optional configuration. When enrolled, you can use the configuration validation option in the Control Hub to validate your Local Gateway configuration and identify any configuration issues. Currently, only registration-based trunks support this functionality.

For more information, refer the following:

This section describes how to configure a Cisco Unified Border Element (CUBE) as a Local Gateway for Webex Calling, using certificate-based mutual TLS (mTLS) SIP trunk. The first part of this document illustrates how to configure a simple PSTN gateway. In this case, all calls from the PSTN are routed to Webex Calling and all calls from Webex Calling are routed to the PSTN. The following image highlights this solution and the high-level call routing configuration that will be followed.

In this design, the following principal configurations are used:

  • voice class tenants: Used to create trunk specific configurations.

  • voice class uri: Used to classify SIP messages for the selection of an inbound dial-peer.

  • inbound dial-peer: Provides treatment for inbound SIP messages and determines the outbound route with a dial-peer group.

  • dial-peer group: Defines the outbound dial-peers used for onward call routing.

  • outbound dial-peer: Provides treatment for outbound SIP messages and routes them to the required target.

Call routing from/to PSTN to/from Webex Calling configuration solution

When connecting an on-premises Cisco Unified Communications Manager solution with Webex Calling, you can use the simple PSTN gateway configuration as a baseline for building the solution illustrated in the following diagram. In this case, Unified Communications Manager provides centralized routing and treatment of all PSTN and Webex Calling calls.

Throughout this document, the host names, IP addresses, and interfaces illustrated in the following image are used. Options are provided for public or private (behind NAT) addressing. SRV DNS records are optional, unless load balancing across multiple CUBE instances.

Use the configuration guidance in the rest of this document to complete your Local Gateway configuration as follows:

  • Step 1: Configure router baseline connectivity and security

  • Step 2: Configure Webex Calling Trunk

    Depending on your required architecture, follow either:

  • Step 3: Configure Local Gateway with SIP PSTN trunk

  • Step 4: Configure Local Gateway with existing Unified CM environment

    Or:

  • Step 3: Configure Local Gateway with TDM PSTN trunk

Baseline configuration

The first step in preparing your Cisco router as a Local Gateway for Webex Calling is to build a baseline configuration that secures your platform and establishes connectivity.

  • All certificate-based Local Gateway deployments require Cisco IOS XE 17.9.1a or later versions. For the recommended versions, see the Cisco Software Research page. Search for the platform and select one of the suggested releases.

    • ISR4000 series routers must be configured with both Unified Communications and Security technology licenses.

    • Catalyst Edge 8000 series routers fitted with voice cards or DSPs require DNA Essentials licensing. Routers without voice cards or DSPs require a minimum of DNA Essentials licensing.

    • For high-capacity requirements, you may also require a High Security (HSEC) license and additional throughput entitlement.

      Refer to Authorization Codes for further details.

  • Build a baseline configuration for your platform that follows your business policies. In particular, configure the following and verify the working:

    • NTP

    • ACLs

    • User authentication and remote access

    • DNS

    • IP routing

    • IP addresses

  • The network toward Webex Calling must use a IPv4 address. Local Gateway Fully Qualified Domain Names (FQDN) or Service Record (SRV) addresses must resolve to a public IPv4 address on the internet.

  • All SIP and media ports on the Local Gateway interface facing Webex must be accessible from the internet, either directly or via static NAT. Ensure that you update your firewall accordingly.

  • Install a signed certificate on the Local Gateway (the following provides detailed configuration steps).

    • A public Certificate Authority (CA) as detailed in  What Root Certificate Authorities are Supported for Calls to Cisco Webex Audio and Video Platforms? must sign the device certificate.

    • The FQDN configured in the Control Hub when creating a trunk must be the Common Name (CN) or Subject Alternate Name (SAN) certificate of the router. For example:

      • If a configured trunk in the Control Hub of your organization has cube1.lgw.com:5061 as FQDN of the Local Gateway, then the CN or SAN in the router certificate must contain cube1.lgw.com. 

      • If a configured trunk in the Control Hub of your organization has lgws.lgw.com as the SRV address of the Local Gateway(s) reachable from the trunk, then the CN or SAN in the router certificate must contain lgws.lgw.com. The records that the SRV address resolves to (CNAME, A Record, or IP Address) are optional in SAN.

      • Whether you use an FQDN or SRV for the trunk, the contact address for all new SIP dialogs from your Local Gateway uses the name configured in the Control Hub.

  • Ensure that certificates are signed for client and server usage.

  • Upload the Cisco root CA bundle to the Local Gateway.

Configuration

1

Ensure that you assign valid and routable IP addresses to any Layer 3 interfaces, for example:


interface GigabitEthernet0/0/0
 description Interface facing PSTN and/or CUCM
 ip address 192.168.80.14 255.255.255.0
!
interface GigabitEthernet0/0/1
 description Interface facing Webex Calling (Public address)
 ip address 198.51.100.1 255.255.255.240

2

Protect STUN credentials on the router using symmetric encryption. Configure the primary encryption key and encryption type as follows:


key config-key password-encrypt YourPassword
password encryption aes
3

Create an encryption trustpoint with a certificate signed by your preferred Certificate Authority (CA).

  1. Create an RSA key pair using the following exec command.

    crypto key generate rsa general-keys exportable label lgw-key modulus 4096

  2. Create a trustpoint for the signed certificate with the following configuration commands:

    
    crypto pki trustpoint LGW_CERT
     enrollment terminal pem
     fqdn cube1.lgw.com
     subject-name cn=cube1.lgw.com
     subject-alt-name cube1.lgw.com
     revocation-check none
     rsakeypair lgw-key

  3. Generate Certificate Signing Request (CSR) with the following exec or configuration command and use it to request a signed certificate from a supported CA provider:

    crypto pki enroll LGW_CERT

4

Authenticate your new certificate using your intermediate (or root) CA certificate, then import the certificate (Step 4). Enter the following exec or configuration command:


crypto pki authenticate LGW_CERT
<paste Intermediate X.509 base 64 based certificate here>

5

Import a signed host certificate using the following exec or configuration command:


crypto pki import LGW_CERT certificate
<paste CUBE host X.509 base 64 certificate here>

6

Enable TLS1.2 exclusivity and specify the default trustpoint using the following configuration commands:


 sip-ua
  crypto signaling default trustpoint LGW_CERT
  transport tcp tls v1.2

7

Install the Cisco root CA bundle, which includes the DigiCert CA certificate used by Webex Calling. Use the crypto pki trustpool import clean url command to download the root CA bundle from the specified URL, and to clear the current CA trustpool, then install the new bundle of certificates:


 

If you need to use a proxy for access to the internet using HTTPS, add the following configuration before importing the CA bundle:

ip http client proxy-server yourproxy.com proxy-port 80

ip http client source-interface GigabitEthernet0/0/1 
crypto pki trustpool import clean url https://www.cisco.com/security/pki/trs/ios_core.p7b
1

Create a CUBE certificate-based PSTN trunk for an existing location in Control Hub. For more information, see Configure trunks, route groups, and dial plans for Webex Calling.


 
Make a note of the trunk information that is provided once the trunk is created. These details, as highlighted in the following illustration, will be used in the configuration steps in this guide.
2

Enter the following commands to configure CUBE as a Webex Calling Local Gateway:


voice service voip
 ip address trusted list
  ipv4 x.x.x.x y.y.y.y
 mode border-element
 allow-connections sip to sip
 no supplementary-service sip refer
 stun
  stun flowdata agent-id 1 boot-count 4
  stun flowdata shared-secret 0 Password123$
 sip 
  asymmetric payload full
  early-offer forced
  sip-profiles inbound

Here's an explanation of the fields for the configuration:


ip address trusted list
 ipv4 x.x.x.x y.y.y.y
  • To protect against toll fraud, the trusted address list defines a list of hosts and networks entities from which the Local Gateway expects legitimate VoIP calls.

  • By default, Local Gateway blocks all incoming VoIP messages from IP addresses not in its trusted list. Statically configured dial-peers with “session target IP” or server group IP addresses are trusted by default so do not need to be added to the trusted list.

  • When configuring your Local Gateway, add the IP subnets for your regional Webex Calling data center to the list, see Port Reference Information for Webex Calling for more information. Also, add address ranges for Unified Communications Manager servers (if used) and PSTN trunk gateways.

  • For more information on how to use an IP address trusted list to prevent toll fraud, see IP address trusted.

mode border-element

Enables Cisco Unified Border Element (CUBE) features on the platform.

allow-connections sip to sip

Enable CUBE basic SIP back to back user agent functionality. For more information, see Allow connections.


 

By default, T.38 fax transport is enabled. For more information, see fax protocol t38 (voice-service).

stun

Enables STUN (Session Traversal of UDP through NAT) globally.


 
These global stun commands are only required when deploying your Local Gateway behind NAT.
  • When you forward a call to a Webex Calling user (for example, both the called and calling parties are Webex Calling subscribers and if you anchor media at the Webex Calling SBC), then the media cannot flow to the Local Gateway as the pinhole isn't open.

  • The STUN bindings feature on the Local Gateway allows locally generated STUN requests to be sent over the negotiated media path. This helps to open the pinhole in the firewall.

For more information, see  stun flowdata agent-id and  stun flowdata shared-secret.

asymmetric payload full

Configures SIP asymmetric payload support for both DTMF and dynamic codec payloads. For more information on this command, see asymmetric payload.

early-offer forced

Forces the Local Gateway to send SDP information in the initial INVITE message instead of waiting for acknowledgment from the neighboring peer. For more information on this command, see early-offer.

sip-profiles inbound

Enables CUBE to use SIP profiles to modify messages as they are received. Profiles are applied via dial-peers or tenants.

3

Configure voice class codec 100 codec filter for the trunk. In this example, the same codec filter is used for all trunks. You can configure filters for each trunk for precise control.


voice class codec 100
 codec preference 1 opus
 codec preference 2 g711ulaw
 codec preference 3 g711alaw

Here's an explanation of the fields for the configuration:

voice class codec 100

Used to only allow preferred codecs for calls through SIP trunks. For more information, see voice class codec.


 

Opus codec is supported only for SIP-based PSTN trunks. If the PSTN trunk uses a voice T1/E1 or analog FXO connection, exclude codec preference 1 opus from the voice class codec 100 configuration.

4

Configure voice class stun-usage 100 to enable ICE on the Webex Calling trunk.


voice class stun-usage 100 
 stun usage firewall-traversal flowdata
 stun usage ice lite

Here's an explanation of the fields for the configuration:

stun usage ice lite

Used to enable ICE-Lite for all Webex Calling facing dial-peers to allow media-optimization whenever possible. For more information, see voice class stun usage and stun usage ice lite.


 
The stun usage firewall-traversal flowdata command is only required when deploying your Local Gateway behind NAT.

 
You require stun usage of ICE-lite for call flows using media path optimization. To provide media-optimization for a SIP to TDM gateway, configure a loopback dial-peer with ICE-Lite enabled on the IP-IP leg. For further technical details, contact the Account or TAC teams.
5

Configure the media encryption policy for Webex traffic. (This step is not applicable for Webex for Government)


voice class srtp-crypto 100
 crypto 1 AES_CM_128_HMAC_SHA1_80

Here's an explanation of the fields for the configuration:

voice class srtp-crypto 100

Specifies SHA1_80 as the only SRTP cipher-suite CUBE offers in the SDP in offer and answer messages. Webex Calling only supports SHA1_80. For more information, see voice class srtp-crypto.

6

Configure FIPS-compliant GCM ciphers (This step is applicable only for Webex for Government).


voice class srtp-crypto 100
crypto 1 AEAD_AES_256_GCM

Here's an explanation of the fields for the configuration:

voice class srtp-crypto 100

Specifies GCM as the cipher-suite that CUBE offers. It is mandatory to configure GCM ciphers for Local Gateway for Webex for Government.

7

Configure a pattern to uniquely identify calls to a Local Gateway trunk based on its destination FQDN or SRV:


voice class uri 100 sip
 pattern cube1.lgw.com

Here's an explanation of the fields for the configuration:

voice class uri 100 sip

Defines a pattern to match an incoming SIP invite to an incoming trunk dial-peer. When entering this pattern, use LGW FQDN or SRV configured in Control Hub while creating a trunk.

8

Configure SIP message manipulation profiles. If your gateway is configured with a public IP address, configure a profile as follows or skip to the next step if you are using NAT. In this example, cube1.lgw.com is the FQDN configured for the Local Gateway and "198.51.100.1" is the public IP address of the Local Gateway interface facing Webex Calling:


voice class sip-profiles 100
 rule 10 request ANY sip-header Contact modify "@.*:" "@cube1.lgw.com:" 
 rule 20 response ANY sip-header Contact modify "@.*:" "@cube1.lgw.com:" 
 

Here's an explanation of the fields for the configuration:

rules 10 and 20

To allow Webex to authenticate messages from your local gateway, the 'Contact' header in SIP request and responses messages must contain the value provisioned for the trunk in Control Hub. This will either be the FQDN of a single host, or the SRV domain name used for a cluster of devices.


 

Skip the next step if you have configured your Local Gateway with public IP addresses.

9

If your gateway is configured with a private IP address behind static NAT, configure inbound and outbound SIP profiles as follows. In this example, cube1.lgw.com is the FQDN configured for the Local Gateway, "10.80.13.12" is the interface IP address facing Webex Calling and "192.65.79.20" is the NAT public IP address.

SIP profiles for outbound messages to Webex Calling

voice class sip-profiles 100
 rule 10 request ANY sip-header Contact modify "@.*:" "@cube1.lgw.com:"
 rule 20 response ANY sip-header Contact modify "@.*:" "@cube1.lgw.com:"
 rule 30 response ANY sdp-header Audio-Attribute modify "(a=candidate:1 1.*) 10.80.13.12" "\1 192.65.79.20"
 rule 31 response ANY sdp-header Audio-Attribute modify "(a=candidate:1 2.*) 10.80.13.12" "\1 192.65.79.20"
 rule 40 response ANY sdp-header Audio-Connection-Info modify "IN IP4 10.80.13.12" "IN IP4 192.65.79.20"
 rule 41 request ANY sdp-header Audio-Connection-Info modify "IN IP4 10.80.13.12" "IN IP4 192.65.79.20"
 rule 50 request ANY sdp-header Connection-Info modify "IN IP4 10.80.13.12" "IN IP4 192.65.79.20"
 rule 51 response ANY sdp-header Connection-Info modify "IN IP4 10.80.13.12" "IN IP4 192.65.79.20"
 rule 60 response ANY sdp-header Session-Owner modify "IN IP4 10.80.13.12" "IN IP4 192.65.79.20"
 rule 61 request ANY sdp-header Session-Owner modify "IN IP4 10.80.13.12" "IN IP4 192.65.79.20"
 rule 70 request ANY sdp-header Audio-Attribute modify "(a=rtcp:.*) 10.80.13.12" "\1 192.65.79.20"
 rule 71 response ANY sdp-header Audio-Attribute modify "(a=rtcp:.*) 10.80.13.12" "\1 192.65.79.20
 rule 80 request ANY sdp-header Audio-Attribute modify "(a=candidate:1 1.*) 10.80.13.12" "\1 192.65.79.20"
 rule 81 request ANY sdp-header Audio-Attribute modify "(a=candidate:1 2.*) 10.80.13.12" "\1 192.65.79.20"

Here's an explanation of the fields for the configuration:

rules 10 and 20

To allow Webex to authenticate messages from your local gateway, the 'Contact' header in SIP request and responses messages must contain the value provisioned for the trunk in Control Hub. This will either be the FQDN of a single host, or the SRV domain name used for a cluster of devices.

rules 30 to 81

Convert private address references to the external public address for the site, allowing Webex to correctly interpret and route subsequent messages.

SIP profile for inbound messages from Webex Calling

voice class sip-profiles 110
 rule 10 response ANY sdp-header Video-Connection-Info modify "192.65.79.20" "10.80.13.12"
 rule 20 response ANY sip-header Contact modify "@.*:" "@cube1.lgw.com:"
 rule 30 response ANY sdp-header Connection-Info modify "192.65.79.20" "10.80.13.12"
 rule 40 response ANY sdp-header Audio-Connection-Info modify "192.65.79.20" "10.80.13.12"
 rule 50 response ANY sdp-header Session-Owner modify "192.65.79.20" "10.80.13.12"
 rule 60 response ANY sdp-header Audio-Attribute modify "(a=candidate:1 1.*) 192.65.79.20" "\1 10.80.13.12"
 rule 70 response ANY sdp-header Audio-Attribute modify "(a=candidate:1 2.*) 192.65.79.20" "\1 10.80.13.12"
 rule 80 response ANY sdp-header Audio-Attribute modify "(a=rtcp:.*) 192.65.79.20" "\1 10.80.13.12"

Here's an explanation of the fields for the configuration:

rules 10 to 80

Convert public address references to the configured private address, allowing messages from Webex to be correctly processed by CUBE.

For more information, see voice class sip-profiles.

10

Configure a SIP Options keepalive with header modification profile.


voice class sip-profiles 115
 rule 10 request OPTIONS sip-header Contact modify "<sip:.*:" "<sip:cube1.lgw.com:" 
 rule 30 request ANY sip-header Via modify "(SIP.*) 10.80.13.12" "\1 192.65.79.20"
 rule 40 response ANY sdp-header Connection-Info modify "10.80.13.12" "192.65.79.20"  
 rule 50 response ANY sdp-header Audio-Connection-Info modify "10.80.13.12" "192.65.79.20"
!
voice class sip-options-keepalive 100
 description Keepalive for Webex Calling
 up-interval 5
 transport tcp tls
 sip-profiles 115

Here's an explanation of the fields for the configuration:

voice class sip-options-keepalive 100

Configures a keepalive profile and enters voice class configuration mode. You can configure the time (in seconds) at which an SIP Out of Dialog Options Ping is sent to the dial-target when the heartbeat connection to the endpoint is in UP or Down status.

This keepalive profile is triggered from the dial-peer configured towards Webex.

To ensure that the contact headers include the SBC fully qualified domain name, SIP profile 115 is used. Rules 30, 40, and 50 are required only when the SBC is configured behind static NAT.

In this example, cube1.lgw.com is the FQDN selected for the Local Gateway and if static NAT is used, "10.80.13.12" is the SBC interface IP address towards Webex Calling and "192.65.79.20" is the NAT public IP address.

11

Configure Webex Calling trunk:

  1. Create voice class tenant 100 to define and group configurations required specifically for the Webex Calling trunk. Dial-peers associated with this tenant later will inherit these configurations:


     

    The following example uses the values illustrated in Step 1 for the purpose of this guide (shown in bold). Replace these with values for your trunk in your configuration.

    
    voice class tenant 100
     no remote-party-id
     sip-server dns:us25.sipconnect.bcld.webex.com
     srtp-crypto 100
     localhost dns:cube1.lgw.com
     session transport tcp tls
     no session refresh
     error-passthru
     bind control source-interface GigabitEthernet0/0/1
     bind media source-interface GigabitEthernet0/0/1
     no pass-thru content custom-sdp
     sip-profiles 100 
     sip-profiles 110 inbound
     privacy-policy passthru
    !

    Here's an explanation of the fields for the configuration:

    voice class tenant 100

    We recommend that you use tenants to configure trunks which have their own TLS certificate, and CN or SAN validation list. Here, the tls-profile associated with the tenant contains the trust point to be used to accept or create new connections, and has the CN or SAN list to validate the incoming connections. For more information, see voice class tenant.

    no remote-party-id

    Disable SIP Remote-Party-ID (RPID) header as Webex Calling supports PAI, which is enabled using CIO asserted-id pai. For more information, see remote-party-id.

    sip-server dns:us25.sipconnect.bcld.webex.com

    Configures the target SIP server for the trunk. Use the edge proxy SRV address provided in Control Hub when you created your trunk

    srtp-crypto 100

    Configures the preferred cipher-suites for the SRTP call leg (connection) (specified in Step 5). For more information, see voice class srtp-crypto.

    localhost dns: cube1.lgw.com

    Configures CUBE to replace the physical IP address in the From, Call-ID, and Remote-Party-ID headers in outgoing messages with the provided FQDN.

    session transport tcp tls

    Sets transport to TLS for associated dial-peers. For more information, see session-transport.

    no session refresh

    Disables SIP session refresh globally.

    error-passthru

    Specifies SIP error response pass-thru functionality. For more information, see error-passthru.

    bind control source-interface GigabitEthernet0/0/1

    Configures the source interface and associated IP address for messages sent to Webex Calling. For more information, see bind.

    bind media source-interface GigabitEthernet0/0/1

    Configures the source interface and associated IP address for media sent to Webex Calling. For more information, see bind.

    voice-class sip profiles 100

    Applies the header modification profile (Public IP or NAT addressing) to use for outbound messages. For more information, see voice-class sip profiles.

    voice-class sip profiles 110 inbound

    Applies the header modification profile (NAT addressing only) to use for inbound messages. For more information, see voice-class sip profiles.

    privacy-policy passthru

    Configures the privacy header policy options for the trunk to pass privacy values from the received message to the next call leg. For more information, see privacy-policy.

  2. Configure the Webex Calling trunk dial-peer.

    
    dial-peer voice 100 voip
     description Inbound/Outbound Webex Calling
     destination-pattern BAD.BAD
     session protocol sipv2
     session target sip-server
     incoming uri request 100
     voice-class codec 100
     voice-class stun-usage 100
     voice-class sip rel1xx disable
     voice-class sip asserted-id pai
     voice-class sip tenant 100
     voice-class sip options-keepalive profile 100
     dtmf-relay rtp-nte 
     srtp
     no vad
    

    Here's an explanation of the fields for the configuration:

    
    dial-peer voice 100 voip
     description Inbound/Outbound Webex Calling

    Defines a VoIP dial-peer with a tag of 100 and gives a meaningful description for ease of management and troubleshooting. For more information, see dial-peer voice.

    destination-pattern BAD.BAD

    A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. Any valid destination pattern may be used in this case.

    session protocol sipv2

    Specifies that dial-peer 100 handles SIP call legs. For more information, see session protocol (dial-peer).

    session target sip-server

    Indicates that the SIP server defined in tenant 100 is inherited and used for the destination for calls from this dial peer.

    incoming uri request 100

    To specify the voice class used to match a VoIP dial peer to the uniform resource identifier (URI) of an incoming call. For more information, see  incoming uri.

    voice-class codec 100

    Indicates codec filter list for calls to and from Webex Calling. For more information, see voice class codec.

    voice-class stun-usage 100

    Allows locally generated STUN requests on the Local Gateway to be sent over the negotiated media path. STUN help to open a firewall pinhole for media traffic.

    voice-class sip asserted-id pai

    Sets the outgoing calling information using the privacy asserted ID (PAI) header. For more information, see voice-class sip asserted-id.

    voice-class sip tenant 100

    The dial-peer inherits all parameters configured globally and in tenant 100. Parameters may overridden at the dial-peer level. For more information, see  voice-class sip tenant.

    voice-class sip options-keepalive profile 100

    This command is used to monitor the availability of a group of SIP servers or endpoints using a specific profile (100).

    srtp

    Enables SRTP for the call leg.

Having built a trunk towards Webex Calling above, use the following configuration to create a non-encrypted trunk towards a SIP based PSTN provider:


 

If your Service Provider offers a secure PSTN trunk, you may follow a similar configuration as detailed above for the Webex Calling trunk. Secure to secure call routing is supported by CUBE.


 

To configure TDM interfaces for PSTN call legs on the Cisco TDM-SIP Gateways, see  Configuring ISDN PRI.

1

Configure the following voice class uri to identify inbound calls from the PSTN trunk:


voice class uri 200 sip
  host ipv4:192.168.80.13

Here's an explanation of the fields for the configuration:

voice class uri 200 sip

Defines a pattern to match an incoming SIP invite to an incoming trunk dial-peer. When entering this pattern, use the IP address of you IP PSTN gateway. For more information, see  voice class uri.

2

Configure the following IP PSTN dial-peer:


dial-peer voice 200 voip
 description Inbound/Outbound IP PSTN trunk
 destination-pattern BAD.BAD
 session protocol sipv2
 session target ipv4:192.168.80.13
 incoming uri via 200
 voice-class sip bind control source-interface GigabitEthernet0/0/0 
 voice-class sip bind media source-interface  GigabitEthernet0/0/0 
 voice-class codec 100
 dtmf-relay rtp-nte 
 no vad

Here's an explanation of the fields for the configuration:


dial-peer voice 200 voip
 description Inbound/Outbound IP PSTN trunk

Defines a VoIP dial-peer with a tag of 200 and gives a meaningful description for ease of management and troubleshooting. For more information, see dial-peer voice.

destination-pattern BAD.BAD

A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface).

session protocol sipv2

Specifies that dial-peer 200 handles SIP call legs. For more information, see session protocol (dial peer).

session target ipv4:192.168.80.13

Indicates the destination’s target IPv4 address to send the call leg. The session target here is ITSP’s IP address. For more information, see  session target (VoIP dial peer).

incoming uri via 200

Defines a match criterion for the VIA header with the IP PSTN’s IP address. Matches all incoming IP PSTN call legs on the Local Gateway with dial-peer 200. For more information, see  incoming url.

bind control source-interface GigabitEthernet0/0/0

Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see  bind.

bind media source-interface GigabitEthernet0/0/0

Configures the source interface and associated IP address for media sent to PSTN. For more information, see  bind.

voice-class codec 100

Configures the dial-peer to use the common codec filter list 100. For more information, see voice-class codec.

dtmf-relay rtp-nte

Defines RTP-NTE (RFC2833) as the DTMF capability expected on the call leg. For more information, see DTMF Relay (Voice over IP).

no vad

Disables voice activity detection. For more information, see vad (dial peer).

3

If you are configuring your Local Gateway to only route calls between Webex Calling and the PSTN, add the following call routing configuration. If you are configuring your Local Gateway with a Unified Communications Manager platform, skip to the next section.

  1. Create dial-peer groups to route calls towards Webex Calling or the PSTN. Define DPG 100 with outbound dial-peer 100 toward Webex Calling. DPG 100 is applied to the incoming dial-peer from the PSTN. Similarly, define DPG 200 with outbound dial-peer 200 toward the PSTN. DPG 200 is applied to the incoming dial-peer from Webex.

    
    voice class dpg 100 
     description Route calls to Webex Calling 
     dial-peer 100 
    voice class dpg 200 
     description Route calls to PSTN 
     dial-peer 200

    Here's an explanation of the fields for the configuration:

    dial-peer 100

    Associates an outbound dial-peer with a dial-peer group. For more information, see  voice-class dpg.

  2. Apply dial-peer groups to route calls from Webex to the PSTN and from the PSTN to Webex:

    
    dial-peer voice 100
     destination dpg 200
    dial-peer voice 200
     destination dpg 100 

    Here's an explanation of the fields for the configuration:

    destination dpg 200

    Specifies which dial-peer group, and therefore dial-peer should be used for the outbound treatment for calls presented to this incoming dial-peer.

    This concludes your Local Gateway configuration. Save the configuration and reload the platform if this is the first time CUBE features are configured.

The PSTN-Webex Calling configuration in the previous sections may be modified to include additional trunks to a Cisco Unified Communications Manager (UCM) cluster. In this case, all calls are routed via Unified CM. Calls from UCM on port 5060 are routed to the PSTN and calls from port 5065 are routed to Webex Calling. The following incremental configurations may be added to include this calling scenario.

1

Configure the following voice class URIs:

  1. Classifies Unified CM to Webex calls using SIP VIA port:

    
    voice class uri 300 sip
     pattern :5065
    
  2. Classifies Unified CM to PSTN calls using SIP via port:

    
    voice class uri 400 sip
     pattern :192\.168\.80\.6[0-5]:5060
    

    Classify incoming messages from the UCM towards the PSTN trunk using one or more patterns that describe the originating source addresses and port number. Regular expressions may be used to define matching patterns if required.

    In the example above, a regular expression is used to match any IP address in the range 192.168.80.60 to 65 and port number 5060.

2

Configure the following DNS records to specify SRV routing to Unified CM hosts:


 

IOS XE uses these records for locally determining target UCM hosts and ports. With this configuration, it is not required to configure records in your DNS system. If you prefer to use your DNS, then these local configurations are not required.


ip host ucmpub.mydomain.com 192.168.80.60
ip host ucmsub1.mydomain.com 192.168.80.61
ip host ucmsub2.mydomain.com 192.168.80.62
ip host ucmsub3.mydomain.com 192.168.80.63
ip host ucmsub4.mydomain.com 192.168.80.64
ip host ucmsub5.mydomain.com 192.168.80.65
ip host _sip._udp.wxtocucm.io srv 0 1 5065 ucmpub.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub1.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub2.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub3.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub4.mydomain.com
ip host _sip._udp.wxtocucm.io srv 2 1 5065 ucmsub5.mydomain.com
ip host _sip._udp.pstntocucm.io srv 0 1 5060 ucmpub.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub1.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub2.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub3.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub4.mydomain.com
ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub5.mydomain.com

Here's an explanation of the fields for the configuration:

The following command creates a DNS SRV resource record. Create a record for each UCM host and trunk:

ip host _sip._udp.pstntocucm.io srv 2 1 5060 ucmsub5.mydomain.com

_sip._udp.pstntocucm.io: SRV resource record name

2: The SRV resource record priority

1: The SRV resource record weight

5060: The port number to use for the target host in this resource record

ucmsub5.mydomain.com: The resource record target host

To resolve the resource record target host names, create local DNS A records. For example:

ip host ucmsub5.mydomain.com 192.168.80.65

ip host: Creates a record in the local IOS XE database.

ucmsub5.mydomain.com: The A record host name.

192.168.80.65: The host IP address.

Create the SRV resource records and A records to reflect your UCM environment and preferred call distribution strategy.

3

Configure the following dial-peers:

  1. Dial-peer for calls between Unified CM and Webex Calling:

    
    dial-peer voice 300 voip
     description UCM-Webex Calling trunk
     destination-pattern BAD.BAD
     session protocol sipv2
     session target dns:wxtocucm.io
     incoming uri via 300
     voice-class codec 100
     voice-class sip bind control source-interface GigabitEthernet 0/0/0
     voice-class sip bind media source-interface GigabitEthernet 0/0/0
     dtmf-relay rtp-nte
     no vad
    

    Here's an explanation of the fields for the configuration:

    
    dial-peer voice 300 voip
     description UCM-Webex Calling trunk

    Defines a VoIP dial-peer with a tag 300 and gives a meaningful description for ease of management and troubleshooting.

    destination-pattern BAD.BAD

    A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. Any valid destination pattern may be used in this case.

    session protocol sipv2

    Specifies that dial-peer 300 handles SIP call legs. For more information, see  session protocol (dial-peer).

    session target dns:wxtocucm.io

    Defines the session target of multiple Unified CM nodes through DNS SRV resolution. In this case, the locally defined SRV record wxtocucm.io is used to direct calls.

    incoming uri via 300

    Uses voice class URI 300 to direct all incoming traffic from Unified CM using source port 5065 to this dial-peer. For more information, see  incoming uri.

    voice-class codec 100

    Indicates codec filter list for calls to and from Unified CM. For more information, see  voice class codec.

    bind control source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see  bind.

    bind media source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for media sent to PSTN. For more information, see  bind.

    dtmf-relay rtp-nte

    Defines RTP-NTE (RFC2833) as the DTMF capability expected on the call leg. For more information, see  DTMF Relay (Voice over IP).

    no vad

    Disables voice activity detection. For more information, see  vad (dial peer).

  2. Dial-peer for calls between Unified CM and the PSTN:

    
    dial-peer voice 400 voip
     description UCM-PSTN trunk
     destination-pattern BAD.BAD
     session protocol sipv2
     session target dns:pstntocucm.io
     incoming uri via 400
     voice-class codec 100 
     voice-class sip bind control source-interface GigabitEthernet 0/0/0
     voice-class sip bind media source-interface GigabitEthernet 0/0/0
     dtmf-relay rtp-nte
     no vad
    

    Here's an explanation of the fields for the configuration:

    
    dial-peer voice 400 voip
     description UCM-PSTN trunk

    Defines a VoIP dial-peer with a tag of 400 and gives a meaningful description for ease of management and troubleshooting.

    destination-pattern BAD.BAD

    A dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. Any valid destination pattern may be used in this case.

    session protocol sipv2

    Specifies that dial-peer 400 handles SIP call legs. For more information, see  session protocol (dial-peer).

    session target dns:pstntocucm.io

    Defines the session target of multiple Unified CM nodes through DNS SRV resolution. In this case, the locally defined SRV record pstntocucm.io is used to direct calls.

    incoming uri via 400

    Uses voice class URI 400 to direct all incoming traffic from the specified Unified CM hosts using source port 5060 to this dial-peer. For more information, see  incoming uri.

    voice-class codec 100

    Indicates codec filter list for calls to and from Unified CM. For more information, see  voice class codec.

    bind control source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see  bind.

    bind media source-interface GigabitEthernet0/0/0

    Configures the source interface and associated IP address for media sent to PSTN. For more information, see  bind.

    dtmf-relay rtp-nte

    Defines RTP-NTE (RFC2833) as the DTMF capability expected on the call leg. For more information, see  DTMF Relay (Voice over IP).

    no vad

    Disables voice activity detection. For more information, see  vad (dial peer).

4

Add call routing using the following configurations:

  1. Create dial-peer groups to route calls between Unified CM and Webex Calling. Define DPG 100 with outbound dial-peer 100 towards Webex Calling. DPG 100 is applied to the associated incoming dial-peer from Unified CM. Similarly, define DPG 300 with outbound dial-peer 300 toward Unified CM. DPG 300 is applied to the incoming dial-peer from Webex.

    
    voice class dpg 100
     description Route calls to Webex Calling
     dial-peer 100
    voice class dpg 300
     description Route calls to Unified CM Webex Calling trunk
     dial-peer 300 
  2. Create a dial-peer groups to route calls between Unified CM and the PSTN. Define DPG 200 with outbound dial-peer 200 toward the PSTN. DPG 200 is applied to the associated incoming dial-peer from Unified CM. Similarly, define DPG 400 with outbound dial-peer 400 toward Unified CM. DPG 400 is applied to the incoming dial-peer from the PSTN.

    
    voice class dpg 200
     description Route calls to PSTN
     dial-peer 200
    voice class dpg 400
     description Route calls to Unified CM PSTN trunk
     dial-peer 400

    Here's an explanation of the fields for the configuration:

    dial-peer  100

    Associates an outbound dial-peer with a dial-peer group. For more information, see  voice-class dpg.

  3. Apply dial-peer groups to route calls from Webex to Unified CM and from Unified CM to Webex:

    
    dial-peer voice 100
     destination dpg 300
    dial-peer voice 300
     destination dpg 100

    Here's an explanation of the fields for the configuration:

    destination dpg 300

    Specifies which dial-peer group, and therefore dial-peer should be used for the outbound treatment for calls presented to this incoming dial-peer.

  4. Apply dial-peer groups to route calls from the PSTN to Unified CM and from Unified CM to the PSTN:

    
    dial-peer voice 200
     destination dpg 400
    dial-peer voice 400
     destination dpg 200 

    This concludes your Local Gateway configuration. Save the configuration and reload the platform if this is the first time CUBE features have been configured.

Diagnostic Signatures (DS) proactively detects commonly observed issues in the Cisco IOS XE-based Local Gateway and generates email, syslog, or terminal message notification of the event. You can also install the DS to automate diagnostics data collection and transfer collected data to the Cisco TAC case to accelerate resolution time.

Diagnostic Signatures (DS) are XML files that contain information about problem trigger events and actions to inform, troubleshoot, and remediate the issue. Use syslog messages, SNMP events and through periodic monitoring of specific show command outputs to define the problem detection logic. The action types include:

  • Collecting show command outputs

  • Generating a consolidated log file

  • Uploading the file to a user provided network location such as HTTPS, SCP, FTP server

TAC engineers author DS files and digitally sign it for integrity protection. Each DS file has the unique numerical ID assigned by the system. Diagnostic Signatures Lookup Tool (DSLT) is a single source to find applicable signatures for monitoring and troubleshooting various problems.

Before you begin:

  • Do not edit the DS file that you download from DSLT. The files that you modify fail installation due to the integrity check error.

  • A Simple Mail Transfer Protocol (SMTP) server you require for the Local Gateway to send out email notifications.

  • Ensure that the Local Gateway is running IOS XE 17.6.1 or higher if you wish to use the secure SMTP server for email notifications.

Prerequisites

Local Gateway running IOS XE 17.6.1 or higher

  1. Diagnostic Signatures is enabled by default.

  2. Configure the secure email server that you use to send proactive notification if the device is running IOS XE 17.6.1 or higher.

    
    configure terminal 
    call-home  
    mail-server <username>:<pwd>@<email server> priority 1 secure tls 
    end 

  3. Configure the environment variable ds_email with the email address of the administrator to you notify.

    
    configure terminal 
    call-home  
    diagnostic-signature 
    LocalGateway(cfg-call-home-diag-sign)environment ds_email <email address> 
    end 

Install diagnostic signatures for proactive monitoring

Monitoring high CPU utilization

This DS tracks 5-seconds CPU utilization using the SNMP OID 1.3.6.1.4.1.9.2.1.56. When the utilization reaches 75% or more, it disables all debugs and uninstalls all diagnostic signatures that you install in the Local Gateway. Use these steps below to install the signature.

  1. Ensure that you enabled SNMP using the command show snmp. If SNMP is not enabled, then configure the snmp-server manager command.

    
    show snmp 
    %SNMP agent not enabled  
    
    config t 
    snmp-server manager 
    end  
    
    show snmp 
    Chassis: ABCDEFGHIGK 
    149655 SNMP packets input 
        0 Bad SNMP version errors 
        1 Unknown community name 
        0 Illegal operation for community name supplied 
        0 Encoding errors 
        37763 Number of requested variables 
        2 Number of altered variables 
        34560 Get-request PDUs 
        138 Get-next PDUs 
        2 Set-request PDUs 
        0 Input queue packet drops (Maximum queue size 1000) 
    158277 SNMP packets output 
        0 Too big errors (Maximum packet size 1500) 
        20 No such name errors 
        0 Bad values errors 
        0 General errors 
        7998 Response PDUs 
        10280 Trap PDUs 
    Packets currently in SNMP process input queue: 0 
    SNMP global trap: enabled 
    
  2. Download DS 64224 using the following drop-down options in Diagnostic Signatures Lookup Tool:

    copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software

    Product

    CUBE Enterprise in Webex Calling solution

    Problem Scope

    Performance

    Problem Type

    High CPU Utilization with Email Notification

  3. Copy the DS XML file to the Local Gateway flash.

    copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash:

    The following example shows copying the file from an FTP server to the Local Gateway.

    copy ftp://user:pwd@192.0.2.12/DS_64224.xml bootflash: 
    Accessing ftp://*:*@ 192.0.2.12/DS_64224.xml...! 
    [OK - 3571/4096 bytes] 
    3571 bytes copied in 0.064 secs (55797 bytes/sec) 
    
  4. Install the DS XML file in the Local Gateway.

    
    call-home diagnostic-signature load DS_64224.xml 
    Load file DS_64224.xml success  
  5. Use the show call-home diagnostic-signature command to verify that the signature is successfully installed. The status column must have a “registered” value.

    
    show call-home diagnostic-signature  
    Current diagnostic-signature settings: 
     Diagnostic-signature: enabled 
     Profile: CiscoTAC-1 (status: ACTIVE) 
     Downloading  URL(s):  https://tools.cisco.com/its/service/oddce/services/DDCEService 
     Environment variable: 
               ds_email: username@gmail.com 

    Download DSes:

    DS ID

    DS Name

    Revision

    Status

    Last Update (GMT+00:00)

    64224

    DS_LGW_CPU_MON75

    0.0.10

    Registered

    2020-11-07 22:05:33


     

    When triggered, this signature uninstalls all running DSs including itself. If necessary, please reinstall DS 64224 to continue monitoring high CPU utilization on the Local Gateway.

Monitoring abnormal call disconnects

This DS uses SNMP polling every 10 minutes to detect abnormal call disconnect with SIP errors 403, 488 and 503.  If the error count increment is greater than or equal to 5 from the last poll, it generates a syslog and email notification. Please use the steps below to install the signature.

  1. Ensure that SNMP is enabled using the command show snmp. If SNMP is not enabled, configure the snmp-server manager command.

    show snmp 
    %SNMP agent not enabled  
    
    config t 
    snmp-server manager 
    end  
    
    show snmp 
    Chassis: ABCDEFGHIGK 
    149655 SNMP packets input 
        0 Bad SNMP version errors 
        1 Unknown community name 
        0 Illegal operation for community name supplied 
        0 Encoding errors 
        37763 Number of requested variables 
        2 Number of altered variables 
        34560 Get-request PDUs 
        138 Get-next PDUs 
        2 Set-request PDUs 
        0 Input queue packet drops (Maximum queue size 1000) 
    158277 SNMP packets output 
        0 Too big errors (Maximum packet size 1500) 
        20 No such name errors 
        0 Bad values errors 
        0 General errors 
        7998 Response PDUs 
        10280 Trap PDUs 
    Packets currently in SNMP process input queue: 0 
    SNMP global trap: enabled 
  2. Download DS 65221 using the following options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Performance

    Problem Type

    SIP abnormal call disconnect detection with Email and Syslog Notification.

  3. Copy the DS XML file to the Local Gateway.

    copy ftp://username:password@<server name or ip>/DS_65221.xml bootflash:
  4. Install the DS XML file in the Local Gateway.

    
    call-home diagnostic-signature load DS_65221.xml 
    Load file DS_65221.xml success 
  5. Use the command show call-home diagnostic-signature to verify that the signature is successfully installed. The status column should have a “registered” value.

Install diagnostic signatures to troubleshoot a problem

You can also use Diagnostic Signatures (DS) to resolve issues quickly. Cisco TAC engineers have authored several signatures that enable the necessary debugs that are required to troubleshoot a given problem, detect the problem occurrence, collect the right set of diagnostic data and transfer the data automatically to the Cisco TAC case. This eliminates the need to manually check for the problem occurrence and makes troubleshooting of intermittent and transient issues a lot easier.

You can use the Diagnostic Signatures Lookup Tool to find the applicable signatures and install them to selfsolve a given issue or you can install the signature that is recommended by the TAC engineer as part of the support engagement.

Here is an example of how to find and install a DS to detect the occurrence “%VOICE_IEC-3-GW: CCAPI: Internal Error (call spike threshold): IEC=1.1.181.1.29.0" syslog and automate diagnostic data collection using the following steps:

  1. Configure another DS environment variable ds_fsurl_prefix as the Cisco TAC file server path (cxd.cisco.com) to upload the diagnostics data. The username in the file path is the case number and the password is the file upload token which can be retrieved from Support Case Manager as shown in the following. The file upload token can be generated in the Attachments section of the Support Case Manager, as required.

    
    configure terminal 
    call-home  
    diagnostic-signature 
    LocalGateway(cfg-call-home-diag-sign)environment ds_fsurl_prefix "scp://<case number>:<file upload token>@cxd.cisco.com"  
    end 

    Example:

    
    call-home  
    diagnostic-signature 
    environment ds_fsurl_prefix " environment ds_fsurl_prefix "scp://612345678:abcdefghijklmnop@cxd.cisco.com"  
  2. Ensure that SNMP is enabled using the command show snmp. If SNMP not enabled, configure the snmp-server manager command.

    
    show snmp 
    %SNMP agent not enabled 
     
    config t 
    snmp-server manager 
    end 
  3. We recommend installing the High CPU monitoring DS 64224 as a proactive measure to disable all debugs and diagnostics signatures during the time of high CPU utilization. Download DS 64224 using the following options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Performance

    Problem Type

    High CPU Utilization with Email Notification.

  4. Download DS 65095 using the following options in Diagnostic Signatures Lookup Tool:

    Field Name

    Field Value

    Platform

    Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software

    Product

    CUBE Enterprise in Webex Calling Solution

    Problem Scope

    Syslogs

    Problem Type

    Syslog - %VOICE_IEC-3-GW: CCAPI: Internal Error (Call spike threshold): IEC=1.1.181.1.29.0

  5. Copy the DS XML files to the Local Gateway.

    
    copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash: 
    copy ftp://username:password@<server name or ip>/DS_65095.xml bootflash: 
  6. Install the high CPU monitoring DS 64224 and then DS 65095 XML file in the Local Gateway.

    
    call-home diagnostic-signature load DS_64224.xml 
    Load file DS_64224.xml success 
    call-home diagnostic-signature load DS_65095.xml 
    Load file DS_65095.xml success 
    
  7. Verify that the signature is successfully installed using show call-home diagnostic-signature. The status column should have a “registered” value.

    
    show call-home diagnostic-signature  
    Current diagnostic-signature settings: 
     Diagnostic-signature: enabled 
     Profile: CiscoTAC-1 (status: ACTIVE) 
     Downloading  URL(s):  https://tools.cisco.com/its/service/oddce/services/DDCEService 
     Environment variable: 
               ds_email: username@gmail.com 
               ds_fsurl_prefix: scp://612345678:abcdefghijklmnop@cxd.cisco.com 

    Downloaded DSes:

    DS ID

    DS Name

    Revision

    Status

    Last Update (GMT+00:00)

    64224

    00:07:45

    DS_LGW_CPU_MON75

    0.0.10

    Registered

    2020-11-08:00:07:45

    65095

    00:12:53

    DS_LGW_IEC_Call_spike_threshold

    0.0.12

    Registered

    2020-11-08:00:12:53

Verify diagnostic signatures execution

In the following command, the “Status” column of the command show call-home diagnostic-signature changes to “running” while the Local Gateway executes the action defined within the signature. The output of show call-home diagnostic-signature statistics is the best way to verify whether a diagnostic signature detects an event of interest and executed the action. The “Triggered/Max/Deinstall” column indicates the number of times the given signature has triggered an event, the maximum number of times it is defined to detect an event and whether the signature deinstalls itself after detecting the maximum number of triggered events.

show call-home diagnostic-signature  
Current diagnostic-signature settings: 
 Diagnostic-signature: enabled 
 Profile: CiscoTAC-1 (status: ACTIVE) 
 Downloading  URL(s):  https://tools.cisco.com/its/service/oddce/services/DDCEService 
 Environment variable: 
           ds_email: carunach@cisco.com 
           ds_fsurl_prefix: scp://612345678:abcdefghijklmnop@cxd.cisco.com 

Downloaded DSes:

DS ID

DS Name

Revision

Status

Last Update (GMT+00:00)

64224

DS_LGW_CPU_MON75

0.0.10

Registered

2020-11-08 00:07:45

65095

DS_LGW_IEC_Call_spike_threshold

0.0.12

Running

2020-11-08 00:12:53

show call-home diagnostic-signature statistics

DS ID

DS Name

Triggered/Max/Deinstall

Average Run Time (seconds)

Max Run Time (seconds)

64224

DS_LGW_CPU_MON75

0/0/N

0.000

0.000

65095

DS_LGW_IEC_Call_spike_threshold

1/20/Y

23.053

23.053

The notification email that is sent during Diagnostic Signature execution contains key information such as issue type, device details, software version, running configuration and show command outputs that are relevant to troubleshoot the given problem.

Uninstall diagnostic signatures

Use the diagnostic signatures for troubleshooting purposes are typically defined to uninstall after detection of some problem occurrences. If you wish to uninstall a signature manually, retrieve the DS ID from the output of show call-home diagnostic-signature and run the following command:

call-home diagnostic-signature deinstall <DS ID> 

Example:

call-home diagnostic-signature deinstall 64224 

 

New signatures are added to the Diagnostics Signatures Lookup Tool periodically, based on issues that are observed in deployments. TAC currently doesn’t support requests to create new custom signatures.