Configurar o gateway local no Cisco IOS XE do Webex Calling
Depois de configurar Webex Calling para sua organização, você pode configurar um tronco para conectar o Gateway local ao Webex Calling. O transporte SIP TLS proteja o tronco entre o Gateway local e a nuvem Webex. A mídia entre o Gateway Local e o Webex Calling usa SRTP.
Visão geral
Webex Calling currently supports two versions of Local Gateway:
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Gateway local
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Local Gateway for Webex for Government
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Before you begin, understand the premises-based Public Switched Telephone Network (PSTN) and Local Gateway (LGW) requirements for Webex Calling. Consulte Arquitetura de preferência da Cisco para Webex Calling para obter mais informações.
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Este artigo pressu que uma plataforma de Gateway local dedicada está no lugar sem a configuração de voz existente. 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.
For information on the supported third-party SBCs, refer to the respective product reference documentation.
Há duas opções para configurar o Gateway local para seu Webex Calling de acesso:
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Tronco baseado em registro
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Tronco baseado em certificado
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. Execute os seguintes passos no próprio Gateway local, usando a Interface de Linha de Comando (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.
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Select CUBE as your Local Gateway. Webex for Government doesn’t currently support any third-party Session Border Controllers (SBCs). To review the latest list, see Get started with Local Gateway.
- Install Cisco IOS XE Dublin 17.12.1a or later versions for all Webex for Government Local Gateways.
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To review the list of root Certificate Authorities (CAs) that Webex for Government support, see Root certificate authorities for Webex for Government.
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For details on the external port ranges for Local Gateway in Webex for Government, see Network requirements for Webex for Government (FedRAMP).
Local Gateway for Webex for Government doesn’t support the following:
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STUN/ICE-Lite for media path optimization
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Fax (T.38)
To configure Local Gateway for your Webex Calling trunk in Webex for Government, use the following option:
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Tronco baseado em certificado
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.
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:
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voice class tenants: Used to create trunk specific configurations.
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voice class uri: Used to classify SIP messages for the selection of an inbound dial-peer.
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inbound dial-peer: Provides treatment for inbound SIP messages and determines the outbound route with a dial-peer group.
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dial-peer group: Defines the outbound dial-peers used for onward call routing.
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outbound dial-peer: Provides treatment for outbound SIP messages and routes them to the required target.
While IP and SIP have become the default protocols for PSTN trunks, TDM (Time Division Multiplexing) ISDN circuits are still widely used and are supported with Webex Calling trunks. To enable media optimization of IP paths for Local Gateways with TDM-IP call flows, it is currently necessary to use a two-leg call routing process. This approach modifies the call routing configuration shown above, by introducing a set of internal loop-back dial-peers between Webex Calling and PSTN trunks as illustrated in the image below.
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:
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Etapa 1: Configure router baseline connectivity and security
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Etapa 2: Configure Webex Calling Trunk
Depending on your required architecture, follow either:
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Passo 2: Configure Local Gateway with SIP PSTN trunk
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Passo 4: Configure Local Gateway with existing Unified CM environment
Ou:
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Passo 2: 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.
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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.
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ISR4000 series routers must be configured with both Unified Communications and Security technology licenses.
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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.
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Build a baseline configuration for your platform that follows your business policies. In particular, configure the following and verify the working:
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Ntp
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Acls
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User authentication and remote access
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DNS
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Roteamento IP
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IP addresses
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The network toward Webex Calling must use an IPv4 address.
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Upload the Cisco root CA bundle to the Local Gateway.
Configuração
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Ensure that you assign valid and routable IP addresses to any Layer 3 interfaces, for example:
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Protect registration and STUN credentials on the router using symmetric encryption. Configure the primary encryption key and encryption type as follows:
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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. |
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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.
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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 |
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Create a registration based PSTN trunk for an existing location in the Control Hub. Make a note of the trunk information that is provided once the trunk has been created. The details highlighted in the illustration are used in the configuration steps in this guide. For more information, see Configure trunks, route groups, and dial plans for Webex Calling. |
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Enter the following commands to configure CUBE as a Webex Calling Local Gateway: Aqui está uma explicação dos campos para a configuração:
Enables Cisco Unified Border Element (CUBE) features on the platform. media statisticsPermite o monitoramento de mídia no Gateway local. media bulk-statsPermite que o avião de controle sondar o avião de dados para as estatísticas de chamada em massa. For more information on these commands, see Media. permitir conexões sip para sipEnable 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). Enables STUN (Session Traversal of UDP through NAT) globally.
For more information, see stun flowdata agent-id and stun flowdata shared-secret. asymmetric payload fullConfigures SIP asymmetric payload support for both DTMF and dynamic codec payloads. For more information on this command, see asymmetric payload. oferta antecipada forçadaForces 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. |
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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. Aqui está uma explicação dos campos para a configuração: voice class codec 100Used 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. |
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Configure voice class stun-usage 100 to enable ICE on the Webex Calling trunk. Aqui está uma explicação dos campos para a configuração: stun usage ice liteUsed 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 |
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Configure the media encryption policy for Webex traffic. Aqui está uma explicação dos campos para a configuração: voice class srtp-crypto 100Specifies 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. |
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Configure a pattern to identify calls to a Local Gateway trunk based on its destination trunk parameter: Aqui está uma explicação dos campos para a configuração: voice class uri 100 sipDefines 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 the Control Hub when the trunk was created. For more information, see voice class uri. |
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Configure sip profile 100, which will be used to modify SIP messages before they are sent to Webex Calling.
Aqui está uma explicação dos campos para a configuração:
United States or Canadian PSTN provider can offer the Caller ID verification for Spam and fraud calls, with the additional configuration mentioned in the Spam or fraud call indication in Webex Calling article. |
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Configure Webex Calling trunk: |
After you define the 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 recognized, 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:
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The username, password, and realm parameters from the credentials configuration is used in the response.
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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.
If you are using a TDM / ISDN PSTN trunk, skip to next section Configure Local Gateway with TDM PSTN trunk.
To configure TDM interfaces for PSTN call legs on the Cisco TDM-SIP Gateways, see Configuring ISDN PRI.
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Configure the following voice class uri to identify inbound calls from the PSTN trunk: Aqui está uma explicação dos campos para a configuração: voice class uri 200 sipDefines 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. |
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Configure the following IP PSTN dial-peer: Aqui está uma explicação dos campos para a configuração: Define um novo VoIP de discagem com uma tag de 200 e fornece uma descrição significativa para facilitar o gerenciamento e a solução de problemas. For more information, see dial-peer voice. padrão de destino BAD. RuimA dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface). sessão protocolo sipv2Especifica que o grupo de discagem 200 lida com os colegas de chamada SIP. For more information, see session protocol (dial peer). sessão de destino ipv4:192.168.80.13Indica o endereço IPv4 de destino do destino para enviar o trecho de chamada. O destino da sessão aqui é o endereço de IP do ITSP. For more information, see session target (VoIP dial peer). incoming uri via 200Define um critério de responsabilidade para o header VIA com o endereço de IP PSTN IP de . 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 GigabitEthernet0/0/0Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see bind. bind media source-interface GigabitEthernet0/0/0Configures the source interface and associated IP address for media sent to PSTN. For more information, see bind. codec de classe de voz 100Configures the dial-peer to use the common codec filter list 100. For more information, see voice-class codec. dtmf-relay rtp-nteDefine o RTP-NTE (RFC2833) como o recurso DTMF esperado no retorno de chamada. For more information, see DTMF Relay (Voice over IP). sem vadDesativa a detecção de atividade de voz. For more information, see vad (dial peer). |
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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. |
Having built a trunk towards Webex Calling, use the following configuration to create a TDM trunk for your PSTN service with loop-back call routing to allow media optimization on the Webex call leg.
If you do not require IP media optimization, follow the configuration steps for a SIP PSTN trunk. Use a voice port and POTS dial-peer (as shown in Steps 2 and 3) instead of the PSTN VoIP dial-peer.
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The loop-back dial-peer configuration uses dial-peer groups and call routing tags to ensure that calls pass correctly between Webex and the PSTN, without creating call routing loops. Configure the following translation rules that will be used to add and remove the call routing tags: Aqui está uma explicação dos campos para a configuração: voice translation-ruleUses regular expressions defined in rules to add or remove call routing tags. Over-decadic digits (‘A’) are used to add clarity for troubleshooting. In this configuration, the tag added by translation-profile 100 is used to guide calls from Webex Calling towards the PSTN via the loopback dial-peers. Similarly, the tag added by translation-profile 200 is used to guide calls from the PSTN towards Webex Calling. Translation-profiles 11 and 12 remove these tags before delivering calls to the Webex and PSTN trunks respectively. This example assumes that called numbers from Webex Calling are presented in +E.164 format. Rule 100 removes the leading + to maintain a valid called number. Rule 12 then adds a national or international routing digit(s) when removing the tag. Use digits that suit your local ISDN national dial plan. If Webex Calling presents numbers in national format, adjust rules 100 and 12 to simply add and remove the routing tag respectively. For more information, see voice translation-profile and voice translation-rule. |
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Configure TDM voice interface ports as required by the trunk type and protocol used. For more information, see Configuring ISDN PRI. For example, the basic configuration of a Primary Rate ISDN interface installed in NIM slot 2 of a device might include the following: |
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Configure the following TDM PSTN dial-peer: Aqui está uma explicação dos campos para a configuração: Define um VoIP discagem com uma tag de 200 e fornece uma descrição significativa para facilitar o gerenciamento e a solução de problemas. For more information, see dial-peer voice. padrão de destino BAD. RuimA dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface). translation-profile incoming 200Assigns the translation profile that will add a call routing tag to the incoming called number. direct-inward-dialRoutes the call without providing a secondary dial-tone. For more information, see direct-inward-dial. port 0/2/0:15The physical voice port associated with this dial-peer. |
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To enable media optimization of IP paths for Local Gateways with TDM-IP call flows, you can modify the call routing by introducing a set of internal loop-back dial-peers between Webex Calling and PSTN trunks. Configure the following loop-back dial-peers. In this case, all incoming calls will be routed initially to dial-peer 10 and from there to either dial-peer 11 or 12 based on the applied routing tag. After removal of the routing tag, calls will be routed to the outbound trunk using dial-peer groups. Aqui está uma explicação dos campos para a configuração: Defines a VoIP dial-peer and gives a meaningful description for ease of management and troubleshooting. For more information, see dial-peer voice. translation-profile incoming 11Applies the translation profile defined earlier to remove the call routing tag before passing to the outbound trunk. padrão de destino BAD. RuimA dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface). sessão protocolo sipv2Specifies that this dial-peer handles SIP call legs. For more information, see session protocol (dial peer). session target 192.168.80.14Specifies the local router interface address as the call target to loop-back. For more information, see session target (voip dial peer). bind control source-interface GigabitEthernet0/0/0Configures the source interface and associated IP address for messages sent through the loop-back. For more information, see bind. bind media source-interface GigabitEthernet0/0/0Configures the source interface and associated IP address for media sent through the loop-back. For more information, see bind. dtmf-relay rtp-nteDefine o RTP-NTE (RFC2833) como o recurso DTMF esperado no retorno de chamada. For more information, see DTMF Relay (Voice over IP). codec g711alaw Forces all PSTN calls to use G.711. Select a-law or u-law to match the companding method used by your ISDN service. sem vadDesativa a detecção de atividade de voz. For more information, see vad (dial peer). |
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Add the following call routing configuration: This concludes your Local Gateway configuration. Save the configuration and reload the platform if this is the first time CUBE features are configured.
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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.
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Configure as seguintes URIs de classe de voz: |
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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. Aqui está uma explicação dos campos para a configuração: 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. Por exemplo: 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. |
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Configure the following dial-peers: |
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Add call routing using the following configurations: |
As Assinaturas de Diagnóstico (DS) detectam proativamente problemas observados no Gateway local baseado no IOS XE e gera notificação de e-mail, syslog ou mensagem terminal do evento. You can also install the DS to automate diagnostics data collection and transfer-collected data to the Cisco TAC case to accelerate resolution time.
As assinaturas de diagnóstico (DS) são arquivos XML que contêm informações sobre eventos de acionador de problemas e ações a serem tomadas para informar, solucionar o problema e solucionar o problema. You can define the problem detection logic using syslog messages, SNMP events and through periodic monitoring of specific show command outputs.
Os tipos de ação incluem a coleta de saídas mostrar comandos:
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Gerando um arquivo de registro consolidado
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Uploading the file to a user-provided network location such as HTTPS, SCP, FTP server.
Os engenheiros do TAC autorizam os arquivos DS e os assinam digitalmente para uma proteção de integridade. Cada arquivo DS possui uma ID numérica exclusiva atribuída pelo sistema. Diagnostic Signatures Lookup Tool (DSLT) is a single source to find applicable signatures for monitoring and troubleshooting various problems.
Antes de você começar:
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Não edite o arquivo DS que você baixa do DSLT. Os arquivos que você modifica falham na instalação devido ao erro de verificação de integridade.
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Um servidor de Protocolo de Transferência de E-mail Simples (SMTP) que você precisa para o Gateway Local enviar notificações por e-mail.
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Certifique-se de que o Gateway Local está executando o IOS XE 17.6.1 ou superior se você desejar usar o servidor SMTP seguro para notificações por e-mail.
Pré-requisitos
Local Gateway running IOS XE 17.6.1a or higher
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As Assinaturas de diagnóstico estão ativadas por padrão.
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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
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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"
Um Gateway local em execução no software Cisco IOS XE não é um cliente do Gmail comum baseado na web que suporta o OAuth, então é preciso configurar uma configuração específica de conta de Gmail e fornecer permissão específica para que o e-mail do dispositivo seja processado corretamente:
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Go to Less secure app access setting.
and turn on the -
Responda "Sim, foi eu" quando você recebeu um e-mail do Gmail indicando "O Google impediu alguém de entrar na sua conta usando um aplicativo que não é o Google".
Instalar assinaturas de diagnóstico para monitoramento proativo
Monitoramento de alta utilização da CPU
This DS tracks CPU utilization for five seconds using the SNMP OID 1.3.6.1.4.1.9.2.1.56. Quando a utilização atingir 75% ou mais, ela desativa todas as depurações e desinstala todas as assinaturas de diagnóstico que estão instaladas no Gateway local. Use as etapas abaixo para instalar a assinatura.
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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 Intercepta global SNMP: habilitada
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Baixe o DS 64224 usando as seguintes opções suspensas na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series or Cisco CSR 1000V Series
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Desempenho
Tipo de problema
Alta utilização da CPU com notificação por e-mail.
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Copie o arquivo DS XML para o flash do Gateway local.
LocalGateway# copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash:
O exemplo a seguir mostra a cópia do arquivo de um servidor FTP para o Gateway Local.
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)
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Instale o arquivo DS XML no Gateway local.
call-home diagnostic-signature load DS_64224.xml Load file DS_64224.xml success
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Use o comando mostrar assinatura de diagnóstico de chamada de casa para verificar se a assinatura foi instalada com êxito. A coluna de status deve ter um valor "registrado".
show call-home diagnostic-signature Current diagnostic-signature settings: Assinatura de diagnóstico: Perfil habilitado: Cisco SENA-1 (status: ATIVO) URL(s) de download: https://tools.cisco.com/its/service/oddce/services/DDCEService Environment variable: ds_email: username@gmail.com
Baixe o DSes:
ID de DS
Nome DS
Revisão
Status
Última atualização (GMT+00:00)
64224
DS_LGW_CPU_MON75
0.0.10
Registrado
2020-11-07 22:05:33
Quando acionada, esta assinatura desinstala todos os DSs em execução, incluindo ela própria. If necessary, reinstall DS 64224 to continue monitoring high CPU utilization on the Local Gateway.
Monitoramento de registro de tronco SIP
Este DS verifica a irretribuição de um gateway local Tronco SIP com Webex Calling nuvem a cada 60 segundos. 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:
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Baixe o DS 64117 usando as seguintes opções suspensas na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series ou Cisco CSR 1000V Series
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
SIP-SIP
Tipo de problema
Tronco SIP não-aviso com a notificação por e-mail.
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Copie o arquivo DS XML para o Gateway local.
copy ftp://username:password@<server name or ip>/DS_64117.xml bootflash:
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Instale o arquivo DS XML no Gateway local.
call-home diagnostic-signature load DS_64117.xml Load file DS_64117.xml success LocalGateway#
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Use o comando mostrar assinatura de diagnóstico de chamada de casa para verificar se a assinatura foi instalada com êxito. A coluna status deve ter um valor "registrado".
Monitorando desconectações anormals de chamada
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.
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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 Intercepta global SNMP: habilitada
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Baixe o DS 65221 usando as seguintes opções na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series ou Cisco CSR 1000V Series
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Desempenho
Tipo de problema
Detecção anormal de desconexão de chamada SIP com e-mail e notificação Syslog.
-
Copie o arquivo DS XML para o Gateway local.
copy ftp://username:password@<server name or ip>/DS_65221.xml bootflash:
-
Instale o arquivo DS XML no Gateway local.
call-home diagnostic-signature load DS_65221.xml Load file DS_65221.xml success
-
Use o comando mostrar assinatura de diagnóstico de chamada de casa para verificar se a assinatura foi instalada com êxito. A coluna status deve ter um valor "registrado".
Instalar assinaturas de diagnóstico para solucionar um problema
Use assinaturas de diagnóstico (DS) para resolver os problemas rapidamente. Os engenheiros do TAC da Cisco criaram várias assinaturas que permitem as depurações necessárias que são necessárias para solucionar um determinado problema, detectar a ocorrência do problema, coletar o conjunto certo de dados de diagnóstico e transferir os dados automaticamente para o caso do CISCO TAC. Diagnostic Signatures (DS) eliminate the need to manually check for the problem occurrence and makes troubleshooting of intermittent and transient issues a lot easier.
Você pode usar a Ferramenta de pesquisa de assinaturas de diagnóstico para encontrar as assinaturas aplicáveis e instalá-las para auto-resolver um determinado problema ou você pode instalar a assinatura que é recomendada pelo engenheiro de TAC como parte do envolvimento do suporte.
Aqui está um exemplo de como encontrar e instalar um DS para detectar a ocorrência "%VOICE_IEC-3-GW: CCAPI: Erro interno (limite de pico de chamadas): IEC=1.1.181.1.29.0" syslog e automatize a coleta de dados de diagnóstico usando as seguintes etapas:
-
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
Exemplo:
call-home diagnostic-signature environment ds_fsurl_prefix " environment ds_fsurl_prefix "scp://612345678:abcdefghijklmnop@cxd.cisco.com"
-
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
-
Certifique-se de instalar o monitoramento de alta CPU DS 64224 como uma medida proativa para desativar todas as assinaturas de depuração e diagnósticos durante o momento de alta utilização da CPU. Baixe o DS 64224 usando as seguintes opções na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series ou Cisco CSR 1000V Series
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Desempenho
Tipo de problema
Alta utilização da CPU com notificação por e-mail.
-
Baixe o DS 65095 usando as seguintes opções na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series ou Cisco CSR 1000V Series
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Syslogs
Tipo de problema
Syslog - %VOICE_IEC-3-GW: CCAPI: Erro interno (limite de pico de chamadas): IEC=1.1.181.1.29.0
-
Copie os arquivos DS XML para o Gateway local.
copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash: copy ftp://username:password@<server name or ip>/DS_65095.xml bootflash:
-
Instale o monitoramento de Alta CPU DS 64224 e, em seguida, o arquivo XML do DS 65095 no Gateway local.
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
-
Verify that the signature is successfully installed using the show call-home diagnostic-signature command. A coluna status deve ter um valor "registrado".
show call-home diagnostic-signature Current diagnostic-signature settings: Assinatura de diagnóstico: Perfil habilitado: Cisco SENA-1 (status: ATIVO) URL(s) de download: 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
DSes baixados:
ID de DS
Nome DS
Revisão
Status
Última atualização (GMT+00:00)
64224
00:07:45
DS_LGW_CPU_MON75
0.0.10
Registrado
2020-11-08
65095
00:12:53
DS_LGW_IEC_Call_spike_threshold
0.0.12
Registrado
2020-11-08
Verificar a execução de assinaturas de diagnóstico
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. O resultado de mostrar as estatísticas de diagnóstico de chamada de casa é a melhor maneira de verificar se uma assinatura de diagnóstico detecta um evento de interesse e executa a ação. A coluna "Acionado/Máximo/Desinstalação" indica o número de vezes que a assinatura dada acionou um evento, o número máximo de vezes que é definido para detectar um evento e se a assinatura desinstala-se depois de detectar o número máximo de eventos acionados.
show call-home diagnostic-signature Current diagnostic-signature settings: Assinatura de diagnóstico: Perfil
habilitado: Cisco SENA-1 (status: ATIVO)
URL(s) de download: 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
DSes baixados:
ID de DS |
Nome DS |
Revisão |
Status |
Última atualização (GMT+00:00) |
---|---|---|---|---|
64224 |
DS_LGW_CPU_MON75 |
0.0.10 |
Registrado |
2020-11-08 00:07:45 |
65095 |
DS_LGW_IEC_Call_spike_threshold |
0.0.12 |
Em execução |
2020-11-08 00:12:53 |
mostrar estatísticas de assinatura do diagnóstico de chamada de casa
ID de DS |
Nome DS |
Triggered/Max/Deinstall |
Tempo médio de execução (segundos) |
Tempo máximo de execução (segundos) |
---|---|---|---|---|
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 |
O e-mail de notificação que é enviado durante a execução da assinatura de diagnóstico contém informações importantes como tipo de problema, detalhes do dispositivo, versão do software, configuração em execução e mostra saídas de comando que são relevantes para solucionar o problema dado.
Desinstalar assinaturas de diagnóstico
Use assinaturas de Diagnóstico para fins de solução de problemas são tipicamente definidas para desinstalar após a detecção de algumas ocorrências de problemas. 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>
Exemplo:
call-home diagnostic-signature deinstall 64224
Novas assinaturas são adicionadas à Ferramenta de pesquisa de assinaturas de diagnóstico periodicamente, com base em problemas que são geralmente observados nas implantações. Atualmente, o TAC não oferece suporte a solicitações de criação de novas assinaturas personalizadas.
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.
While IP and SIP have become the default protocols for PSTN trunks, TDM (Time Division Multiplexing) ISDN circuits are still widely used and are supported with Webex Calling trunks. To enable media optimization of IP paths for Local Gateways with TDM-IP call flows, it is currently necessary to use a two-leg call routing process. This approach modifies the call routing configuration shown above, by introducing a set of internal loop-back dial-peers between Webex Calling and PSTN trunks as illustrated in the image below.
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:
-
Etapa 1: Configure router baseline connectivity and security
-
Etapa 2: Configure Webex Calling Trunk
Depending on your required architecture, follow either:
-
Passo 2: Configure Local Gateway with SIP PSTN trunk
-
Passo 4: Configure Local Gateway with existing Unified CM environment
Ou:
-
Passo 2: 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
-
Roteamento IP
-
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. Por exemplo:
-
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. Os registros que o SRV de usuário resolve para (CNAME, Um Registro ou Endereço IP) são opcionais em 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.
-
-
-
Certifique-se de que os certificados sejam assinados para o uso do cliente e do servidor.
-
Upload the Cisco root CA bundle to the Local Gateway.
Configuração
1 |
Ensure that you assign valid and routable IP addresses to any Layer 3 interfaces, for example:
|
2 |
Protect STUN credentials on the router using symmetric encryption. Configure the primary encryption key and encryption type as follows: |
3 |
Create an encryption trustpoint with a certificate signed by your preferred Certificate Authority (CA). |
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:
|
5 |
Import a signed host certificate using the following exec or configuration command:
|
6 |
Enable TLS1.2 exclusivity and specify the default trustpoint using the following configuration commands:
|
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 |
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: Aqui está uma explicação dos campos para a configuração:
Enables Cisco Unified Border Element (CUBE) features on the platform. permitir conexões sip para sipEnable 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). Enables STUN (Session Traversal of UDP through NAT) globally. These global stun commands are only required when deploying your Local Gateway behind NAT.
For more information, see stun flowdata agent-id and stun flowdata shared-secret. asymmetric payload fullConfigures SIP asymmetric payload support for both DTMF and dynamic codec payloads. For more information on this command, see asymmetric payload. oferta antecipada forçadaForces 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 inboundEnables 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. Aqui está uma explicação dos campos para a configuração: voice class codec 100Used 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. (This step is not applicable for Webex for Government) Aqui está uma explicação dos campos para a configuração: stun usage ice liteUsed 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) Aqui está uma explicação dos campos para a configuração: voice class srtp-crypto 100Specifies 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). Aqui está uma explicação dos campos para a configuração: voice class srtp-crypto 100Specifies 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: Aqui está uma explicação dos campos para a configuração: voice class uri 100 sipDefines 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: Aqui está uma explicação dos campos para a configuração: rules 10 and 20To 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
Aqui está uma explicação dos campos para a configuração: rules 10 and 20To 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 81Convert 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 Aqui está uma explicação dos campos para a configuração: rules 10 to 80Convert 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. United States or Canadian PSTN provider can offer the Caller ID verification for Spam and fraud calls, with the additional configuration mentioned in the Spam or fraud call indication in Webex Calling article. |
10 |
Configure a SIP Options keepalive with header modification profile. Aqui está uma explicação dos campos para a configuração: voice class sip-options-keepalive 100Configures 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: |
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.
If you are using a TDM / ISDN PSTN trunk, skip to next section Configure Local Gateway with TDM PSTN trunk.
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: Aqui está uma explicação dos campos para a configuração: voice class uri 200 sipDefines 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: Aqui está uma explicação dos campos para a configuração: Define um novo VoIP de discagem com uma tag de 200 e fornece uma descrição significativa para facilitar o gerenciamento e a solução de problemas. For more information, see dial-peer voice. padrão de destino BAD. RuimA dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface). sessão protocolo sipv2Especifica que o grupo de discagem 200 lida com os colegas de chamada SIP. For more information, see session protocol (dial peer). sessão de destino ipv4:192.168.80.13Indica o endereço IPv4 de destino do destino para enviar o trecho de chamada. O destino da sessão aqui é o endereço de IP do ITSP. For more information, see session target (VoIP dial peer). incoming uri via 200Define um critério de responsabilidade para o header VIA com o endereço de IP PSTN IP de . 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 GigabitEthernet0/0/0Configures the source interface and associated IP address for messages sent to the PSTN. For more information, see bind. bind media source-interface GigabitEthernet0/0/0Configures the source interface and associated IP address for media sent to PSTN. For more information, see bind. codec de classe de voz 100Configures the dial-peer to use the common codec filter list 100. For more information, see voice-class codec. dtmf-relay rtp-nteDefine o RTP-NTE (RFC2833) como o recurso DTMF esperado no retorno de chamada. For more information, see DTMF Relay (Voice over IP). sem vadDesativa a detecção de atividade de voz. 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. |
Having built a trunk towards Webex Calling, use the following configuration to create a TDM trunk for your PSTN service with loop-back call routing to allow media optimization on the Webex call leg.
If you do not require IP media optimization, follow the configuration steps for a SIP PSTN trunk. Use a voice port and POTS dial-peer (as shown in Steps 2 and 3) instead of the PSTN VoIP dial-peer.
1 |
The loop-back dial-peer configuration uses dial-peer groups and call routing tags to ensure that calls pass correctly between Webex and the PSTN, without creating call routing loops. Configure the following translation rules that will be used to add and remove the call routing tags: Aqui está uma explicação dos campos para a configuração: voice translation-ruleUses regular expressions defined in rules to add or remove call routing tags. Over-decadic digits (‘A’) are used to add clarity for troubleshooting. In this configuration, the tag added by translation-profile 100 is used to guide calls from Webex Calling towards the PSTN via the loopback dial-peers. Similarly, the tag added by translation-profile 200 is used to guide calls from the PSTN towards Webex Calling. Translation-profiles 11 and 12 remove these tags before delivering calls to the Webex and PSTN trunks respectively. This example assumes that called numbers from Webex Calling are presented in +E.164 format. Rule 100 removes the leading + to maintain a valid called number. Rule 12 then adds a national or international routing digit(s) when removing the tag. Use digits that suit your local ISDN national dial plan. If Webex Calling presents numbers in national format, adjust rules 100 and 12 to simply add and remove the routing tag respectively. For more information, see voice translation-profile and voice translation-rule. |
2 |
Configure TDM voice interface ports as required by the trunk type and protocol used. For more information, see Configuring ISDN PRI. For example, the basic configuration of a Primary Rate ISDN interface installed in NIM slot 2 of a device might include the following: |
3 |
Configure the following TDM PSTN dial-peer: Aqui está uma explicação dos campos para a configuração: Define um VoIP discagem com uma tag de 200 e fornece uma descrição significativa para facilitar o gerenciamento e a solução de problemas. For more information, see dial-peer voice. padrão de destino BAD. RuimA dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface). translation-profile incoming 200Assigns the translation profile that will add a call routing tag to the incoming called number. direct-inward-dialRoutes the call without providing a secondary dial-tone. For more information, see direct-inward-dial. port 0/2/0:15The physical voice port associated with this dial-peer. |
4 |
To enable media optimization of IP paths for Local Gateways with TDM-IP call flows, you can modify the call routing by introducing a set of internal loop-back dial-peers between Webex Calling and PSTN trunks. Configure the following loop-back dial-peers. In this case, all incoming calls will be routed initially to dial-peer 10 and from there to either dial-peer 11 or 12 based on the applied routing tag. After removal of the routing tag, calls will be routed to the outbound trunk using dial-peer groups. Aqui está uma explicação dos campos para a configuração: Defines a VoIP dial-peer and gives a meaningful description for ease of management and troubleshooting. For more information, see dial-peer voice. translation-profile incoming 11Applies the translation profile defined earlier to remove the call routing tag before passing to the outbound trunk. padrão de destino BAD. RuimA dummy destination pattern is required when routing outbound calls using an inbound dial-peer group. For more information, see destination-pattern (interface). sessão protocolo sipv2Specifies that this dial-peer handles SIP call legs. For more information, see session protocol (dial peer). session target 192.168.80.14Specifies the local router interface address as the call target to loop-back. For more information, see session target (voip dial peer). bind control source-interface GigabitEthernet0/0/0Configures the source interface and associated IP address for messages sent through the loop-back. For more information, see bind. bind media source-interface GigabitEthernet0/0/0Configures the source interface and associated IP address for media sent through the loop-back. For more information, see bind. dtmf-relay rtp-nteDefine o RTP-NTE (RFC2833) como o recurso DTMF esperado no retorno de chamada. For more information, see DTMF Relay (Voice over IP). codec g711alaw Forces all PSTN calls to use G.711. Select a-law or u-law to match the companding method used by your ISDN service. sem vadDesativa a detecção de atividade de voz. For more information, see vad (dial peer). |
5 |
Add the following call routing configuration: 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 as seguintes URIs de classe de voz: |
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. Aqui está uma explicação dos campos para a configuração: 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. Por exemplo: 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: |
4 |
Add call routing using the following configurations: |
As Assinaturas de Diagnóstico (DS) detectam proativamente problemas observados no Gateway local baseado no Cisco IOS XE e gera notificações por e-mail, syslog ou mensagem terminal do evento. Você também pode instalar o DS para automatizar a coleta de dados de diagnóstico e transferir os dados coletados para o caso do TAC da Cisco a fim de acelerar o tempo de resolução.
Assinaturas de Diagnóstico (DS) são arquivos XML que contêm informações sobre eventos de acionador de problemas e ações para informar, solucionar problemas e resolver o problema. Use mensagens syslog, eventos SNMP e através do monitoramento periódico de mostrar saídas de comando específicas para definir a lógica de detecção de problemas. Os tipos de ações incluem:
-
Coletando saídas de comando show
-
Gerando um arquivo de registro consolidado
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Carregando o arquivo para um usuário fornecido localização de rede como HTTPS, SCP, servidor FTP
Os arquivos DS dos engenheiros do TAC são assinados digitalmente para uma proteção de integridade. Cada arquivo DS tem a ID numérica exclusiva atribuída pelo sistema. Diagnostic Signatures Lookup Tool (DSLT) is a single source to find applicable signatures for monitoring and troubleshooting various problems.
Antes de você começar:
-
Não edite o arquivo DS que você baixa do DSLT. Os arquivos que você modifica falham na instalação devido ao erro de verificação de integridade.
-
Um servidor de Protocolo de Transferência de E-mail Simples (SMTP) que você precisa para o Gateway Local enviar notificações por e-mail.
-
Certifique-se de que o Gateway Local está executando o IOS XE 17.6.1 ou superior se você desejar usar o servidor SMTP seguro para notificações por e-mail.
Pré-requisitos
Gateway local executando o IOS XE 17.6.1 ou superior
-
As Assinaturas de diagnóstico estão ativadas por padrão.
-
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
-
Configure a variável de ambiente ds_email com o endereço de e-mail do administrador para você notificar.
configure terminal call-home diagnostic-signature LocalGateway(cfg-call-home-diag-sign)environment ds_email <email address> end
Instalar assinaturas de diagnóstico para monitoramento proativo
Monitoramento de alta utilização da CPU
Este DS rastreia a utilização da CPU de 5 segundos usando o OID SNMP 1.3.6.1.4.1.9.2.1.56. Quando a utilização atingir 75% ou mais, ela desativa todas as depurações e desinstala todas as assinaturas de diagnóstico que você instalar no Gateway local. Use as etapas abaixo para instalar a assinatura.
-
Certifique-se de que você habilitar o SNMP usando o comando mostrar 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 Intercepta global SNMP: habilitada
Baixe o DS 64224 usando as seguintes opções suspensas na Ferramenta de pesquisa de assinaturas de diagnóstico:
copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software
Produto
CUBE Enterprise in Webex Calling solution
Escopo do problema
Desempenho
Tipo de problema
Alta utilização da CPU com notificação por e-mail
-
Copie o arquivo DS XML para o flash do Gateway local.
copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash:
O exemplo a seguir mostra a cópia do arquivo de um servidor FTP para o Gateway Local.
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)
-
Instale o arquivo DS XML no Gateway local.
call-home diagnostic-signature load DS_64224.xml Load file DS_64224.xml success
-
Use o comando mostrar assinatura de diagnóstico de chamada de casa para verificar se a assinatura foi instalada com êxito. A coluna status deve ter um valor "registrado".
show call-home diagnostic-signature Current diagnostic-signature settings: Assinatura de diagnóstico: Perfil habilitado: Cisco SENA-1 (status: ATIVO) URL(s) de download: https://tools.cisco.com/its/service/oddce/services/DDCEService Environment variable: ds_email: username@gmail.com
Baixe o DSes:
ID de DS
Nome DS
Revisão
Status
Última atualização (GMT+00:00)
64224
DS_LGW_CPU_MON75
0.0.10
Registrado
2020-11-07 22:05:33
Quando acionada, esta assinatura desinstala todos os DSs em execução, incluindo ela própria. Se necessário, reinstale o DS 64224 para continuar a monitorar a alta utilização da CPU no Gateway local.
Monitorando desconectações anormals de chamada
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.
-
Certifique-se de que o SNMP está habilitado usando o comando mostrar 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 Intercepta global SNMP: habilitada
-
Baixe o DS 65221 usando as seguintes opções na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Desempenho
Tipo de problema
Detecção anormal de desconexão de chamada SIP com e-mail e notificação Syslog.
-
Copie o arquivo DS XML para o Gateway local.
copy ftp://username:password@<server name or ip>/DS_65221.xml bootflash:
-
Instale o arquivo DS XML no Gateway local.
call-home diagnostic-signature load DS_65221.xml Load file DS_65221.xml success
-
Use the command show call-home diagnostic-signature to verify that the signature is successfully installed. A coluna de status deve ter um valor "registrado".
Instalar assinaturas de diagnóstico para solucionar um problema
Você também pode usar Assinaturas de Diagnóstico (DS) para resolver os problemas rapidamente. Os engenheiros do TAC da Cisco criaram várias assinaturas que permitem as depurações necessárias que são necessárias para solucionar um determinado problema, detectar a ocorrência do problema, coletar o conjunto certo de dados de diagnóstico e transferir os dados automaticamente para o caso do CISCO TAC. Isso elimina a necessidade de verificar manualmente a ocorrência do problema e torna a solução de problemas intermitentes e temporários muito mais fácil.
Você pode usar a Ferramenta de pesquisa de assinaturas de diagnóstico para encontrar as assinaturas aplicáveis e instalá-las para se auto-resolver um determinado problema ou você pode instalar a assinatura que é recomendada pelo engenheiro de TAC como parte do envolvimento do suporte.
Aqui está um exemplo de como encontrar e instalar um DS para detectar a ocorrência "%VOICE_IEC-3-GW: CCAPI: Erro interno (limite de pico de chamadas): IEC=1.1.181.1.29.0" syslog e automatize a coleta de dados de diagnóstico usando as seguintes etapas:
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
Exemplo:
call-home diagnostic-signature environment ds_fsurl_prefix " environment ds_fsurl_prefix "scp://612345678:abcdefghijklmnop@cxd.cisco.com"
-
Certifique-se de que o SNMP está habilitado usando o comando mostrar snmp. If SNMP not enabled, configure the snmp-server manager command.
show snmp %SNMP agent not enabled config t snmp-server manager end
-
Recomendamos instalar o monitoramento de alta CPU DS 64224 como uma medida proativa para desativar todas as assinaturas de depuração e diagnósticos durante o momento de alta utilização da CPU. Baixe o DS 64224 usando as seguintes opções na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Desempenho
Tipo de problema
Alta utilização da CPU com notificação por e-mail.
-
Baixe o DS 65095 usando as seguintes opções na Ferramenta de pesquisa de assinaturas de diagnóstico:
Nome do campo
Valor do campo
Plataforma
Cisco 4300, 4400 ISR Series, or Catalyst 8000V Edge Software
Produto
CUBE Enterprise na solução do Webex Calling
Escopo do problema
Syslogs
Tipo de problema
Syslog - %VOICE_IEC-3-GW: CCAPI: Erro interno (limite de pico de chamadas): IEC=1.1.181.1.29.0
-
Copie os arquivos DS XML para o Gateway local.
copy ftp://username:password@<server name or ip>/DS_64224.xml bootflash: copy ftp://username:password@<server name or ip>/DS_65095.xml bootflash:
-
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
-
Verifique se a assinatura foi instalada com êxito usando show call-home diagnostic-signature. A coluna de status deve ter um valor "registrado".
show call-home diagnostic-signature Current diagnostic-signature settings: Assinatura de diagnóstico: Perfil habilitado: Cisco SENA-1 (status: ATIVO) URL(s) de download: 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
DSes baixados:
ID de DS
Nome DS
Revisão
Status
Última atualização (GMT+00:00)
64224
00:07:45
DS_LGW_CPU_MON75
0.0.10
Registrado
2020-11-08:00:07:45
65095
00:12:53
DS_LGW_IEC_Call_spike_threshold
0.0.12
Registrado
2020-11-08:00:12:53
Verificar a execução de assinaturas de diagnóstico
No seguinte comando, a coluna "Status" do comando mostra as alterações de assinatura de diagnóstico de chamada de casa para "em execução" enquanto o Gateway local executa a ação definida dentro da assinatura. O resultado de mostrar as estatísticas de diagnóstico de chamada de casa é a melhor maneira de verificar se uma assinatura de diagnóstico detecta um evento de interesse e executa a ação. A coluna "Acionado/Máximo/Desinstalação" indica o número de vezes que a assinatura dada acionou um evento, o número máximo de vezes que é definido para detectar um evento e se a assinatura desinstala-se depois de detectar o número máximo de eventos acionados.
show call-home diagnostic-signature Current diagnostic-signature settings: Assinatura de diagnóstico: Perfil
habilitado: Cisco SENA-1 (status: ATIVO)
URL(s) de download: 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
DSes baixados:
ID de DS |
Nome DS |
Revisão |
Status |
Última atualização (GMT+00:00) |
---|---|---|---|---|
64224 |
DS_LGW_CPU_MON75 |
0.0.10 |
Registrado |
2020-11-08 00:07:45 |
65095 |
DS_LGW_IEC_Call_spike_threshold |
0.0.12 |
Em execução |
2020-11-08 00:12:53 |
mostrar estatísticas de assinatura do diagnóstico de chamada de casa
ID de DS |
Nome DS |
Triggered/Max/Deinstall |
Tempo médio de execução (segundos) |
Tempo máximo de execução (segundos) |
---|---|---|---|---|
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 |
O e-mail de notificação que é enviado durante a execução da Assinatura de Diagnóstico contém informações importantes como tipo de problema, detalhes do dispositivo, versão do software, configuração em execução e mostra saídas de comando que são relevantes para solucionar o problema dado.
Desinstalar assinaturas de diagnóstico
Use as assinaturas de diagnóstico para fins de solução de problemas que são tipicamente definidas para desinstalar após a detecção de algumas ocorrências de problemas. Se você desejar desinstalar uma assinatura manualmente, recupere a ID DS da saída da assinatura de diagnóstico de chamada de entrada de chamada de entrada e execute o seguinte comando:
call-home diagnostic-signature deinstall <DS ID>
Exemplo:
call-home diagnostic-signature deinstall 64224
Novas assinaturas são adicionadas periodicamente à Ferramenta de Pesquisa de Assinaturas de Diagnósticos, com base nos problemas que são observados nas implantações. Atualmente, o TAC não oferece suporte a solicitações de criação de novas assinaturas personalizadas.