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Configuring Voice Gateways

Cisco Unified Communications Manager supports a variety of Cisco Unified Communications gateway types. When the internal IP Telephony infrastructure needs to communicate with the PSTN and other non-IP telecommunications devices like private branch exchanges (PBXs), key systems, analog devices such as analog phones, fax machines, and modems, it utilizes the Voice (VOIP) Gateway for VOIP phone system call termination. Analog gateways play a crucial role in integrating these legacy telephone systems with VoIP networks, ensuring seamless communication between traditional and modern systems by converting analog signals into digital data packets using internet protocol. Business VoIP services are essential for modernizing communication infrastructure, enabling businesses to upgrade their systems efficiently.

Trunk interfaces specify how the gateway communicates with the PSTN or other external devices via time-division-multiplexing (TDM) signaling. Cisco Unified Communications Manager and Cisco gateways use a variety of TDM interfaces, however, the supported TDM interfaces may vary based on the gateway generation. Generally, the Cisco Unified Communications Manager supports two primary gateway types: Media Gateway Control Protocol (MGCP) and H.323 gateways, depending on the controlling protocol. These gateways convert analog signals into digital data packets for transmission over IP networks. Additionally, analog telephone adapters (ATAs) are essential for integrating traditional telecommunication devices into VoIP systems, converting phone signals for digital transmission. Analog telephone adapters are particularly important for businesses transitioning from analog to Gateway VoIP, allowing them to leverage existing infrastructure while benefiting from modern VoIP functionalities. IP PBXs play a significant role in connecting traditional phone systems with VoIP technologies, facilitating seamless communication.

However, in recent years, there have been more and more implementations of SIP trunks between the CUCM and the voice gateway, sometimes referred to as the Cisco Unified Border Element (CUBE). The public switched telephone network (PSTN) contrasts with digital VoIP systems, which offer cost benefits by transitioning to IP-based solutions. Voice over internet protocol gateway (VoIP) is crucial in modern telecommunications, enabling efficient and flexible communication.

 

Task 1: Add a H.323 Gateway to Cisco Unified Communications Manager for VoIP over an IP network

In this task, we will add both routers as H.323 gateways to Cisco Unified Communications Manager.

Activity Procedure

Complete these steps:

  1. On the Cisco Unified Communications Manager Administration page, from the menu, select Device > Gateway and click Add New.
  2. For gateway type, select H.323 Gateway, and click Next.
  3. On the Gateway Configuration page, use the following parameters:
    – Device Name: 10.1.1.101 (the IP address of the H323 Gateway connected to the Voice LAN)
    – Device Pool: SanJose
    – Uncheck Wait for Far End Terminal Capability Set (H.245)
    – Significant Digits: 4 (four)
    – Leave other parameters to default.
  4. Click Save.
  5. Click the Reset and another Reset buttons.
  6. Now the IP address of the H323 gateway will appear.
  7. To configure the second router, repeat steps 1–5 with the following parameters:
    – Device Name: 10.1.1.102 (the IP address of the H323 Gateway connected to the Voice LAN)
    – Device Pool: Chicago
    – Uncheck Wait for Far End Terminal Capability Set (H.245)
    – Significant Digits: 4 (four)
    – Leave other parameters to default.

Activity Verification for Integrated Services Digital Network

Once you achieve these results, you have successfully completed this task.

  • Both routers are listed as a H.323 gateway in Cisco Unified Communications Manager.

Configured H.323 VOIP Gateways

Task 2: Configure a voice gateway for Cisco Unified Communications Manager for VoIP controlled H.323 SIP Trunk

In this task we will configure both routers as a H.323 voice gateways controlled by Cisco Unified Communications Manager.

Activity Procedure

Complete these steps:

  1. Open a telnet session to the router Site 1.
  2. Type in configure terminal to access the configuration mode.
  3. Type in voice service voip command to enter into Voice over IP configuration.
  4. Type in allow-connections h323 to sip in order to enable H.323 to SIP communication.
  5. Type in allow-connections sip to h323 in order to enable SIP to H.323 communication.
  6. Enter exit to leave Voice over IP configuration.
  7. Enter interface configuration mode with interface FastEthernet 0/0 command.
  8. Type in h323-gateway voip bind srcaddr 10.1.1.101 command.
  9. Enter exit to leave interface configuration mode.
  10. Type in dial-peer voice 14085552 voip command to create dial peer.
  11. Enter a destination pattern with destination-pattern 14085552… command.
  12. Enter session target with session target ipv4:10.1.1.10 command.
  13. Specify the codec to be used when establishing a session through this dial-peer with codec g711ulaw.
  14. Type Exit.
  15. Type in dial-peer voice 9 voip command to create another dial peer.
  16. Enter a destination pattern with destination-pattern 9.T command.
  17. Enter session target with session target ipv4:10.1.2.2 command.
  18. Configure the signalling protocol with session protocol sipv2.
  19. Specify the codec to be used when establishing a session through this dial-peer with codec g711ulaw.
  20. Enter exit to leave dial peer configuration mode.
  21. Type in exit to leave configuration mode.
  22. Save router configuration with write command.
  23. Repeat steps 111111 to configure the second router but change the parameters to match the specification:
    – dial-peer voice 9 voip – session target ipv4:10.1.2.1
    – dial-peer voice 13124443 voip – destination-pattern 13124443…

Activity Verification

You have completed this task when you attain these results:

– Execute the command show dial-peer voice summary and get the following output

R1#show dial-peer voice summary
dial-peer hunt 0
             AD                                    PRE PASS                OUT
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT
14085- voip  up   up             14085552…        0  syst ipv4:10.1.1.10
552
9      voip  up   up             9.T                0  syst ipv4:10.1.2.2
R2#show dial-peer voice summary
dial-peer hunt 0
             AD                                    PRE PASS                OUT
TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT
9      voip  up   up             9.T                0  syst ipv4:10.1.2.1
13124- voip  up   up             13124443…        0  syst ipv4:10.1.1.10
443