Cisco Systems H.323/SIP manual MeetingPlace IP outdial. Phone=651515 IRC=0 PSTN=46 Unit=0

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Chapter 4 Troubleshooting Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Troubleshooting Caller Connectivity

Checking the Cisco Unified MeetingPlace Audio Server System When Unable to Dial Out on IP Ports

Step 1 Verify that incoming calls to the server are connecting. If not, perform the following procedures:

•Checking the Cisco Unified MeetingPlace Audio Server System When IP Ports Do Not Answer, page 4-4

•Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server When IP Ports Do Not Answer, page 4-4

•Checking Cisco Unified CallManager When IP Ports Do Not Answer, page 4-5

Step 2 Verify that the port group is enabled for outgoing calls by using the port command.

Step 3 Check the translation table to verify IP calls are being directed to a port group that is configured for IP.

Tip You can use the xltest utility to check which port group will be used for the dialed number. This is especially important for mixed PSTN and IP systems.

Step 4 At the tech$ prompt, enter cptrace -T 5.

Step 5 While monitoring the output of the trace command, make a test call.

Step 6 At the tech$ prompt, enter viewexlog -s info -l more.

Tip Enter f to move forward in the log.

Step 7 Check for warnings and alarms, especially those that occur in “cpiphandler.cc” and “cpplacecall.cc”. Step 8 At the tech$ prompt, enter activity.

Step 9 Choose option 4 to make a test call.

Step 10 Test internal extensions and outside numbers to isolate the problem.

Checking the Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Server When Unable to Dial Out on IP Ports

Step 1 Open the Cisco Unified MeetingPlace Gateway SIM eventlog and verify that the IP-gateway server receives the outdial command from the Cisco Unified MeetingPlace Audio Server system.

Step 2 In the Cisco Unified MeetingPlace Gateway SIM eventlog, verify that the correct phone number was received by the IP-gateway server, as shown in the following example:

MeetingPlace IP outdial. Phone=651515 IRC=0 PSTN=46 Unit=0

Step 3 In the Release 5.2.1 configuration, verify that the outdial is sent by using the appropriate protocol.

Step 4 Verify that the gateway, gatekeeper, and proxy server addresses and ports are correct according to the desired protocol.

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

 

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Contents Text Part Number OL-6571-02 Corporate HeadquartersCopyright 2005-2006 Cisco Systems, Inc. All rights reserved N T E N T S Fast Busy Signal Heard When Using a Cisco IP Phone Worksheets OL-6571-02 Scope AudienceProduct Naming Convention New Features in This ReleaseFeature Description Naming Conventions Used in This GuideCisco Unified MeetingPlace System SIP RTP SIP Protocol DescriptionAudio Quality During a Cisco Unified MeetingPlace Meeting Step Cisco IP Phone Description Pstn Phone Description Step Cisco IP Phone Description Pstn Phone Description Step Device Description Cisco SIP IP Phone Description Step H.323 Device Description Cisco SIP IP Phone Description Additional ReferencesA P T E R CallManager Administration Field Description Click the Add a New Gateway linkChoose Originator Field Description Task Tone box Server to the Cisco Unified CallManager ConfigurationChoose Start Run Select MeetingPlace IP Gateway and click Remove Configuring Cisco Unified MeetingPlace Gateway SIMChanging Cisco Unified MeetingPlace Gateway SIM Settings To apply the configuration settings, click OK again OL-6571-02 A P T E R General Settings Setting Description DefaultSettings SIP Settings Field Name Setting SIP Field Name Setting Field Task Assigning the Primary IP AddressChoose Specify an IP address How to Configure a Dialing Group Information About Configuring a Dialing GroupConfiguring a Dialing Group Example OL-6571-02 Troubleshooting Network Connectivity Unable to Make Calls From a Cisco IP Phone Troubleshooting Caller ConnectivityDead Air Heard When Using a Cisco IP Phone Dead Air Heard When Using an H.323 DeviceFast Busy Signal Heard When Using a Cisco IP Phone MP Resp. Msg=3 CPerr=0 SeqNum=0x16 Page Unable to Dial Out on IP Ports MeetingPlace IP outdial. Phone=651515 IRC=0 PSTN=46 Unit=0 Poor or Low-Audio Quality Troubleshooting Audio ProblemsEcho OL-6571-02 Description Value WorksheetsDescription Value MeetingPlace IP call flow Value OL-6571-02 D E IN-2 IN-3 IN-4

H.323/SIP specifications

Cisco Systems has been a leading force in the development and implementation of voice and video communication technologies, prominently featuring H.323 and SIP (Session Initiation Protocol). These protocols have become cornerstones in the realm of IP-based communication, facilitating seamless interaction across disparate devices and networks.

H.323 is a set of protocols that provides multimedia communication over packet-switched networks, such as the Internet. It supports audio, video, and data communications across IP networks, utilizing a variety of components including terminals, gateways, and multipoint control units (MCUs). One key feature of H.323 is its ability to handle both point-to-point and multipoint video conferencing. This makes it particularly suitable for enterprise applications where group communication is essential.

On the other hand, SIP is a more recent and flexible signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video, and messaging applications. Renowned for its simplicity and interoperability, SIP can work with various communication mediums and provides extensive capabilities for managing multimedia sessions. One of the primary advantages of SIP is its scalability, allowing it to accommodate everything from small-scale personal communications to vast corporate systems.

Both H.323 and SIP support features such as call transfer, call hold, and caller ID. H.323, however, can be more complex due to its broader array of standards and components, which might require substantial configuration. SIP, conversely, is designed to be lightweight, easily integrated with existing systems and applications, making it more user-friendly for developers.

Cisco Systems enhances these protocols through their robust telecommunication infrastructure that facilitates performance optimization, security, and unparalleled user experience. With advanced technologies like Cisco Unified Communications Manager and Cisco Webex, organizations can leverage H.323 and SIP to create cohesive communication environments. Security features such as encryption and authentication ensure that sensitive conversations remain private and secure, while Quality of Service (QoS) protocols manage bandwidth effectively to maintain consistent call quality.

Ultimately, the combination of H.323's established framework and SIP's flexibility ensures that organizations using Cisco Systems can effectively manage their communication needs, fostering collaboration and connectivity in today's fast-paced digital landscape. These protocols continue to evolve, adapting to the ever-increasing demands placed on modern communication systems.
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