You do not need to manually dial a telephone number on the desk phone key pad and contacts on the Cisco Jabber application are identified from your Jabber contacts, the Tufts Directory, and your Outlook contacts if you are using a PC.Use the application-name argument to define a specific interactive voice response (IVR) application.These services are used by a SIP IP phone in the event of a WAN connection outage when the SIP phone is unable to communicate with its primary SIP proxy.
The Cisco Unified SIP SRST device also provides PSTN gateway access for placing and receiving PSTN calls. This is done because during a WAN failure (and the resulting Cisco Unified SIP SRST functionality), DNS servers may not be available. By default, Cisco Unified SIP SRST is not enabled and cannot accept SIP Register messages. The following configuration must be set up to accept incoming SIP Register messages. Cisco Voip Phone Setup Rar Service ToSee the Configuring Backup Registrar Service to SIP Phones section. The voice register pool configuration provides registration permission control and can also be used to configure some dial-peer attributes that are applied to the dynamically created VoIP dial peers when SIP phone registrations match the pool. Registrations that match this pool create VoIP SIP dial peers with the dial-peer attributes set to these configurations. Although only the id command is mandatory, this configuration example shows basic functionality. See the Configuring the SIP Registrar section for complete instructions. Thus, the id command configured in Step 5 is required and must be configured before any other voice register pool commands. Cisco Voip Phone Setup Mac Address KeywordWhen the mac address keyword and argument are used, the IP phone must be in the same subnet as that of the routers LAN interface, such that the phones MAC address is visible in the routers Address Resolution Protocol (ARP) cache. Once a MAC address is configured for a specific voice register pool, remove the existing MAC address before changing to a new MAC address. When a SIP phone registers to Cisco Unified SIP SRST and the proxy command is enabled, two dial peers are automatically created. The first dial peer routes to the proxy, and the second (or fallback) dial peer routes to the SIP phone. The same functionality can also be achieved with the appropriate creation of static dial peers (manually creating dial peers that point to the proxy). Proxy dial peers can be monitored to one proxy IP address, only. That is, only one proxy from a voice registration pool can be monitored at a time. If more than one proxy address needs to be monitored, you must manually create and configure additional dial peers. Cisco Voip Phone Setup Full Information OnFor full information on the call fallback active command, see PSTN Fallback Feature. The preference must be less (higher priority) than the preference configured with the preference command. The tag argument is a codec group number between 1 and 10000.
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