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CISCO UNIFIED IP PHONE TUTORIALS

2015-02-12T12:14:00.002+05:30

CISCO UNIFIED IP PHONE 7975G

Get an in-depth view of the features and functionality of Cisco Unified IP Phone 797xG models by accessing the tutorial linked below.

CISCO UNIFIED IP PHONE 7971G-GE

Get an in-depth view of the features and functionality of Cisco Unified IP Phone 797xG models by accessing the tutorial linked below.

CISCO UNIFIED IP PHONE 7970G

CISCO UNIFIED IP PHONE 7965G

Get an in-depth view of the features and functionality of Cisco Unified IP Phone 796xG models by accessing the tutorial linked below.

CISCO UNIFIED IP PHONE 7962G

Get an in-depth view of the features and functionality of Cisco Unified IP Phone 796xG models by accessing the tutorial linked below.

CISCO UNIFIED IP PHONE 7961G-GE

CISCO UNIFIED IP PHONE 7961G

CISCO UNIFIED IP PHONE 7960G

CISCO UNIFIED IP PHONE 7945G

Get an in-depth view of the features and functionality of Cisco Unified IP Phone 794xG models by accessing the tutorial linked below.

CISCO UNIFIED IP PHONE 7942G

Get an in-depth view of the features and functionality of Cisco Unified IP Phone 794xG models by accessing the tutorial linked below.

CISCO UNIFIED IP PHONE 7941G-GE

CISCO UNIFIED IP PHONE 7941G

CISCO UNIFIED IP PHONE 7940G

CISCO UNIFIED IP PHONE 7911G

CISCO UNIFIED IP PHONE 7912G

CISCO UNIFIED IP PHONE 7906G

All About Cisco MPLS

2015-02-12T12:04:00.001+05:30

Basic MPLS Concepts

• MPLS is a new forwarding mechanism in which packets are forwarded based on labels
• Labels may correspond to IP destination networks (equal to traditional IP forwarding)
• Labels can also correspond to other parameters (QoS, source address, etc.)
• MPLS was designed to support forwarding of other protocols as well

Presentation and Trainings
Introduction to MPLS PDF
Introduction to MPLS VPN PDF Video
Cisco MPLS Concepts PDF/Slides/Video
Application of Label Switched Multicast for Video Delivery
Building Enterprise MPLS Networks PDF
MPLS Developments and Advanced Concepts PDF
MPLS Traffic Engineering – Traffic Protection Using Fast Re-route (FRR) PDF
MPLS VPNs and Security PDF
Troubleshooting MPLS Networks PDF
Understanding How Routes are propagated in an MPLS VPN PDF
Configuring Simple MPLS VPNs – PDF

E-Learning Modules on MPLS VPN Technology
Introducing VPNs
Understanding MPLS Traffic Engineering Concepts
Configuring Small-Scale Routing Protocols Between PE and CE Routers
Advanced_MPLS_Traffic_Engineering__CourseWare_Configuration .Zip
MPLS Frame-Mode MPLS Lab CourseWare and ConfigurationFiles .Zip
MPLS_Concepts .Zip
MPLS_Initial_MPLS_VPN_Setup_CourseWare_Configuration .Zip
MPLS_Internet_Connectivity_Through_Route_Leaking_CourseWare_Configuration .Zip
MPLS_Internet_in_a_VPN_CourseWare_Configuration .Zip
MPLS_Network_Management_VPN_Topology_Configuration .Zip
MPLS_Overlapping_VPN_Topology_Configuration .Zip
MPLS_Separate_Interface_for_Internet_Connectivity_Configuration .Zip
Traffic_Engineering_in_MPLS_VPN_Environment_CourseWare_Configuration .Zip

If you have access to Cisco PEC (Partner Education Center) website, then here are few labs you can do online.
1. Advanced MPLS Traffic Engineering – Challenge Lab
2. MPLS Frame-Mode MPLS Configuration – Challenge Lab
3. MPLS Initial MPLS VPN Setup – Challenge Lab
4. MPLS Internet Connectivity Through Route Leaking – Challenge
5. MPLS Internet in a VPN – Challenge Lab
6. MPLS Network Management VPN Topology – Challenge
7. MPLS Overlapping VPN Topology – Challenge Lab
8. MPLS Running OSPF Between PE and CE Routers – Challenge Lab
9. MPLS Separate Interface for Internet Connectivity – Challenge Lab
10. MPLS Traffic Engineering in MPLS VPN Environment – Challenge Lab

Exams :
Implementing Cisco MPLS 642-611 MPLS
BGP + MPLS Exam 642-691 BGP + MPLS
MPLS Set 1 Practice Test
MPLS Set 2 Practice Test
Multi Player Game on MPLS and various other technologies

Books:
MPLS and VPN Architectures
MPLS and VPN Architectures, Vol. II

MPLS Q&A
MPLS FAQ For Beginners
Cisco Any Transport over MPLS Q&A
Cisco MPLS Tunnel Builder Q&A
MPLS for Managed Shared Services – Q&A
Cisco IOS MPLS Embedded Management Q&A

Design Guide
Cisco MPLS Tunnel Builder Product Details
MPLS Bandwidth Protection Brochure
MPLS Class of Service (CoS)
MPLS Traffic Engineering (TE)—Automatic Bandwidth Adjustment for TE Tunnels
MPLS Traffic Engineering (TE)—Automatic Bandwidth Adjustment for TE Tunnels
MPLS Traffic Engineering (TE)—Configurable Path Calculation Metric for Tunnels
MPLS Traffic Engineering (TE)—IP Explicit Address Exclusion
MPLS Traffic Engineering and Enhancements
MPLS Virtual Private Networks (VPN)
MPLS Virtual Private Networks (VPN) – Enhancements
MPLS VPN—Carrier Supporting Carrier
MPLS VPN—Carrier Supporting Carrier—IPv4 BGP Label Distribution
Changes in RSVP Debug Commands in Cisco IOS Software Release 12.0(24)S and Later
DiffServ Tunneling Modes for MPLS Networks
MPLS/VPN with EIGRP on the Customer Side Configuration Example
The Traceroute Command in MPLS
Cisco IOS MPLS High Availability: Component and Management
Network Integration with MPLS: MPLS High Availability

Configuration Guide
Configuring a Basic MPLS VPN
Configuring Basic MPLS Using IS-IS
Configuring Basic MPLS Using OSPF
Configuring MPLS Basic Traffic Engineering Using IS-IS
Configuring MPLS Basic VPN with RIP on Customer Side
Configuring VPN MPLS over ATM with Cisco 7500 Routers and LightStream 1010 Switches
DiffServ Tunneling Modes for MPLS Networks
GRE Tunnel with VRF Configuration Example
Implementing an MPLS VPN over TE Tunnels
Inter-AS MPLS VPN Configuration with VPNv4 eBGP Sessions Between ASBRs
Internet Access from an MPLS VPN Using a Global Routing Table
MPLS Basic Traffic Engineering Using OSPF Configuration Example
MPLS Command Guide
Multicast Support for MPLS VPNs Configuration Example

Troubleshooting
MPLS Troubleshooting
Changes in RSVP Debug Commands in Cisco IOS Software Release 12.0(24)S and Later
DiffServ Tunneling Modes for MPLS Networks
How to Troubleshoot the MPLS VPN
Implementing an MPLS VPN over TE Tunnels
MPLS/VPN with EIGRP on the Customer Side Configuration Example
The Traceroute Command in MPLS
Troubleshooting LSP Failure in MPLS VPN

MPLS Management
Cisco IOS MPLS Embedded Management Application Note
Cisco IOS MPLS Management Business Overview (PDF – 4 MB)
Cisco IOS MPLS Management Technical Deployment Overview (PDF – 2 MB)
Cisco IOS MPLS Management Technology Overview (PDF – 2 MB)
Cisco Multiprotocol Label Switching Management Strategy
Mulitprotocol Label Switching (MPLS) Label Switching Routers (LSR) have the MPLS-LSR-MIB to provide label switching information.

Refer to this table:

Table Name	SNMP Object	Purpose
MPLS Interface Configuration Table	mplsInterfaceConfTable	Provides information for each MPLS-capable interface on an LSR.
MPLS Interface Performance Table	mplsInterfacePerfTable	Augments the MPLS interface configuration table.
MPLS In-Segment Table	mplsInSegmentTable	Contains a description of incoming segments (labels) at an LSR and their associated parameters.
MPLS In-Segment Performance Table	mplsInSegmentPerfTable	Augments the MPLS in-segment table, providing performance information and counters for incoming segments on an LSR.
MPLS Out-Segment Table	mplsOutSegmentTable	Contains a description of outgoing segments from an LSR and their associated parameters.
MPLS Out-Segment Performance Table	mplsOutSegmentPerfTable	Augments the MPLS out-segment table, providing performance information and counters for outgoing segments on an LSR.
MPLS Cross-Connect Table	mplsXCTable	Associates inSegments (labels) to outSegments (labels) to show the manager how the LSR is currently swapping these labels.

White Papers
Cisco IOS MPLS Quality of Service
Deploying Guaranteed-Bandwith Services with MPLS
Internet Connectivity Options
Managed VPN – Multicast Music-On-Hold over Permanent Virtual Circuit Switched Networks
MPLS Bandwidth Protection White Paper
Positioning MPLS
Security of the MPLS Architecture
Service Definitions: Virtual Leased Lines
Service Definitions: Voice & Toll Bypass Trunking
Tunnel Builder White Paper (PDF – 1 MB)
Understanding MPLS-TP and Its Benefits
Cisco IOS Software and Multiprotocol Label Switching
IETF MPLS Charter
MPLS Bandwidth Protection (PPT – 2 MB)
MPLS for Managed Shared Services – Data Sheet
MPLS for Managed Shared Services – Customer Presentation (PPT – 2 MB)
Tunnel Builder Application Brief (PDF – 390 KB)
Tunnel Builder v1.0: A Light at the End of the Tunnel (PDF – 71 KB)

RFC
RFC 3031 – MPLS Architecture
RFC 2917 – A core MPLS IP VPN Architecture

Terminology
1. Customer Edge Router (CE)—Router at the edge of a network and that has interfaces to at least one PE router.
2. Data Multicast Distribution Tree (MDT)—Tree created dynamically by the existence of active sources in the network and that is sent to active receivers located behind separate PE routers. Data MDT connects only to PE routers that are attached to CE routers with active sources or receivers of traffic from active sources or that are directly attached to active sources or receivers of traffic.
3. Default-MDT—Tree created by the multicast virtual private network (MVPN) configuration. The Default-MDT is used for customer Control Plane and low rate Data Plane traffic. It uses Routing and Forwarding (MVRFs) to connect all of the PE routers with in a particular multicast domain (MD). One Default-MD exists in every MD whether there is any active source in the respective customer network.
4. LEAF—Describes the recipient of multicast data. The source is thought of as the route and the destination is the leaf.
5. Multicast domain (MD)—Collection of MVRFs that can exchange multicast traffic
6. Multicast Virtual Route Forwarding (MVRF)—Used by a PE router to determine how to forward multicast traffic across an MPLS core.
7. Provider Router (P)—Router in the core of the provider network that has interfaces only to other P routers and other PE routers
8. Provider Edge Router (PE)—Router at the edge of the provider network that has interfaces to other P and PE routers and to at least one CE router
9. PIM-SSM—PIM Source Specific Multicast
10. TE Tunnel: A Unidirectional Label Switched Path (LSP) across an MPLS network
12. RSVP: Resource Reservation Protocol, the protocol used to establish MPLS TE tunnels
13. RSVP-TE: RSVP extensions for MPLS TE (RFC 3209)
14. CR-LDP: Constraint-based LSP setup using LDP (RFC 3212)
15. CBR: Constraint-based routing
16. Head-end: The router initiating and MPLS TE tunnel
17. Tail-end: The router on which an MPLS TE tunnel terminates
18. PUSH : Adding a Label to Incoming Packet and also aka Label imposition
19. SWAP : Replace the label of incoming packet
20. POP : Remove the label from outgoing packet and also aka Label disposition.

CUCM IP Phone Firmware Upgrade Process

2015-02-10T16:16:00.001+05:30

7941-7961 phone firmware upgrade

This same process works for all phones, but the phone firmware load names will change accordingly.

1. get your ES firmware available and "unzip" it on your desktop.
2. get FreeFTPd software and set up a user with a password and point the directory to your desktop so the files will be found and used from there.
3. open the CUCM OS Admin web page you should go to Software Upgrades > Install/Upgrade

Fill out all the correct information, make sure the SFTP server is STARTED!!

Then click “next” on the page…

If it works correctly you should see the following….which is the name of the file that the server will use for all the loads that you are wanting to use.

The icon will “scroll” and that’s how you can confirm that its working, don’t touch it at this point, let it continue on.

Next you will see this information screen…

Keep going, click “next”

Then you will see…

Don’t touch anything, this will run through just like its supposed to do.

Then you will see this screen if everything worked out perfectly…

Don’t click “install another”, because its done!!

Now you need to go verify that everything is working correctly.

Do the following…

Go to Software Upgrades > TFTP File Management

Then do a search for your load version…something like 8-4-1 and you will see something like this…click “contains” on the search…

Those are all the files that the phone will use during the download process to upgrade and work properly. So we know its worked because those shouldn't have been there prior to this “upgrade”

Go back to the CUCM Admin Webpage…

Go to Device > Device Settings > Default Devices ….Then check the 7941/7961 SCCP or SIP phones “Load Information” and it should say something like…SCCP41.8-4-1ES10S which is the exact file that we installed earlier. By confirming this we now know that the SFTP upload went perfectly and it changed this information for us automatically. Now we need to finish up the rest.

Go restart the TFTP service on the Serviceability page to enable to new changes that are being made now. This will ensure that the “new” information gets pushed instead of the old firmware.

After restarting the service, go and unplug or “reset and restart” the phone. Once this is done you will then need to go to the phone web page and verify that its taken the new load. See the screen below…

Once it says the exact version that you are wanting, then you are all set and ready to make calls.

Cisco Cue Backup Procedure

2015-02-10T13:06:00.003+05:30

CONFIGURING THE BACKUP SERVER

Before you begin the backup process from the Administration Backup/Restore Start Backup window, you must first set the parameters in this window.

Use this procedure to configure the backup server.

Choose Administration > Backup/Restore > Configuration.
Enter the information shown in the following fields:
Server URL—The URL of the server on the network where backup files are stored. The format should be ftp://server/directory/ where server is the IP address or hostname of the server.
User ID—The account name or user ID on the backup server. You must have an account on the system to which you are backing up your data. Do not use an anonymous user ID.
Password—The password for the account name or user ID on the backup server.
Confirm Password—The password entered again.
Maximum Revisions—The maximum number of revisions of the backup data that you want to keep on the server. The maximum number is 50. The default value is 5.
Click Apply to save the information.

STARTING A BACKUP

You must do the following before starting a backup:

Configure the server used to back up the data. See Configuring the Backup Server.
Save your configuration. See Saving or Reloading Your Configuration.

Caution : Performing a backup for the system data and configuration requires taking the system offline. Active voice-mail calls at that time are dropped, and callers reaching voice mail during the backup receive a message that the voice-mail system is unavailable. Active calls to the autoattendant are not dropped and the autoattendant continues to receive and process calls during the backup.

Use this procedure to begin the data backup process.

Click Administration > Backup/Restore > Start Backup. The Start Backup window appears and the system automatically generates a backup ID. The backup ID number increases by 1 every time you back up the server.
Enter a description of the backup file; for example, “backupdata6-2-04.”
Select the category of backup, or type of data that you want to save:
Configuration—Saves the configurations of the system and applications.
Data—Saves your voice-mail messages.
Historical Reporting Data—Saves your historical reporting information.
Click Start Backup.

Interview Question for GK & CUBE

2012-10-17T15:55:00.002+05:30

What are the benefits of using Cisco Unified Border Element ?
A:
• Multiple physical interconnects, intelligent OAM, Call Admission Control, Billing etc
• Security Demarc: FireWall, DOS protection, VPN, etc
• Signaling, Protocol & Media Interworking: H.323/SIP, transcoding, DTMF etc.
• Media QoS control and monitoring
• Interoperability: various network elements, CUCM, etc.
• Co-existence/co-operation with TDM trunking

Q: What type of Calls are supported in CUBE ?

A: CUBE supports,
• IP-to-IP Calls
• TDM Voice Calls
• Gatekeeper
• Routing

Q: How the Media is handled in CUBE ?

A: Media is handled in 2 different modes

Media Flow-Through
Media Flow-Around

Q: What Platforms are supported for CUBE?

A: Cisco CUBE is an Integrated application with Cisco IOS software.
CUBE functionality is supported in Cisco 2600XM, 2691, Cisco ISR 2800, 3800 series, Cisco VXR 7200, Cisco XR 12000, AS5400XM Universal gateways and the Service Provider Gateways.

Q: Do we need Digital signal processors (DSPs) for CUBE?

A: Digital Signal Processors (DSPs) are only required for calls with dissimilar codecs

Q: What is the licensing requirements of the CUBE?

A: If you have any existing Cisco Gateway then you can download the Cisco IOS supported version for running CUBE on your Gateway. Also you need to buy license for the no. of sessions in CUBE for different platform. Say for an example if you want to have a CUBE supporting 500 sessions , you need to buy the license for this amount.

Q: Do CUCM need special license for CUBE or it will be just a SIP trunk without license ?
A: No, there is no special license for connecting CUCM to CUBE..

Q: WE have a CUBE for a SIP connection. I know there is a license that we purchased from Cisco. Is there a command to enter on the CUBE to view the licenses?

A: The CUBE licenses are not entered into the router. As of now, they are paper licenses.

Q: Whether CUBE supports Voice and Data traffic simultaneously?

A: Yes. CUBE is in IOS based platform hence it can handle both Voice and Data traffic.

Q: Does CUBE have to be run on a dedicated device/router?

A: Not Necessary. It depends on the voice traffic you want to route using CUBE. For large enterprise its recommended to use standalone device for CUBE functionality. If the traffic is not high you can run other services along with the CUBE application for voice on the same platform simultaneously.

Q: Do we need DSP resources in CUBE if the SIP trunk from city (external) to the SIP trunk of CUCM will be used ? IP phones will use g711 codec. It could be e a need in DSP resources for audio conference and others supplementary services…

A: No DSP resources are required to simply connect to SIP Trunk. DSP resources are required if you plan on doing transcoding between g711 to g729 codec. For conferences if all parties are using g711 codec, then you can use the software resources in UCM to do the conferencing. If one of the parties is using g729 codec and if that needs to be conferenced, you will need to use voice gateways for hardware conferencing which will require DSPs. Digital Signal Processors (DSPs) are only required for calls with dissimilar codecs.

Q: Will SRST work on CUBE ?
A: Yes, SRST and CUBE can be co-located on the same router.

Q: Explain the Call flow in CUBE.

A:

1.Incoming VoIP setup message from OGW to CUBE
2.This matches an inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF method, protocol etc
3.CUBE then looks up called number in setup and matches outbound VoIP dial peer 2
4.Outgoing VoIP setup message from CUBE to TGW
Q: How to enable the IP-IP Calls and Protocol support in CUBE ?

A: Enabling the IP-to-IP Calls
CUBE#config t
CUBE(config)# voice service voip
CUBE(conf-voi-serv)#allow-connections h323 to h323
CUBE(conf-voi-serv)#allow-connections h323 to sip
CUBE(conf-voi-serv)#allow-connections sip to h323
CUBE(conf-voi-serv)#allow-connections sip to sip

It is mandatory to have Incoming and Outgoing VoIP Dial-peers with required parameters like Protocol, Transport, Codec, CAC, QoS, etc

Q: Whether CUBE supports Universal Transcoding ?

A: Yes CUBE supports Universal Transcoding, any to any Voice codec. Example. iLBC to G.711 or iLBC to G.729
List of codecs supported

g711alaw 64Kbps
g711ulaw 64Kbps
g723r53 5.3Kbps
g723r63 6.3Kbps
g723r63 6.3Kbps
g729 (all variants) 8Kbps
iLBC

Q: What is CAC ?

A: CAC is Call Admission Control. It control number of calls based on resources and bandwidth and Proactively reserve resources for good quality video calls. Also it ensure traffic adheres to QoS policies within each network. CUBE can provide six different CAC mechanisms

Q: Is CP-PWR-CUBE-4= compatible with the 7900 series phones?

A: No this power cube is for the new 8900/9900 phones, compatable with Communications Manager. The power tip is different size, won’t fit in other phones.

Q: Im trying to find a way to route inbound SIP calls, based on their source. For example:

SIPCarrier1 is coming from 1.1.1.1
SIPCarrier2 is coming from 2.2.2.2

If calls originate from SIPCarrier1, I want them to be passed to my UCM Cluster.

If call originate from SIPCarrier2, I want them to be passed to my Asterisk box.

I’ve played around with incoming called-number, trying to make it work with IP addresses, rather than phone numbers, but no luck.

Any ideas on how to enable source based routing?

A: Yes this can be achieved by Carrier based Routing. Refer this link for more details.

VoIP Gateway Trunk and Carrier Based Routing Enhancements

Q: CUBE Sizing, how many simultaneous calls/ sessions are supported on each Router model to terminate a SIP Trunk at the service provider end ?

A: Refer the given table for more details.

Troubleshooting T1 and E1 Problems

2012-10-17T15:45:00.001+05:30

T1 or E1 Interface Troubleshooting
To troubleshoot the T1 or E1 interface, perform the following steps:

SUMMARY STEPS

show controller {t1 | e1}
Check if the line is down.
Check for reported alarms.
Check for error events.
Check if the interface is T1 or E1 CAS, E1 R2, or PRI.

DETAILED STEPS

1. Enter the show controller t1 or show controller e1 command with the controller number for the voice port you are troubleshooting.

 Router# show controller {t1 | e1} controller-number

2. Check if the line is down. If so, see the Troubleshooting T1 and E1 Layer 1 Problems.
3. Check if there are any reported alarms. If so, refer to T1 Alarm Troubleshooting, document ID 14172 to troubleshoot alarm indications.
4. Check if there are any error events. If you encounter framing, line coding, or clock timing errors, see the Checking T1/E1 Controller Configuration.
Refer to T1 Alarm Troubleshooting Flowchart, document ID 14174 for a flowchart to troubleshoot error events.
5. Check if the interface is T1 or E1 CAS, E1 R2, or PRI. See the following sections for more information:

Troubleshooting T1 and E1 Layer 1 Problems

Most T1 and E1 errors are caused by incorrectly configured lines. Ensure that line coding, framing, and clock source are configured according to the recommendations of your Service Provider.
Use the show controllers t1 and show controllers e1 commands in privileged EXEC mode to display information about the T1 or E1 links or to display the hardware and software driver information for the controller. These commands show what state the T1 or E1 controller is in. The controller can be in one of three states:

Administratively down-If the controller is administratively down, you can manually bring it up using the procedure in the Controller Is Administratively Down section.
Down-If the controller is down, then the cause is one of the following:
- Loss of frame-See the Controller Has Loss of Frame section to resolve this issue.
- Loss of signal-See the Controller Has Loss of Signal section to resolve this issue.
Up-The controller is functioning properly on Layer 1.

Solutions for bringing the controller up follow.

Controller Is Administratively Down

The controller is administratively down when it has been manually shut down. Follow these steps to restart the controller to correct this error.

SUMMARY STEPS

enable
configure {terminal | memory | network}
controller t1 number
no shutdown
end

DETAILED STEPS

1. Enable privileged EXEC mode: enable Example:

 Router> enable

Enter your password if prompted.
2. Enter global configuration mode:
configure terminal
Example:

Router(config)# configure terminal

3. Enter controller configuration mode:
controller t1 number
The number syntax is platform-specific. For more information about the syntax of this command, see the Cisco IOS Interface and Hardware Component Command Reference.
Example:

Router(config)# controller t1 0

4. Restart the controller: no shutdown
Example:

Router (config-controller)#  no shutdown

5. Exit to privileged EXEC mode: end
Example:

Router(config)# end

What to Do Next

The controller should be running and normal configuration can continue.

Controller Has Loss of Frame

Complete the following steps if the receiver has loss of frame:

Ensure that the framing format configured on the port matches the framing format of the line. Check the framing format of the controller from the running configuration or the show controller t1 or show controller e1 command output. To change the framing format, use the framing command in controller configuration mode.

For T1, the options are sf and esf. For example:

Router# configure terminal  
Enter configuration commands, one per line. End with CNTL/Z.  
 Router(config)# controller t1 0  
 Router(config-controlle)# framing esf

For E1, the options are crc4 and no-crc4. For example:

Router>configure terminal  
Enter configuration commands, one per line. End with CNTL/Z.  
Router(config)# controller e1 0  
Router(config-controlle)# framing crc4

If the first framing format does not work, try the other framing format to see if the alarm clears. For more information on framing formats, see the Framing Formats on Digital T1/E1 Voice Ports.

For T1 lines, change the line build-out setting using the cablelength long or cablelength short command.

Line build-out (LBO) or cable length compensates for the loss in decibels based on the distance from the device to the first repeater in the circuit. A longer distance from the device to the repeater requires that the signal strength on the circuit be boosted to compensate for loss over that distance.

To configure transmit and receive levels for a cable length (line build-out) longer than 655 feet for a T1 trunk with a channel service unit (CSU) interface, use the cablelength long controller configuration command. To configure transmit attenuation for a cable length (line build-out) of 655 feet or shorter for a T1 trunk with a DSX-1 interface, use the cablelength short controller configuration command.

Contact your service provider and refer to the Cisco IOS Interface and Hardware Component Configuration Guide for details on build-out settings.

What to Do Next

If the preceding steps do not fix the problem, see the Controller Has Loss of Signal section.

Controller Has Loss of Signal

If your controller is experiencing loss of signal, check the following.

Note:

Use the show controller t1 EXEC command after each step to see if the controller exhibits any errors.

Ensure that the cable between the interface port and the T1 service provider’s equipment or T1 terminal equipment is connected correctly. Ensure that the cable is hooked up to the correct ports. Correct the cable connections if necessary.
Check the cable integrity by looking for breaks or other physical abnormalities in the cable. Ensure that the pinouts are set correctly. Replace the cable if necessary.
Check the cable connectors. A reversal of the transmit and receive pairs or an open receive pair can cause errors. Depending on the type of module used, the cable terminates on a male DB-15 or RJ-45/48 connector.

On a DB-15 connector, the receive pair should be on pins 2 and 9, and the transmit pair on pins 8 and 15. A DB-15 connector is shown in Figure: DB-15 Connector.

Figure: DB-15 Connector

The pins on a RJ-45/48 jack are numbered from 1 through 8. With the metal pins facing toward you, pin 1 is the left-most pin. Figure: RJ-45 Pin Numbering shows the pin numbering on an RJ-45 jack. The receive pair should be on lines 1 and 2, and the transmit pair should be on lines 4 and 5.

Figure: RJ-45 Pin Numbering

If you have completed all of the steps above and you are still experiencing problems, try using a crossover cable. Crossover cable pinouts are shown in the Using the Cisco 2524-2525 Back-to-Back document. Additionally, you can try a rollover cable to check if the T1/E1 is passing through a reverse-wired wiring path. Pin locations are shown in Figure: Pin Location.

Figure: Pin Location

Checking T1/E1 Controller Configuration

Digital T1/E1 voice ports must have controller configurations that match the configuration of the router to the line characteristics of the telephony network connection being made so that voice and signaling can be transferred between them. This configuration also affects allows the logical voice ports, or DS0 groups, to be established. Make sure to check the following:

Contact your service provider for framing and line coding settings. Common settings are as follows:

For T1 lines, it is common to use binary 8-zero substitution (B8ZS) line coding with extended superframe (ESF), and alternate mark inversion (AMI) line coding with superframe (SF).
For E1 lines, both HDB3 and AMI line coding are available, but CRC4 framing is most widely used.

Framing Formats on Digital T1/E1 Voice Ports

The framing format parameter describes the way that bits are robbed from specific frames to be used for signaling purposes. The controller must be configured to use the same framing format as the line from the PBX or CO that connects to the voice port you are configuring.
Digital T1 lines use the SF or ESF framing format. SF provides two-state, continuous supervision signaling, in which a 0 bit value is used to represent on-hook and a 1 bit value is used to represent off-hook. ESF robs four bits instead of two, yet has little impact on voice quality. ESF is required for 64-kbps operation on DS0 and is recommended for Primary Rate Interface (PRI) configurations.
E1 lines can be configured for cyclic redundancy check (CRC4) or no cyclic redundancy check, with an optional argument for E1 lines in Australia.
To change the framing format, use the framing command in controller configuration mode. Refer to the Cisco IOS Voice Port Configuration Guide for configuration information.

Path Code Violations Increasing

Ensure that the framing format configured on the port matches the framing format of the line. Path code violations and line code violations are typically present simultaneously. Always verify that your line coding is correct. Look for the following in the show controller command output:

For E1 lines, look for “Framing is {crc4|no-crc4}”
For T1 lines, check the line coding, as described in the Line Coding on Digital T1/E1 Voice Ports. For T1 lines, path code violation error event is a frame synchronization bit error in the D4 (SF) format, or a CRC error in the ESF format.

If path code violations keep increasing, contact your service provider to check the line, because path code violations can also be caused by physical line problems.

Clock Sources on Digital T1/E1 Voice Ports

Digital T1/E1 interfaces use clock timers to ensure that voice packets are delivered and assembled properly. All interfaces handling the same packets must be configured to use the same source of timing so that packets are not lost or delivered late. The timing source that is configured can be external (from the line) or internal to the router’s digital interface.
If the timing source is internal, timing derives from the onboard phase-lock loop (PLL) chip in the digital voice interface. If the timing source is line (external), timing derives from the PBX or PSTN CO to which the voice port is connected. It is generally preferable to derive timing from the PSTN because PSTN clocks are maintained at an extremely accurate level. This is the default setting for the clock source. When two or more controllers are configured, one should be designated as the primary clock source; it drives the other controllers.
To change the clock source, use the clock source command in controller configuration mode. Refer to the Cisco IOS Voice Port Configuration Guide for configuration information.

Slip Seconds Error Counter Increasing

Use the show controller command to see if there are alarms or errors displayed by the controller. To see if the framing, line coding, and slip seconds error counters are increasing, use the show controller e1 command repeatedly. Note the values of the counters for the current interval.
If slips are present on the line, there is a clocking problem. The customer premises equipment (CPE) needs to synchronize to the clocking from the T1/E1 service provider. Complete the following steps to correct this problem:

SUMMARY STEPS

enable
show controller
configure terminal
controller {t1 | e1}
clock source {line [primary | secondary} | internal}

DETAILED STEPS

1. Enter enable to enter privileged EXEC mode. Enter a password, if necessary.
2. Ensure that the clock source is derived from the network. In the show controller EXEC command output, look for "Clock Source is Line Primary."

Note:

If there are multiple lines into an access server, only one can be the primary source. The other lines derive the clock from the primary source. If there are multiple lines, ensure that the line designated as the primary clock source is configured correctly. You can also configure a second line to provide clocking in case the primary source goes down. To do this, use the clock source line secondary command from controller configuration mode.

3. Enter configure terminal to enter global configuration mode.
4. Enter controller t1 or controller e1 to enter controller configuration mode.
5. Set both the primary and secondary clock sources from controller configuration mode. For example:

 Router(config-controlle)#clock source line primary

and

 Router(config-controlle)#clock source line secondary 1

Ensure that the lines that you specify as the primary and secondary are both active and stable.

Note:

On Cisco universal gateways and access servers, the clock source is specified using the dial-tdm-clock command. Refer to the "Managing Dial Shelves" chapter in the Cisco IOS Interface Configuration Guide.

Framing Loss Seconds Increasing

Follow these instructions when dealing with a framing loss seconds increase:

SUMMARY STEPS

enable
show controller
configure terminal
controller {t1 | e1}
framing {sf | esf} or framing {crc4|no-crc4}
cablelength {long|short}

DETAILED STEPS

1. Enter enable to enter privileged EXEC mode. Enter a password, if necessary.
2. Ensure that the framing format configured on the port matches the framing format of the line. Look for the following in the show controller output

For T1 lines, look for "Framing is {ESF|SF}"
For E1 lines, look for "Framing is {crc4|no-crc4}"

3. Enter configure terminal to enter global configuration mode.
4. Enter controller t1 or controller e1 to enter controller configuration mode.
5. To change the framing format, use the following commands in controller configuration mode:

For T1 lines, use framing {sf | esf}. For example:

 Router(config-controlle)#framing esf

For E1 lines, use framing {crc4|no-crc4}. For example:

 Router>(config-controller)#framing crc4

6. For T1 lines, change the line build-out using the cablelength long or cablelength short command.
Contact your service provider and consult the Voice Port Configuration Guide for details on settings.

Line Coding on Digital T1/E1 Voice Ports

Digital T1/E1 interfaces require that line encoding be configured to match that of the PBX or CO that is being connected to the voice port. Line encoding defines the type of framing used on the line.
T1 line encoding methods include alternate mark inversion (AMI) and binary 8 zero substitution (B8ZS). AMI is used on older T1 circuits and references signal transitions with a binary 1, or "mark." B8ZS, a more reliable method, is more popular and is recommended for PRI configurations as well. B8ZS encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations.
Supported E1 line encoding methods are AMI and high-density bipolar 3 (HDB3), which is a form of zero-suppression line coding.
Use the show controller command to see if there are alarms or errors displayed by the controller. To see if the framing, line coding, and slip seconds error counters are increasing, use the show controller command repeatedly. Note the values of the counters for the current interval.

Line Code Violations Increasing

Ensure that the line coding configured on the port matches the line coding of the line. Change the line code in controller configuration mode if necessary.

For E1 lines, look for "Line Code is HDB3" in the show controller e1 output.
For T1 lines, look for "Line Code is {B8ZS|AMI}" in the show controller t1 output.

For T1 lines, change the line build-out using the cablelength long or cablelength short command.
If line code violations keep increasing, contact your service provider to check the line, because line code violations can also be caused by physical line problems.

Path Code Violations Increasing

Ensure the framing format configured on the port matches the framing format of the line. Path code violations and line code violations are typically present simultaneously. Always verify that your line coding is correct. Look for the following in the show controller output:

For T1 lines, look for "Line Code is {B8ZS|AMI}"

For T1 lines, path code violation error event is a frame synchronization bit error in the D4 (SF) format, or a CRC error in the ESF format.

For E1 lines, check the framing as described in the Framing Formats on Digital T1/E1 Voice Ports.

For T1 lines, change the line build-out using the cablelength long or cablelength short command.
If path code violations keep increasing, contact your service provider to check the line. Path code violations can also be caused by physical line problems.

T1/E1 Channel-Associated Signaling

CAS exists in many networks today. CAS systems carry signaling information in the same channels in which voice and data are carried. Current telecommunication networks require more efficient means of signaling. CAS exists in many varieties that operate over analog and digital facilities. The analog facilities are either two- or four-wire, and the digital facilities are either North American T1 or European E1. Each CAS system uses either supervision signaling or address signaling over analog and digital facilities.
Three groups of signals are present in these facilities:

Supervision signals represent events occurring on a trunk and can be specific to CAS. Signal types include seizure, wink, and answer.
Address signals represent the digits dialed or called party number and, in some instances, other information. Address signals are based on multiflex signaling.
Tone and announcement signals include ringing and busy tones and announcements specific to an event. Service circuits are used in most exchanges to send and receive address signals and tones as well as to play announcements.

This section describes CAS signaling, which is sometimes called robbed-bit signaling because user bandwidth is robbed by the network for signaling. A bit is taken from every sixth frame of voice data to communicate on- or off-hook status, wink, ground start, dialed digits, and other information about the call.
In addition to setting up and tearing down calls, CAS provides for the receipt and capture of dialed number identification (DNIS) and automatic number identification (ANI) information, which are used to support authentication and other functions. The main disadvantage of CAS signaling is its use of user bandwidth to perform these signaling functions.
For more information about troubleshooting CAS, refer to Configuring and Troubleshooting T1 CAS Signaling, document ID 24642.
If your CAS-configured router gets stuck in the EM_PARK state, refer to Troubleshooting EM_PARK Issues for E&M Digital CAS Signaling, document ID 18959.

Troubleshooting Commands

Certain show commands are supported by the Output Interpreter tool, which allows you to view an analysis of show command output.

debug voip ccapi inout-Traces the execution path through the call control API, which serves as the interface between the call session application and the underlying network-specific software. You can use the output from this command to understand how calls are being handled by the router.
debug vpm all-Enables all of the debug vpm commands: debug vpm spi, debug vpm signal, and debug vpm dsp.

Note:

This debug command generates a large amount of output.

show call active voice-Displays the contents of the active call table, which shows all of the calls currently connected through the router.
show call history voice-Displays the call history table. The call history table contains a listing of all calls connected through this router in descending time order since VoIP was enabled. You can display subsets of the call history table by using specific keywords.
show voice port-Displays configuration information about a specific voice port.
debug vtsp all-Enables the following debug vtsp commands: debug vtsp session, debug vtsp error, and debug vtsp dsp.

E1 R2 Interfaces

R2 signaling is a CAS system developed in the 1960s and still in use today in Europe, Latin America, Australia, and Asia. R2 signaling exists in several country versions or variants in an international version called Consultative Committee for International Telegraph and Telephone-R2 (CCITT-R2). The R2 signaling specifications are contained in International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Recommendations Q.400 through Q.490. E1 R2 signaling is an international signaling standard that is common to channelized E1 networks.
E1 R2 signaling support allows the Cisco gateways to communicate with a central office (CO) or PBX trunk and act as a tie-line replacement. Although R2 signaling has been defined in ITU-T Q.400-Q.490 recommendations, R2 is implemented in many different ways. (Various countries implement R2 differently.) Cisco's implementation of R2 signaling on routers can accommodate most of the variations.

Troubleshooting E1 R2 Failures

Follow the instructions below to troubleshoot your configuration.

SUMMARY STEPS

show controller e1
show vfc slot number interface
Configure DID on the POTS peer.
cptone
Match line and register signaling provisions to the switch configuration.
Turn on appropriate debugs.
Check for communication between the router and PBX or switch.

DETAILED STEPS

1. Verify that controller E1 0 is up with the show controller e1 0 command.
If it is down, check framing, line coding, clock source, and alarms. Replace the cable and reseat the card. To run these tests, see the following sections:

2. If you are using an AS5300, check that the DSPs are correctly installed with the show vfc slot number interface command.
3. Configure direct inward dial (DID) on the plain old telephone service (POTS) peer, so that the received digits are used to choose an outgoing peer.
4. Specify cptone'(cptone is specific for your country) on the voice-ports.
A cptone country must be configured to match cas-custom country. The cptone parameter sets the call progress tones for a particular country, and more importantly sets the encoding to a-law or u-law, depending on the country. For u-law, use the us keyword. For a-law, use the gb keyword. To configure cptone, see the Voice Port Configuration Guide.
5. Match line and register signaling provisions to the switch configuration.
6. Turn on some of the debugs shown in the following section and study the outputs.
7. Check for communication between the router and PBX or switch:

Is the line seized?
Does the router receive/send digits?
Find out which side is clearing the call.

If possible, use the latest Cisco IOS software release.

debug and show Commands

Certain show commands are supported by the Output Interpreter Tool, which allows you to view an analysis of show command output.
For Cisco IOS Software Release 12.0 and newer, use the following debugs:

debug cas-For line signaling
debug csm voice-For interregister signaling
debug vtsp all -Exchanges output of all messages (digits) between the PBX and the router

For Cisco IOS Software Release IOS 11.3, use the following commands:

modem-mgmt csm debug-rbs -For line signaling (specify service internal in global configuration mode.)
debug csm voice - For interregister signaling
debug vtsp all -Exchanges output of all messages (digits) between the PBX and the router

For the AS5400 and AS5350 platforms, use the following debugs:

debug sigsm r2-For interregister signaling
debug vtsp all -Exchanges output of all messages (digits) between the PBX and the router

ISDN Interfaces

An ISDN network can consist of T1, T3, E1, and E3 and has two types of subscriber access: Basic Rate Interface (BRI) and Primary Rate Interface (PRI). Each access comprises B and D channels.
ISDN BRI provides two B channels, each capable of transferring voice or data at 64 kbps, and one 16-kbps D channel that carries signaling traffic. The D channel is used by the telephone network to carry instructions about how to handle each of the B channels. ISDN BRI (also referred to as "2 B + D") provides a maximum transmission speed of 128 kbps.
ISDN PRI provides 23 B channels plus a D channel (in North America and Japan) or 30 B channels plus a D channel (in the rest of the world). Similar to the ISDN BRI D channel, the ISDN PRI D channel carries signaling traffic. ISDN PRI is often referred to as "23 B + D" (in North America and Japan) or "30 B + D" (in the rest of the world). The D channel notifies the central office switch to send the incoming call to particular time slots on the Cisco access server or router. Each one of the B channels carries data or voice. The D channel carries signaling for the B channels. The D channel indicates whether the call is a circuit-switched digital call or an analog modem call. Analog modem calls are decoded and then sent to the onboard modems. Circuit-switched digital calls are relayed directly to the ISDN processor in the router.
The ISDN interfaces for Cisco gateways enable Cisco IOS software to replicate the PSTN interface to a PBX that is compatible with European Telecommunications Standards Institute (ETSI) NET3 and QSIG switch types.
The application shown in Figure: Typical Application Using ISDN BRI NT/TE VICs or ISDN BVMs allows enterprise customers with a large installed base of legacy telephony equipment to bypass the PSTN.

Figure: Typical Application Using ISDN BRI NT/TE VICs or ISDN BVMs

Topics for ISDN voice troubleshooting are as follows:

ISDN PRI Troubleshooting Tips

If you are having trouble connecting a call and you suspect that the problem is associated with voice port configuration, you can try to resolve the problem by performing the following tasks:

Ping the associated IP address to confirm connectivity.
Determine if the voice feature card (VFC) has been correctly installed. For more information, refer to Installing Voice-over-IP Feature Cards in Cisco AS5300 Universal Access Servers.
To learn if the VFC is operational, use the show vfc slot_number command.
To view layer status information, use the show isdn status command. If you receive a status message stating that Layer 1 is deactivated, make sure the cable connection is not loose or disconnected.
With T1 lines, determine if your a-law setting is correct. With E1 lines, determine if your u-law setting is correct. To configure both a-law and u-law values, use the cptone command. For more information about the cptone command, refer to the Cisco IOS Voice Command Reference.
If dialing cannot occur, use the debug isdn q931 command to check the ISDN configuration.

For more information about troubleshooting T1 PRI, refer to T1 PRI Troubleshooting, document ID 9344.

Verifying the ISDN Switch Type and PRI Group Timeslot Configuration

Use the show running-config command to ensure that isdn switch-type and pri-group timeslots are configured correctly. To specify the central office switch type on the ISDN interface, use the isdn switch-type global configuration command. Options for this command include primary-net5. Contact your service provider for the correct values to use.

Note:

If you have defined ISDN PRI groups and channel groups on the same controller, ensure that you do not overlap time slots or use the ISDN D-channel timeslot in a channel group. When configuring a Primary Rate Interface (PRI), use the isdn switch-type global configuration command to configure the switch type.

To configure the isdn switch-type and pri-group:

 Router# configure terminal  
 Router(config)# isdn switch-type primary-net5
 Router(config)# controller e1 0
 Router(config-controlle)# pri-group timeslots 1-31

Note:

In some countries, service providers offer fractional PRI lines. This means that fewer than 30 B-channels may be used for ISDN connections. For fractional PRI lines, the time slot range must include the operational B channels, plus the D channel (this is fixed on time slot 16).

For example:

pri-group timeslots 1-10, 16 for the first ten B-channels.
timeslots 1-21 for the first 20 B-channels.

Verifying the Signaling Channel

If the error counters do not increase, but the problem persists, complete the following steps to verify that the signaling channel is up and configured correctly

SUMMARY STEPS

show interfaces serial number :15
Ensure that the interface is up.
Ensure that encapsulation is PPP.
Ensure that the interface is not in loopback mode.
Power cycle the router.
Turn on appropriate debugs.
Contact TAC.

DETAILED STEPS

1. Run the show interfaces serial number:15 command, where the number is the interface number.
2. Ensure that the interface is up. If the interface is not up, use the no shutdown command to bring the interface up. For example:

 Router# config terminal  
 Enter configuration commands, one per line. End with CNTL/Z.  
 Router(config)# interface serial 0:15
 Router(config-if)# no shutdown

3. Ensure that encapsulation is PPP. If not, use the encapsulation ppp command to set encapsulation. For example:

 Router(config-if)# encapsulation ppp

4. Ensure that the interface is not in loopback mode. Loopback should be set only for testing purposes. Use the no loopback command to remove loopbacks. For example:

 Router(config-if)# no loopback

5. Power cycle the router.
6. Turn on appropriate debugs.
7. If the problem persists, contact your service provider or the Cisco Technical Assistance Center (TAC).

Ringback

In some situations, a gateway might intermittently fail to provide ringback tone (RBT) to an incoming ISDN caller. This problem has been seen on local, long-distance, and international calls.
The gateway generates ringback towards the network side (PSTN or PBX) if the setup contains Progress IE = 3, meaning the originating address (calling party) is NON-ISDN.
The gateway does not generate a ringback towards the network side (PSTN or PBX) if the setup contains no Progress IE (Progress IE =0), meaning the originating address (calling party) is ISDN.
The following figure is an example of when this might occur. International calls are arriving on ISDN.
PSTN (ISDN) --------- Cisco IOS gateway ------------ Cisco CallManager ------------ IP phone

Example of a Call That Gets Ringback Tone

You receive a call from a non-ISDN terminal. The setup contains a Progress IE = 3. The gateway generates ringback when it receives an alert from Cisco CallManager.
The following debugs were captured with the Cisco IOS command debug isdn q931:

 01:34:48: ISDN Se0:15: RX <- SETUP pd = 8 callref = 0x002B  
 01:34:48: Sending Complete  
 01:34:48: Bearer Capability i = 0x9090A3  
 01:34:48: Channel ID i = 0xA9838D  
 01:34:48: Progress Ind i = 0x8583 - Origination address is non-ISDN  
 01:34:48: Calling Party Number i = 0x2183, '27045000', Plan:ISDN, Type:National  
 01:34:48: Called Party Number i = 0xA1, '27182145', Plan:ISDN, Type:National  
 01:34:48: ISDN Se0:15: TX -> CALL_PROC pd = 8 callref = 0x802B  
 01:34:48: Channel ID i = 0xA9838D  
 01:34:48: act_alert: Tone Ring Back generated in direction Network One  
 01:34:48: act_gen_tone: Tone Ring Back generated in direction Network  
 01:34:48: ISDN Se0:15: TX -> ALERTING pd = 8 callref = 0x802B

Example of a Call That Does Not Get Ringback Tone

You receive a call from an ISDN terminal. There is no Progress IE in the setup. (Progress IE = 0). The gateway is not generating ringback when it receives the alert from Cisco CallManager.

 01:37:01: ISDN Se0:15: RX <- SETUP pd = 8 callref = 0x002E  
 01:37:01: Sending Complete  
 01:37:01: Bearer Capability i = 0x8090A3  
 01:37:01: Channel ID i = 0xA98391  
 01:37:01: Calling Party Number i = 0x2183, '478681058', Plan:ISDN, Type:International  
 01:37:01: Called Party Number i = 0xA1, '27182145', Plan:ISDN, Type:International  
 01:37:01: High Layer Compat i = 0x9181  
 01:37:01: ISDN Se0:15: TX -> CALL_PROC pd = 8 callref = 0x802E  
 01:37:01: Channel ID i = 0xA98391  
 01:37:01: ISDN Se0:15: TX -> ALERTING pd = 8 callref = 0x802E

In the case above, the gateway is expecting the ISDN to generate the ringback (due to no PI of 3). The ISDN, however, is not generating a ringback tone. This results in the caller hearing only silence until the called party answers the call. This might be due to an ISDN interworking issue caused because the call originated internationally (normally ring is generated at the terminating device for international calls).

Forcing Ringback

You can force the gateway to generate a ringback with the progress_ind setup enable 3 command. Configure the forced ringback on the VoIP dial peer that points to Cisco CallManager.

 !  
 dial-peer voice 500 voip  
 destination-pattern 5...  
 progress_ind setup enable 3  
 !--forces ring back tone for this peer  
 session target ipv4:10.200.73.15  
 codec g711ulaw  
 !

For more information, refer to PSTN Callers not Hearing any Ring Back When they Call IP Phones, document ID 8331.

QSIG Protocol Support

QSIG is a peer-to-peer signaling system used in corporate voice networking. Internationally, QSIG is known as Private Signaling System No. 1 (PSS1). This open standard is a variant of ISDN D-channel voice signaling that is based on the ISDN Q.921 and Q.931 standards. Therefore, as well as providing inter-PBX communications, QSIG is compatible with public and private ISDN systems.
QSIG also has one important mechanism known as Generic Functional Procedures (QSIG GF). This mechanism provides a standard method for transporting features transparently across a network.
Integration of QSIG protocol support with Cisco voice switching services allows Cisco devices to connect PBXs, key systems, and central office switches (COs) that communicate by using the QSIG protocol. With QSIG, Cisco networks emulate the functionality of the PSTN, and QSIG signaling messages allow the dynamic establishment of voice connections across a Cisco WAN to a peer router, which can then transport the signaling and voice packets to a second PBX, as shown in Figure: QSIG Signaling.

Figure: QSIG Signaling

The Cisco voice packet network appears to the traditional QSIG PBXs as a distributed transit PBX that can establish calls to any PBX, non-QSIG PBX, or other telephony endpoint served by a Cisco gateway, including non-QSIG endpoints.

QSIG Support Troubleshooting Tips

Table: QSIG Troubleshooting Commands lists debug and show commands that can help you analyze problems with your QSIG configuration.

Table: QSIG Troubleshooting Commands

!Command	!Purpose
show isdn status	Displays the status of all ISDN interfaces, including active layers, timer information, and switch type settings.
show controller t1/e1	Displays information about T1 and E1 controllers.
show voice port summary	Displays summary information about voice port configuration.
show dial-peer voice	Displays how voice dial peers are configured.
show cdapi	Displays the Call Distributor Application Programming Interface (CDAPI) information.
show call history voice record	Displays information about calls made to and from the router.
show rawmsg	Displays information about any memory leaks.
debug isdn event	Displays events occurring on the user side (on the router) of the ISDN interface. The ISDN events that can be displayed are Q.931 events (call setup and teardown of ISDN network connections).
debug tsp	Displays information about the telephony service provider (TSP).
debug cdapi {events \| detail}	Displays information about CDAPI application events, registration, messages, and so on.

Troubleshooting Drop-and-Insert

When a router is configured for drop-and-insert, also called TDM cross connect, traffic is passed as a transparent bit stream between the configured ports. The router acts as a conduit between the ports, ensuring that the bit stream and clocking are preserved. Because of this, there are no commands to monitor traffic or debug signaling bits. You can confirm the physical status of the T1 interfaces (carrier loss) and line quality (line errors, clock slips, framing errors) using the show controller t1 slot/port command.
You can connect the PBX directly to the voice mail system to isolate signaling problems. If the system is still not working when the router has been bypassed, you might need to use T1 analyzers (for example, the Acterna Tberd T1 analyzer) to verify that the PBX or voice mail system is sending the correct information on the T1 trunk. You can also use the analyzer to verify that the drop-and-insert feature is working correctly from one port to the other.
For more information on troubleshooting drop-and-insert, refer to Integrating PBXs into VoIP Networks Using the TDM Cross Connect Feature, document ID 27789.

Troubleshooting Transparent Common Channel Signaling

Transparent Common Channel Signaling (T-CCS) allows the connection of two PBXs with digital interfaces that use a proprietary or unsupported CCS protocol without the need for interpretation of CCS signaling for call processing.
With T-CCS, the PBX voice channels can be "nailed-up" and compressed between sites, and the accompanying signaling channel(s) can be transparently tunneled across the IP/FR/ATM backbone between PBXs. Thus, calls from the PBXs are not routed by Cisco on a call-by-call basis but follow a preconfigured route to the destination.
For information about troubleshooting T-CCS, refer to Configuring and Troubleshooting Transparent CCS, document ID 19087.

Dial Tone Issues

This section discusses troubleshooting of the voice network when no dial tone is heard from a voice port that is in the off-hook condition. Some issues include:

A common problem in the voice network involves no dial tone being heard from a voice port in the off-hook condition. This might be related to configuration issues, a hardware problem, a DSP problem, or a bug in Cisco IOS software. A voice port configured with connection trunk does not provide dial tone. A faulty network module or Foreign Exchange Station (FXS) card might cause silence or no dial tone in a voice port.
For more information, refer to Troubleshooting No Dial Tone Issues, document ID 22372.
The following sections describe these various problems and related solutions.

Router Does Not Recognize Voice Port

When a router does not recognize a voice port, it might be because the router is not loaded with the Cisco IOS image required for voice support. In the case of a Cisco 1750 router, make sure that it does not have PVDM-256K-4 and PVDM-256K-8 DSPs. These are packet voice/data modules (PVDMs) for Cisco routers 1751 and later. If the Cisco 1750 router does not have the correct PVDM, the voice ports might show up in show version and show diag command output, however, there is no dial tone. Also, no DSPs are seen in the show voice dsp command output. The Cisco 1750 router should carry the proper PVDM-4 and PVDM-8 DSP cards.
For Cisco modular access routers , another problem could be a bad network module. If there is an alarm light on the network module, remove the module, put it back in the slot and power cycle. If the alarm light is still lit, replace the network module. Also, try plugging an analog phone into the FXS port with a good cable. If there is no dial tone, replace the FXS card.

Note:

FXS-Direct Inward Dialing (DID) does not provide dial tone.

Voice Ports Are in the Shutdown State

When voice ports are in the shutdown state, they do not provide dial tone. To fix this, enable the voice port with the no shut command under the voice port.

Voice Ports Configured as Connection Trunk

If voice ports are configured as connection trunk, the ports are in the S_TRUNKED state. Voice ports do not provide dial tone if configured for connection private line auto ringdown (PLAR). Use the show voice call summary command to verify the VPM state. In the example below, the dial-tone is given by the remote PBX/PSTN.

 Router# show voice call summary
 PORT CODEC VAD VTSP STATE VPM STATE  
 ==== ===== === ========== ==========  
 1/0/0 g729ar8 y S_CONNECT S_TRUNKED  
 1/0/1 g729ar8 y S_CONNECT S_TRUNKED

Remove the connection trunk/PLAR configuration to ensure that you are getting the dial-tone.

No Dial Tone on Digital Voice Port

Use the show dial-peer command to see if the dial-peer ports are configured with the direct-inward-dial command. This command disables dial tone from the voice port. For example:

 dial-peer voice 1 pots  
 destination-pattern .T  
 direct-inward-dial  
 port 0:D

Removing the direct-inward-dial command from dial-peer pots causes the digital voice port to provide dial tone.

Debug Command Output Shows VTSP Timeout

The VTSP and DSP timeouts are known issues that appear in many forms. Use the test dsps slot# command to see if processors are alive. Cisco IOS Releases 12.2.6a [12.2.6cM1] and later include fixes for many of these issues, but possibly not all. The problem can be temporarily cleared by a power cycle. If you experience this problem, you should open a case with the Cisco Technical Assistance Center (TAC).

What’s new with SRST 9.x

2012-10-17T15:41:00.002+05:30

Best Practices for SRST Router
Use a Cisco Unified SRST router, rather than Unified CME in SRST mode, for the following deployment scenarios:
•For supporting a maximum of 1,500 phones on a single SRST router.
•For up to 3,000 phones, use two SRST routers. Dial plans must be properly configured to route the calls back and forth between the SRST routers.
•For simple, one-time configuration of basic SRST functions.
•For SRTP media encryption, which is available only in Cisco Unified SRST (Secure SRST).
•For support of the Cisco VG248 Voice Gateway.
For routing calls to and from phones that are unreachable or not registered to the SRST router, use the alias command.
Enhanced Survivable Remote Site Telephony
Enhanced Survivable Remote Site Telephony (E-SRST) simplifies the deployment of Cisco Unified CME running SRST in the branch. E-SRST architecture is built on top of Survivable Remote Site Voicemail (SRSV). During normal operation, E-SRST utilizes Cisco Unified Messaging Gateway E-SRST located at headquarters site to regularly retrieve configurations (for example, calling search space, partition, hunt group, call park, call pickup, and so forth, if configured) from Cisco Unified CM and upload them to provision the branch router with similar functionality for use in SRST mode. Thus, E-SRST reduces manual configuration required in Unified CME running SRST and enables users to have a similar calling experience in both SRST and normal modes.
What’s new with SRST 9.x
· Support for Cisco Unified IP Phone 6901, 6911, 6921,6941, 6945 & 6961 SIP IP Phones
· Forced Authorization Code (FAC)
· Normalized +E.164 support for SRST
· Improved deployment flexibility with support for SSL VPN client on Cisco Unified IP Phones (SCCP)
· Customizable Programmable Line Keys, Button Layout Control
· ISDN overlap sending on PRI/BRI
E-SRST Overview
· The Survivable Provisioning Infrastructure
o Automatically provisions remote branch sites
o Always in-sync with the Centralized Communications Manager and pushes the updates to the Branch routers
o Automatically synchronizes moves, adds, and changes across from CUCM to Branch ISR
· Phone User Experience
o Device display and phone user operations are identical in survivable mode
· Call Treatment:
o Access and security control
o Digit manipulation for both incoming and outgoing calls
· Management:
o GUI interface for provisioning, monitoring, report and troubleshooting.
o On Demand info sync with CUCM

https://supportforums.cisco.com/videos/3031
http://www.cisco.com/go/srst
https://www.cisco.com/web/learning/le31/le46/cln/qlm/CCVP/cipt1/cucme_as_srst/player.html
Sample Configuration
SRST Sample Configuration .txt
Cisco Secure SRST Configuration Example
Unified SRST Interactive Voice Network Configuration Example
SRST Voice Mail Interactive Voice Network Configuration Example
Common Show & Debug Commands
Show Commands

Command	Purpose
show run	Displays the configuration.
show call-manager-fallback all	Displays the detailed configuration of all the Cisco IP Phones, voice ports, and dial peers.
show call-manager-fallback dial-peer	Displays the output of the dial peers of the SRST Catalyst 4224.
show call-manager-fallback ephone-dn	Displays the Cisco IP Phone destination number.
show call-manager-fallback voice-port	Displays output for the voice ports.
show ephone	Displays Cisco IP Phone output.
show ephone-dn	Displays the Cisco IP Phone destination number.
show ephone summary	Displays a summary of the Cisco IP Phone output.
show voice port summary	Displays a summary of all voice ports.
show dial-peer voice summary	Displays a summary of all voice dial peers.

Debug Commands
Debug ephone register
Debug ephone state
Debug ephone details
Management (SNMP)
Cisco Unified SRST SNMP MIB Release 4.0 Guide
Cisco SRST SNMP MIB Release 3.4 Guide
For general SRST product related questions, please contact the Cisco Product team.
mailto:access-ccme-cue

Model	Mbps	CME Max User	SRST Max User
1841	2	0	0
1941	25	35	35
2801	2	25	24
2901	25	35	35
2811	4	35	36
2911	35	50	50
2821	8	50	48
2921	50	100	100
2851	12	100	96
2951	75	150	250
3825	22	175	336
3925	100	200	730
3845	44	250	720
3945	150	300	1200

Compare SRST
SRST vs CME in SRST
SRST vs E-SRST
E-SRST combines all the benefits of SRST, but also includes the following:
• Enhanced user experience in failover mode by maintaining phone displays and providing full call-control features
• Easy to use GUI interface to provision, monitor, report, and troubleshoot remote sites
• Automatic synchronization with Cisco Unified Communications Manager for additions, deletions, and modifications of users and phones
• Calling-rule restrictions continued in failover mode

CME/SRST Release Version	IOS Version
CME/SRST 4.1	12.4(15)T
CME/SRST 4.2	12.4(11)XW3
CME/SRST 7.0	12.4(20)T
CME/SRST 7.1	15.0(1)M2 (Golden Release, recommended for most customers
CME/SRST 8.0	15.1.1T
CME/SRST 8.1	15.1.2T
CME/SRST 8.5	15.1.3T

Features by Cisco Unified SRST Software Version
Cisco Unified SRST	Cisco IOS Release	Enhancements or Modifications
Version 9.0	15.2(2)T	•Support for Cisco Unified 6901 and 6911 SIP IP Phones•Support for Cisco Unified 6921, 6941, 6945, and 6961 SIP IP Phones•Support for Cisco Unified 8941 and 8945 SIP IP Phones •Multiple Calls Per Line •Voice and Fax Support on Cisco ATA-187
Version 8.8	15.2(1)T	Support for Cisco Unified 6945, 8941, and 8945 SCCP IP Phones
Version 8.0	15.1(1)T	Beginning with Cisco IP Phone firmware 8.5(3) and Cisco IOS Release 15.1(1)T, Cisco SRST supports SIP signaling over UDP, TCP, and TLS connections, providing both RTP and SRTP media connections based on the security settings of the IP phone. For more information, see the following sections:•Signaling Security on Unify SRST – TLS •Media Security on Unify SRST – SRTP •Configuring Secure SIP Call Signaling and SRTP Media with Cisco SRST
Version 7.0/4.3	•Configuring Eight Calls per Button (Octo-Line)•Configuring Consultative Transfer
Version 4.2(1)	Enhanced 911 ServicesThe following new features are included: •Assigning ERLs to zones to enable routing to the PSAP that is closest to the caller. •Customizing E911 by defining a default ELIN, identifying a designated number if the 911 caller cannot be reached on callback, specifying the expiry time for data in the Last Caller table, and enabling syslog messages that announce all emergency calls. •Expanding the E911 location information to include name and address. •Adding new permanent call detail records.

JTAPI / CTI Secure Configuring and Troubleshooting

2012-10-17T15:37:00.004+05:30

CUCM Config

Enable the Certificate Authority Proxy Function (CAPF), Certificate Trust List Provider (CTL), and CTIManager services.
Run the CTL client with USB eTokens on a workstation to connect to the cluster, generate a CTL file, and move the cluster to Mixed Mode.
On the Device > Phone page check “Allow Control of Device from CTI” for the phone you want to monitor / control.
Associate your End User or Application User with the phone.
Add the user to the groups CTI Enabled, CTI Secure Connection, and CTI Allow Reception of SRTP Key Material (Note: Adding the user to the Secure CTI and SRTP Key Material groups means that this JTAPI user will ONLY be allowed to connect on the secure port of 2749 using certs. To connected on a non secure port later, remove the user from these secure groups)
Under Users > End / Application User CAPF Profile, select the certificate operation of Install / Upgrade for the JTAPI user.
After restarting the CUCM server you will now have CCM listening on TCP port 2443 for secure SCCP connections and CTIManager listening on 2749 for secure JTAPI/QBE connections. The TFTP process will be providing a CTL file. The CAPF service on the publisher will be providing certificates to any client that securely connects on TCP 3804 and has a certificate operation of Install / Upgrade.

JTAPI Client Config

Install the Cisco JTAPI Client on the server or client PC where JTAPI will be running. Make sure to download JTAPI for either Windows or Linux.
Under either the Start Menu, or the JTAPITools directory, find the JTAPI Preferences tool.
If you’re trying to set this up for the first time I recommend turning on ALL the tracing
Add your CUCM CTIManager servers under Cisco Unified CM.
Under Security set the following settings and click “Update Certificate”. The Authentication String and Instance ID must match the User’s CAPF Profile.
Now we’re ready to verify the secure connection with the JTrace tool below.

Troubleshooting

JTrace

Now that we have the JTAPI integration configured we want to test that it actually works! We can do this by taking control of a single device with a Cisco provided JTAPI application. If you’re ever unsure of whether the problem is with JTAPI, or a customer JTAPI application, you can use JTrace to verify.

In our JTAPITools directory (typically C:\Program Files\JTAPITools) look for the jtrace folder, and the readme.htm for full instructions.

JTrace takes arguments in the following form:

java jtrace.JTrace <CTI Server IP> <JTAPI user> <password> <directory number>

Note
JTrace does not support Instance IDs. If you have one user with multiple Instance IDs (implying mutltiple certificates) do not use JTrace, or it may invalidate your existing certificates by marking them as comprimised since a connection may be made with the wrong certificate to the wrong instance.

You launch JTrace from a cmd window in the JTAPITools directory

When the JTAPI connection to CTIManager is successful and the line and device are opened, two windows pop up.

The first window logs information about the JTAPI Provider that is opened. It will list all of the lines and devices associated with the JTAPI user selected.

The second window will show the actual device that is opened, and any events that JTAPI receives for that device. Here we can see the opened phone (8012001) is receiving a call from 8011111.

The JTrace tool can help identify whether or not a JTAPI connection to the CTIManager server can be established and events received. The error messages displayed in the two new JTrace windows or the cmd window can also help diagnose any issues.

As a sanity check, leave security disabled in the JTAPI Client JTPrefs and run JTrace. Once a non secure connection can be established then enable security and test again. (Note: This requires adding and removing the user from the two secure CTI Groups in CCMAdmin)

For more advanced troubleshooting turn to the logs.

JTAPI Logs

The logs can be found under the JTAPITools installation directory (by default C:\Program Files\JTAPITools) They follow the format of CiscoJTAPIXX.log.

Look for the following sequence of messages to verify the JTAPI client connected to the CTIManager server and logged in successfully.

Opening server "14.48.44.80" login "jasburns" Trying to create TLS/SSL socket connection to 14.48.44.80 JTAPITLSHandshakeCompletedListener:handshakeCompleted : javax.net.ssl.HandshakeCompletedEvent ]] sending: com.cisco.cti.protocol.ProviderOpenRequest received Response: com.cisco.cti.protocol.ProviderOpenResponse queuing com.cisco.cti.protocol.ProviderOpenCompletedEvent

If at any point in the logs an error appears that is a large negative number or a large hex number starting with 0x8CC you can look it up with this magic CTI Error Decoder Ring.

One of the most common errors is
DIRECTORY_LOGIN_FAILED 0x8CCC0060

This error may appear either in the above hex, or if in decimal it can be converted using calc. The decoded text points to a problem with login credentials or the login process.

JTAPI Security Logs

Security logs will show important details regarding the CTL Download and the encapsulated CAPF Certificate. They follow the format of CiscoJtapiSecurityXX.log

CTIManager Logs

Pending

Packet Captures

Wireshark is a useful capture tool to analyze the steps in the secure JTAPI configuration and connection process. Almost everything about the secure JTAPI process can be learned by taking a packet capture on the PC or server where the JTAPI Client is installed.

The first important step happens when clicking “Update Certificate” in the JTAPI Configuration window. The JTAPI Client downloads a CTL file via TFTP (UDP port 69) from the configured TFTP server. This CTL file contains the CAPF Server Certificate

After downloading the CTL File and storing it in the keystore folder, the JTAPI Client has to make a TLS connection to the CAPF Server on TCP 3804. JTAPI compares the cert the CAPF server presents to the one embedded in the CTL file to make sure the connection is trusted. Filter on tcp.port == 3804. It will be necessary to right click any packet in this stream and select “Decode As” > “SSL” for both the source and destination ports so Wireshark knows to decode a stream on these ports as an SSL / TLS session.

If for some reason the CAPF Certificate in the CTL file doesn’t match the CAPF certicicate presented in the TLS Handshake we need to walk through the following steps:

Verify the Subject CN and expiration date of the CAPF cert installed in the OS Admin of the publisher server.
Run the CTL Client to insert the latest CAPF certificate into the CTL file.
Restart the TFTP service.

After the JTAPI Client makes a secure connection to the CAPF service, we walk through the operation set in the CAPF Profile. In the case above that means “Install / Upgrade”. A new Keystore will be created that contains the key and certificate obtained via CAPF in the default keystore location.

Now that the JTAPI Client has a CTL File and certificate we’re ready to connect to the secure CTIManager. Fire up JTrace as mentioned earlier and filter on tcp.port == 2749. Don’t forget to decode as SSL.

Cisco Unity Connection 9.x

2012-10-17T15:33:00.002+05:30

Cisco Unity Connection Version 9.x

Cisco CUCM 9.x SRND Download

2012-10-17T15:31:00.002+05:30

Cisco CUCM 9.x SRND download

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/9x/uc9x.html

Delete Multiple voicemails cisco Unity 7.0

2011-07-24T01:14:00.001+05:30

http://www.ciscounitytools.com/Applications/Unity/MSM/MSM.html

ip prefix-list

2011-06-05T03:18:00.001+05:30

ip prefix-list provides the most powerful prefix based filtering mechanism

Here is a quick little tutorial on Prefix-lists for you.

A normal access-list CANNOT check the subnet mask of a network. It can only check bits to make sure they match, nothing more. A prefix-list has an advantage over an access-list in that it CAN check BOTH bits and subnet mask - both would have to match for the network to be either permitted or denied.

For checking bits a prefix list ALWAYS goes from left to right and CANNOT skip any bits. A basic example would be this:

172.16.8.0/24

If there is only a / after the network (no le or ge) then the number after the / is BOTH bits checked and subnet mask. So in this case it will check the 24 bits from left to right (won't care about the last 8
bits) AND it will make sure that it has a 24 bit mask. BOTH the 24 bits checked and the 24 bit subnet mask must match for the network to be permitted or denied.

No we can do a range of subnet masks also that could be permitted or
denyed:

172.16.8.0/24 ge 25

If we use either the le or ge (or both le and ge) after the /, then the number directly after the / becomes ONLY bits checked and the number after the ge or le (or both) is the subnet mask. So in this case we are still going to check the first 24 bits of the network from left to right. If those match we are then going to check the subnet mask, which in this case can be GREATER THAN OR EQUAL TO 25 bits - meaning that as long as the first 24 bits of the network match the subnet mask could be 25,26,27,28,29,30,31,or 32 bits. They would all match.

We can also do:

172.16.8.0/24 le 28

Again this will check the first 24 bits of the network to make sure that they match. Then it will check to make sure that the subnet mask is LESS THAN OR EQUAL TO 28 bits. Now this isn't going to be 28 bits down to 0 bits, the subnet mask can't be any lower than the bits we are checking. So the valid range of subnet masks for this one would be 28 bits down to 24 bits (24,25,26,27,and 28). All of those would match.

We can also do both ge and le:

172.16.8.0/24 ge 25 le 27

Here again we are checking the first 24 bits to make sure they match.
Then our subnet mask must be GREATER THAN OR EQUAL TO 25 bits LESS THAN OR EQUAL TO 27 bits. Meaning that 25,26,and 27 bit subnet masks would match.

Now for a couple of examples:

If we have the following networks:

172.16.8.0/28
172.16.8.16/28
172.16.8.32/28
172.16.8.48/28
172.16.8.64/28

We could permit all of these networks with on prefix-list statement:

172.16.8.0/24 ge 28 le 28

This will check the first 24 bits to make sure they match. All of these networks have 172.16.8 as the first 24 bits, and it won't care what is in the last 8 bits. Then it will check to make sure that the subnet mask is GREATER THAN OR EQUAL TO 28 bits LESS THAN OR EQUAL TO 28 bits - the only number that works for this is 28 bits. So the first 24 bits in the network must match and it has to have a 28 bit subnet mask. All 5 of our networks would match for this.

We could be even more precise with this and use:

172.16.8.0/25 ge 28 le 28

If we take a look at our 4th octects we will see that for all of them the 128 bit is off so we can check that bit also (25 bits total we are checking).

0 -- 0 0 0 0 0 0 0 0
16 - 0 0 0 1 0 0 0 0
32 - 0 0 1 0 0 0 0 0
48 - 0 0 1 1 0 0 0 0
64 - 0 1 0 0 0 0 0 0

This would be closer to permitting the 5 networks that we have.

We could also permit only the classful networks. The first thing that we need to do is figure out exactly what a classful network is.

For a class A network we know that it has to have an 8 bit mask and must be between 0 and 127 in the first octect. If we break down 0 and 127 we
get:

0 --- 0 0 0 0 0 0 0 0
127 - 0 1 1 1 1 1 1 1

For the first octect of a class A network the first bit has to be a 0, it must be off. So we can do a prefix-list like this:

0.0.0.0/1 ge 8 le 8

In our first octet the first bit is a 0 (which is what it would need to be to be class A), with the /1 we have we are ONLY checking the first bit to make sure it's a 0 (meaning it would be a class A network 0 - 127). We are then making sure that this class A network actually has a class A subnet mask of 8 bits, and only 8 bits would match.

For the class B's we need to make sure that they have a 16 bit subnet mask and that they are in the range of 128 - 191 in the first octet. If we break down 128 and 191 we get:

128 - 1 0 0 0 0 0 0 0
191 - 1 0 1 1 1 1 1 1

The first two bits are what we are going to care about. We need to make sure that the first two bits in the first octet are 1 0 . The first number that we can use as our standard we are checking against is 128 -
128 has a 1 0 as the first two bits in its first octet.

128.0.0.0/2 ge 16 le 16

So we are checking the first two bits to make sure the network has a 1 0, meaning that it must be in the range of 128 - 191. We are then going to check to make sure that it has the classful 16 bit mask, and ONLY a
16 bit mask.

Finally we have the class C networks. Class C networks are in the range of 192 - 223 and they must have a 24 bit mask. If we break down 192 and
223 we get:

192 - 1 1 0 0 0 0 0 0
223 - 1 1 0 1 1 1 1 1

The first 3 bits in the first octet are what we care about. 192 would be the first number we can put in that first octect that will have 1 1 0 as its first 3 bits.

192.0.0.0/3 ge 24 le 24

We are going to check the first 3 bits of the octet and make sure that its 1 1 0 meaning that it has to be in the range of 192 - 223 being class C, then we are going to check to make sure it has a class C classful subnet of 24 bits.

Finally how to permit or deny any could be very helpful since a Prefix-list just like an Access-list has an implicit deny at the end:

0.0.0.0/0 le 32

This is 'any' for a prefix-list. It says check 0 bits; I don't care what any of the bits are. It also says that the subnet mask can be 32 bits or less (down to the number of bits we are checking) down to 0. So we aren't going to check any bits and the network can have a subnet mask of anything between 0 and 32 bits. This would be 'any'.

Now for your Prefix-list:

In the 3rd Octet we have 1, 4, and 5. We'll break these down to binary to see if we can summarize these into one line:

1 - 0 0 0 0 0 0 0 1
4 - 0 0 0 0 0 1 0 0
5 - 0 0 0 0 0 1 0 1

For a Prefix-list we need to go from the left to the right and we can't skip bits. So for these three networks we would need to stop at the 8 bit since it is the last bit from left to right that is the same. This would give us 3 bits that are different, or 8 possible networks. We only have 3 of the 8 possible networks and we should not permit or deny more than we actually have. We should be as specific as possible.

If we leave the 91.86.1.0/24 alone by itself it will give us a Prefix-list of:

91.86.1.0/24

This will check the first 24 bits from left to right to make sure that they match, and it will also check to make sure that it has a 24-bit subnet mask.

For the 4 and 5 networks we can permit or deny both of those with one line. If we take a look at 4 and 5 again we can see that all of the bit's match down to the 2 bit. This would leave 1 bit that doesn't match, which would give us 2 possible networks, both of which we have.
The Prefix-list to permit or deny both 4 and 5 would be:

91.86.4.0/23 ge 24 le 24

This will check the first 23 bits from left to right. The 24th bit could either be off, which would give us 4, or it could be on which would give us 5. Since we have the ge and le involved the /23 is only bits checked. The ge and le specify that our subnet mask must be greater than or equal to 24-bits and less than or equal to 24-bits which means that the subnet mask must be 24-bits for both possible networks.

Cisco Learning Game: There’s an App for that!

2011-06-05T03:03:00.000+05:30

It's Fun! It's Free! It's Mobile! And oh yes, you might just learn something.

Played over a million times on the web, the Cisco Binary game has now gone mobile!

And guess what? It may even be more fun to play on your smart phone than on a web browser.

Never mind that playing the game can help turn you into a binary number guru, or a subnetting superstar, most people like it because it's a lot of fun. The web version of the game is linked to game sites all over the world and has been played in more than 125 countries. It only takes a minute to learn this fast paced arcade-style game, but you quickly get hooked and may have trouble putting it down.

Get it Now!

From your iPhone or iPad go to the App Store; or from any Android device go to the Android Market. Simply search for “Cisco Binary Game”. It only takes a few minutes and the Game App is FREE! You can also play the game online at the Games Arcadehttps://learningnetwork.cisco.com/docs/DOC-11119

If you want to play and learn, any time, any place, download the free game app today. The game records your best scores so you can track your progress – the higher your score, the more you’re learning. Now, get the game and get going!

Self-Paced Lesson: Integrating and Troubleshooting Cisco Unified Presence

2011-06-05T02:56:00.001+05:30

Data_Sheet_CIEVPRES_10-07-09_mjs.pdf (186.0 K)View Download

Cisco IPCC Express Edition Solution Reference Network Design

2011-06-05T02:54:00.000+05:30

cdccont_0900aecd80331ee7.pdf (1.3 MB)View Download

IP Telephony

2011-06-05T02:16:00.000+05:30

Connecting two Cisco Unified Communication Manager Express with H323

The Cisco Unified Communications Manager Express (CUCME) is the new brand name given by Cisco to the older Call Manager Express (CME) system. The concept is the same however: IP Telephony software running on Cisco routers. Therefore, the CUCME is a normal Cisco router (models supported are 1800, 2800, 2900, 3800, 3900 series) with a special IP Telephony software (call manager software) installed on the router’s flash memory. The CUCME system serves as the call control node to facilitate IP Telephony communications in a small to medium size Enterprise.
Usually there is a single CUCME system in each LAN network, with several IP phones connected on the LAN switches. An enterprise with several sites connected over a private IP WAN network can establish full IP voice communications between sites by configuring H323 communication between each CUCME router. A simple example with a two-node topology is shown below.

CME-A node has local IP phones with numbering 500x and a WAN IP address of 1.1.1.1. On the other site, CME-B has local IP phones with numbering 600x and a WAN IP address of 2.2.2.2. By establishing H323 voip communication over the WAN (between 1.1.1.1 and 2.2.2.2) we can have full IP telephony conversations between the IP phones of both sites.
CAUTION: Because the actual VoIP RTP traffic communication between site A and site B will be running from one IP phone to another IP phone, there must be full IP routing established between the IP phone subnets.
The CUCME configuration to establish H323 between the two sites is shown below:
CME-A
CME-A#show running-config
dial-peer voice 6000 voip
destination-pattern 60..
session target ipv4:2.2.2.2
dtmf-relay h245-alphanumeric
codec g729r8
CME-B
CME-B#show running-config
dial-peer voice 5000 voip
destination-pattern 50..
session target ipv4:1.1.1.1
dtmf-relay h245-alphanumeric
codec g729r8
The dial-peer configuration on CME-A tells the system that in order to reach the destination pattern 60xx the session will be established with IP address 2.2.2.2 (i.e CME-B). The inverse applies for CME-B.
Note: Make sure to select one of the high compression codecs ( such as g729, g723) in order to save bandwidth for voice calls over the WAN network. Each VoIP conversation using a high compression codec (g729, g723) will use significantly less bandwidth compared with the traditional G711 codec.

Bsoft bangalore: New CCNP VOICE books download

2011-06-05T02:11:00.000+05:30

Bsoft bangalore: New CCNP VOICE books download: "Solutions Reference Network Design aka SRND - These are pages that cisco provides on their website so that people like us can get a better ..."

Bsoft bangalore: why Bsoft Bangalore ....???

2011-06-05T02:06:00.000+05:30

Bsoft bangalore: why Bsoft Bangalore ....???: "Taking Forward a tradition of Trust BSOFT is one of the leading institutes in the field of IT certification training and the n..."

New CCNP VOICE books download

2011-06-05T02:05:00.000+05:30

Solutions Reference Network Design aka SRND - These are pages that cisco provides on their website so that people like us can get a better understanding of Cisco products. Well, one will get overwhelmed with the amount of info. So pace yourself with it.

Below are a few links -

http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_series_home.html

http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_maintenance_guides_list.html

CCO (Cisco.com) using the 10th: Part Voice Documentation Guide

2011-06-05T01:41:00.000+05:30

CCO (Cisco.com) using the 10th: Part Voice

Documentation Guide

Hello. The bridge. It is hot every day.

As an aside suddenly, Cisco's offices and air conditioning will automatically turn off at 20.

Due to the central air-conditioning abroad, to put off the air conditioner must be put air in English and please ask on the phone yet. .

People to turn on the air conditioner so far not many, after 20:00 every day to work is all in the heat.

Well, this guide is a document edited Voice.

Voice in the illustration below a rough element, is very much a feature of Voice product types.

CCVP, but only to a major product, CCIE Voice exam will be without regret in the end to end.

The router / switch except for the GUI based on the settings.

This CCVP, CCIE Voice of questions referring to the range, Voice Gateway, Switch QoS, CUCM, CME, Unity Connection, Unity Express, CUP, CCX picked up eight.

By the way Voice CCNA CCO recommends the book learning in documentation rather than commercial. Far deeper question is not, itself, has so many products to be fine. . If you are spirited course of interest also to CCO.

Introducing the Voice in the documentation for each product.

Switch QoS ((642-642J) QoS CCIE Voice, etc.)

Switch Please note that QoS is set differently depending on the product.

QoS trial 2950, CCIE Voice QoS is covered in 3750.

http://www.cisco.com/en/US/docs/switches/lan/catalyst2950/software/release/12.1_22_ea11x/configuration/guide/swqos.html

http://www.cisco.com/en/US/docs/switches/lan/catalyst3750/software/release/12.2_52_se/configuration/guide/swqos.html

Voice Gateway ((642-436J) CVOICE 6.0, (642-456J) CIPT2 6.0, (642-642J) QoS, (642-426J) TUC, CCIE Voice, etc.)

Analog IP Gateway Voice is a voice essentially is meant to convert, Cisco's IOS in the Gatekeeper, SRST, CUBE and other features are included in the IOS.

As with switches, IOS QoS of understanding is also required.

It's pretty wide, so luckily the document is Cisco IOS Voice Configuration Library are combined in a page.

http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_voice_configuration_library_glossary/vcl.htm

Cisco Unified Communications Manager ((642-426J) TUC, (642-446J) CIPT1 6.0, (642-456J) CIPT2 6.0, CCIE Voice, etc.)

Voice call control server is performed in the core product. Set in the GUI.

Because of the large amount of documentation, Documentation Guide is a document.

Documentation Guide Tadashi Kono amount of information it also has quite a few, will introduce more choice in my main document.

CCIE Voice on this subject will introduce the 7.0 documentation, please refer to other versions of the documents are needed.

Top Products

Administration Guide

System Guide

Features and Services Guide

Operating System Administration Guide

Serviceability Administration Guide

Troubleshooting Guide

Cisco Unified Communications Manager Express ((640-460J) IIUC, CCIE Voice, etc.)

Call control server running on the IOS.

Capacity and capability to CUCM is lower, but necessary functions as a phone is almost covered.

http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_installation_and_configuration_guides_list.html

Cisco Unity Connection (CCIE Voice)

Voice-mail.

I have a simple automated voice response capabilities as well as send and receive voice mail.

This is also similar to the CUCM configuration GUI.

This product also Documentatin Roadmap is there, picked up the following documents by the example of so many.

Top Products

Design Guide

Interface Reference Guide

Troubleshooting Guide

SCCP Integration Guide

System Administration Guide

User Moves, Adds, and Changes Guide

Cisco Unity Express ((640-460J) IIUC, CCIE Voice)

It is also voice-mail, the router is mounted as a module.

As Express and get a name, capacity and features on Unity Connection is better.

Top Products

Cisco Unified Presence (CCIE Voice)

Enrollment server that manages user information.

Business chat used in conjunction with Cisco Unified Personal Communicator.

This product is comparatively less documentation.

Deployment Guide at the following Configuration Guides please visit the center.

Top Products

Cisco Unified Contact Center Express (CCIE Voice)

Server for Cisco IP Contact Center capabilities utilizing telephony infrastructure

Calls can be dedicated, flexible contact center agents to write the script.

Minute test in which there is a troubling presence. .

Top Products

Solution Reference Network Design for Cisco Unified CCX and Cisco Unified IP IVR, Release 7.0

Administration Guide

Programming Guides

Troubleshooting Guides

It is a product-related documents over, Voice when it is combined with Solution Reference Network Design We recommend that you read it also to the design guide.

Voice of the product work together so that, in order to understand the overall picture is an important document.

Cisco Unified Communications SRND Based on Unified Communications Manager 7.x

Wireless Edition is the next document guide. Update 8 / 20 (Friday) is scheduled.

CCIE VOICE RACK Lab Exam Blueprint

2011-06-05T01:22:00.000+05:30

Written Exam Blueprint

Codecs/Regions
Redundancy (CM Groups/Device Pool)
Dial Plan: Gatekeeper, SIP Proxy, Route Patterns, Route Groups, Route Lists, Digit Manipulation and Digit Analysis
Music On Hold
Conferencing (Audio and Video)
Transcoding
Media Termination Points
CM Features: Extension Mobility, IPMA, Attendant Console, Call Park, Pickup
Phone Settings
CTI, TAPI and JTAPI Interface

QoS

Link Efficiency: LFI, MLPPP, FRF.12, cRTP, VAD
Classification and Marking
Congestion Management: Queuing
CAC: RSVP, GK
Traffic Policing and Shaping
LAN QoS
CM CAC and GK. Hub and Spoke/Fully Meshed MPLS

Telephony Protocols

SCCP
RTP & cRTP
MGCP
SIP
H323
Analog and TDM Signaling

Unity

Integration
MWI, SMDI
DPA
Call Handlers
Unified Messaging
VPIM

IOS IP Telephony Skills

SRST
CME
CUE

IP IVR/IPCC Express

Speech Recognition
ICD Functions
Database Lookups
VXML

Security

DHCP Snooping
MCS OS Hardening
Phone Authentication and Encryption
TCP/UDP Port List
Firewalls and Application Layer Gateways (ALG)
NAT

Infrastructure Protocols

DNS
TFTP
NTP
Inline Power: Cisco and 802.3af

Application Protocols

IP Multicast
Video
Fax and Modem
Operations and Network Management

Lab Exam v3.0 Blueprint

Basic Campus Design
1. DHCP, TFTP
2. Catalyst voice and data VLAN configuration
3. Catalyst VTP configuration
Call Manager and Call Manager Express Configuration
1. Phone registration
2. Phone configuration
Voice Gateway and Signaling
1. Analog and digital voice protocols: analog FXS, FXO-M1, T1, E1, PRI, CAS, R2
2. VoIP protocols: H323, MGCP, SCCP, SIP, RAS
Call Routing
1. CCM route patterns (@ wildcard not tested)
2. CCM route preference and redundancy
3. IOS dial peers
4. Digit manipulation and translation
Voice CODEC
1. G711ulaw, G711alaw, G729, G723
Call Admission Control
1. Location-based
2. RAS-based
High Availability Features
1. SRST
2. AAR
Media Resource Management
1. Conference bridge software and hardware
2. Transcoder
3. MTP
4. MOH
QoS Considerations
1. L2/L3 classifications and policing
2. Queuing mechanisms
3. LFI
4. Catalyst switch QoS
Unified Messaging
1. Unity voicemail integration
2. Unity administration
CRS/IPCC Express Application
1. Default script configuration and integration
2. Custom script configuration and integration
Call Manager Voice Applications
1. Any native applications to Call Manager, examples are extension mobility, attendant console, IPMA
Supplementary Services
Directory Services and Integration
Fax

Fax pass-through
Fax relay

CCIE Voice Lab v3.0 Reading List

Cisco CallManager Fundamentals

Cisco Catalyst QoS: Quality of Service in Campus Networks (paperback)

Cisco Frame Relay Solutions Guide (Chin, ISBN# 1587051168)

Cisco IP Telephony (Lovell, ISBN# 1587050501)

Cisco IP Telephony: Planning, Design, Implementation, Operation, and Optimization

Cisco Voice Gateways and Gatekeepers

Cisco Voice over Frame Relay, ATM, and IP (McQuerry, Foy, McGrew, ISBN# 1578702275)

Configuring CallManager and Unity: A Step-by-Step Guide

Deploying Cisco Voice over IP Solutions

Troubleshooting Cisco IP Telephony (Giralt, Hallmark, Smith, ISBN# 1587050757)

Troubleshooting Cisco IP Telephony

Voice over IP Fundamentals

Voice-Enabling the Data Network: H.323, MGCP, SIP, QoS, SLAs, and Security

CCIE Voice Lab v3.0 Equipment and Software List

Passing the CCIE Voice Lab Exam requires a depth of understanding difficult to obtain without hands-on experience. Early in your preparation, you should arrange access to the equipment listed below:

Lab Equipment:

Cisco MCS-7845 Media Convergence Servers

Cisco 3825 Series Integrated Services Routers (ISR)

Cisco 2821 Series Integrated Services Routers (ISR)

ISR Modules and Interface Cards+ VWIC2-1MFT-T1/E1
    - PVDM2
    - HWIC-4ESW-POE
    - NME-CUE

Cisco Catalyst 3750 Series Switches

IP Phones and Soft Clients

Software Versions

Any major software release which has been generally available for six months is eligible for testing in the CCIE Voice Lab Exam.

Cisco Unified Communications Manager 7.0

Cisco Unified Communications Manager Express 7.0

Cisco Unified Contact Center Express 7.0

Cisco Unified Presence 7.0

Cisco Unity Connection 7.0

All routers use IOS version 12.4T Train.

Cisco Catalyst 3750 Series Switches uses 12.2 Main Train

Network Interfaces

Fast Ethernet

Frame Relay

Telephony Interfaces

why Bsoft Bangalore ....???

2011-06-05T00:45:00.000+05:30

Taking
Forward
a tradition of
Trust
BSOFT is one of the leading institutes in the field of IT certification training and the name BSOFT comes in the top 10 institutions in the category of International Certification in the networking field. This is simply because we at BSOFT provide all Microsoft and CISCO certifications and these classes are handled by totally dedicated staff with a complete backup of a full fledged lab. Each & every student is given individual care because at BSOFT all are special. As an outcome of all this we are able to create certified persons in all levels of networking. The instance that began our growth was the creation of a CCIE lab holder and that too from an academic level and this was soon followed by 18 more CCIE lab holders. We also have to our credit 162 CCVP holders, and many CCSP, CCNP, MCSE holders. Our list of achievers has been growing ever since and there has been no looking back.
To speak about the founders of BSOFT, from the time computers came into existence our MD has been hooked on to it and many of our staffs have also been in this industry from the beginning itself. Being a part of this IT industry right from the beginning has helped us feel and understand the pulse of this world. This in turn has made us understand the needs of this field and we have been upgrading our facilities ever since so that we are able to meet the ever changing needs of this industry. This has immensely boosted our growth and within a very short time BSOFT has grown from scratch to sculpture.
Like we mentioned before, we are totally dedicated & fully equipped with all the facilities and you can be sure that you will avail a good real time exposure and experience. The 24 hour lab facility that we provide is an added advantage to all the students as they can do extra practical sessions. A dedicated class room for each subject gives maximum work time with no interference what so ever. Hostel facility is also provided by us so that students from far off places find it easy to stay and learn.

For fresher’s, we are your safe hand.....for the experienced we will be your dedicated guru. That’s our promise…So
Join BSOFT Bangalore , Be an achiever.

Bsoft Bangalore CCIE VOICE RACK

Bsoft Bangalore CCIE R&S RACK

#30, 1st floor, 1st Main

BTM 2nd stage

Bennergetta Road

Bangalore -76

contact: 080-41465262

Mobile :+919886623909

www.bsoftonline.com