Iam Working as Technical Community Manager for the Voice and Unified communications domain. I will be focusing on IP Telephony and Unified Communication as Community Manager and driving the Document/Videos/Blog Creation and growth of CSC with Customers and Partners.
It is a best practice to configure CUBE for Call Admission Control to make sure it adheres to certain user-defined thresholds. If it exceeds these thresholds, CUBE will stop accepting new calls to prevent any catastrophic effects like crashing the router, or security violation of your network policies. These thresholds can be based on CPU, memory, call spikes, SIP based Denial of Service attacks, etc.
Configuring Call Admission Control on CUBE
Configure CUBE for the following requirements: 1. CAC based on call spike to manage call arrival rate 2. CAC based on maximum simultaneous connections per destination 3. CAC based on different thresholds like total call count, CPU or memory
1. CAC based on call spike:
CUBE rejects calls if call spike is detected
Call spikes can occur due to SIP based Denial of service attacks, e.g., a spike of REGISTER messages, INVITE messages etc. CUBE can detect these spikes defined as a surge in incoming requests within a short window by use of following commands:
Call spike <call-number 1-2147483647> steps <3–10> size <100–250>
Example 1: To configure 10 incoming call requests per 300 milliseconds, configure
CUBE(config)#call spike 10 steps 3 size 100 CUBE(config)#call spike 12
CUBE#show call spike status
2. CAC based on Max Connections:
“Max Connection” feature restricts the number of calls that can be active on a VOIP dial-peer. This feature provides CAC on a per dial-peer basis. For example make sure that only 2 long-distance call is allowed: To achieve this, configure max-conn 2
dial-peer voice 100 voip description *Long distance call to SP* translation-profile outgoing longdistcall
max-conn 2 destination-pattern 91[2-9]..[2-9]......$ session protocol sipv2 session target ipv4:126.96.36.199 session transport udp voice-class sip bind control source-interface GigabitEthernet0/1 voice-class sip bind media source-interface GigabitEthernet0/1 dtmf-relay rtp-nte codec g711ulaw no vad
3. CAC based on different thresholds like total call count, CPU or memory
“Call Threshold” command is used to configure two thresholds, high and low. “Call treatment” is triggered when the current value of a resource goes beyond the configured high. The “call treatment” remains in effect until the current resource value falls below the configured low.
CUBE(config)#call threshold global ? cpu-5sec the CPU utilization in the last 5 seconds cpu-avg the average CPU utilization io-mem the IO memory utilization proc-mem the Processor memory utilization total-calls the total number of calls total-mem the total memory utilization
Call threshold global total-calls low <low threshold> high <high-threshold>
Let’s configure the following thresholds: CUBE(config)#Call threshold global total-calls low 1 high 1 CUBE(config)#Call threshold global cpu-avg low 75 high 85 CUBE(config)#Call threshold global total-mem low 75 high 85 CUBE(config)#call treatment on
CUBE#show call threshold status
CUBE Configuration Guide
... View more
Introduction This document covers the configuration procedure for configuring custom alerts and email settings for Performance counters using Real Time Monitoring Tool (RTMT) Using RTMT it is possible to create custom alerts based upon the values of any Performance Counter. • Open the Performance Tool in RTMT. • Select the counter you wish to use to create a custom alert and double-click to begin monitoring. • Select the counter graph in the main RTMT pane, right-click and select Set Alert/Properties... • Enter a Description and Recommended Action for the alert and choose a Severity then click Next. • Now you must specify the alerting Threshold for the counter, how this threshold is Calculated and the Duration of time that the counter value must be above/below the specified threshold before the alert is triggered. Then click Next. • Now you can use Frequency to limit the amount of times the custom alert will be triggered and Schedule to specify a specific period of time during which the alert is capable of being triggered. This allows you to monitor counter thresholds for a specific period of time only. Then click Next. • Finally you can specify whether to Enable Email messages for your alert, choose which Trigger Alert Action Profile it will use and specify a User-Defined email text message. Configure Example Custom Alert Cisco UP SIP Proxy - NumIdleSipdWorkers It is recommended that on CUP this counter should not be constantly below a value of 5 for a period of 60 minutes. For further details on this counter see the Recommended Counters section. The following is an example of how to configure a custom alert for this threshold. 1. Select the counter in the Performance Tool: 2. Enter a Description and Recommended Action for the alert and choose a Severity: 3. Choose a Threshold, how this threshold is Calculated and its Duration of time: 4. Specify Frequency and Schedule: 5. Configure Email Settings: Related Information Real-Time Monitoring Tool Administration Guide for Cisco Unified Presence Release 8.0, 8.5, and 8.6 Cisco Unified Real-Time Monitoring Tool Administration Guide, Release 10.0(1)
... View more
Some of my information and settings on Jabber Mac client have changed and incorrect.
Reset jabber can fix various issues experienced with Jabber Mac client.
Go to Jabber Menu > Reset Jabber
This option is used to return Cisco Jabber to an initial installation state. The option is only available when the user has signed out of the client.
Allows restoration of the initial state (i.e. sign in as a different user)
All credentials data & cache from service discovery All cached configurations (except provisioned data via URL configuration) Call & chat histories
Will not remove:
Certificates for secure phone Server certificates accepted by user Jabber log files
... View more
This document describes a method to clear the stored cache configuration, logs and Photos on jabber clients.
Sometimes Jabber client displays an incorrect old Photos and contact information because of stored cache on the jabber clients.
Inorder to clear the stored cache on Jabber for windows and Jabber for Mac clients follow the below procedure.
Jabber for windows
The cache is stored in a folder called "Jabber" which can be found at this location:
Under Jabber > CSF you have the following folders Contacts, History, Logs, Photo Cache etc
Exit Jabber and delete the desired folder.
Jabber for Mac
Exit Jabber client and delete the desired folder
Logs - /Users/<userid>/Library/Logs/Jabber/ Config - /Users/<userid>/Library/Application Support/Cisco/Unified Communications / Jabber
Refer this document for more information http://www.cisco.com/c/en/us/support/docs/unified-communications/jabber-mac/116682-technote-jabber-00.html
... View more
In addition to the documents given by Manish, following commands will help you to analyse and troubleshoot the MOH issue you are facing on CUBE.
debug ccm-manager music-on-old all
debug voip rtp debug ccsip all
show version show running configuration When MMOH is being streamed, please collect the following output on CUBE:
show ccm-manager music-on-hold show voip rtp connections show call active voice compact
... View more
Well explained Chris.. Adding more information related to MOH.
MoH is played to a caller when the remote party in the call puts the caller on hold. MoH requires an MoH server to be configured and an audio file that will be played during the hold event. When multicast support is enabled, the multicast stream can increase, based on port number or IP address. For the MoH audio source, additional audio files can be added to the MoH server. Endpoints/devices can be configured to play different audio files as required. An MoH audio source file can be assigned as User Hold Audio Source and/or Network Hold Audio Source to the phones. If an audio source file is defined at the device level, it overrides the device pool audio source preference. To upload an MoH audio file (the file must first be uploaded to the CUCM that is functioning as the MoH server), go to the Cisco Unified CM Administration page and choose Media Resources > Music on Hold Audio Source , click Add New > Upload File , select an audio file you wish to upload as the MoH file, and click Upload .
Media Resource Group and Media Resource Group List After media resources are defined, they need to be managed and assigned to intended devices/endpoints/device pools. Refer the urls
... View more
When matching an outbound dial peer, the Cisco IOS router always uses the destination-pattern command. In case of outbound dial-peer matching, the following rule applies: called number based on DNIS matching the outbound dial-peer destination pattern and most explicit match. Dial-peer routing for POTS is based on the port command and for VoIP is based on the session target command. In certain cases, two or more dial peers may have the same destination pattern—for instance, VoIP dial peers to CUCM subscribers for call processing redundancy. In such a case, the preference command can be added to each dial peer to set a priority, with preference 0 being the default and highest preference. Multiple dial peers can be defined with the same destination pattern with preference 1 , 2 , 3 , and so on.
... View more
SIP Call Trace is a feature in RTMT which let users trace calls and generate SIP message ladder or sequence diagram.Traces provide detailed information about the call and generate SIP messages when enabled on Cisco Unified Communications Manager and that can be useful for troubleshooting call failures on the system.
This document covers the procedure on how to take a SIP call trace on Cisco Unified Communications Manager using Real-Time Monitoring Tool (RTMT)
RTMT is a tool that lets you monitor system health, view graphs and collect logs from Unified CM. There are versions for both Linux and Windows. Unified CM must also be configured to specify what can be traced.
Trace logs called calllogs, is enabled for sip call tracing. RTMT uses these logs to search for calls using the user entered search criteria and generate the ladder diagram.
Log files are downloaded from the server based on time stamps specified in the search criteria.
This tool also allows users to save the ladder diagram in html files on their machine which can be easily emailed for trouble shooting purposes.
How to Enable SIP Call Tracing on CUCM
1. Added Enterprise Parameter to enable/disable Call Tracing globally on Cisco Unified CM
2. Search Calls on RTMT for SIP Calls based on a critera
Session Trace - User can search/trace for calls based on Calling Number, Called Number, Start Time, Duration..
3. Analyze call with the Ladder Diagram
Option to show the "Call flow Diagram" and "SIP message" - Select the corresponding Tab to analyze the call
Installing RTMT and Enable Call Trace on CUCM
... View more
This document covers the procedure for adding the company contacts to the contact list in jabber for windows.
How to add the company contacts to contact list in Jabber for windows.?
You have 2 options.
With the Jabber for windows client 10.x version, contacts can be imported to buddy list from the jabber client itself in the form of xml file
Log on to Jabber for windows client - Choose File -> Import contacts and select the xml file. Refer this link for creating xml file,
The following is an example XML definition of a contact list that you can import into Cisco Jabber: <?xml version="1.0" encoding="utf-8"?> <buddylist> <group> <gname>Sales</gname> <user> <uname>firstname.lastname@example.org</uname> <fname>Adam McKenzie</fname> </user> <user> <uname>email@example.com</uname> <fname>Anita Perez</fname> </user> </group> <group> <gname>Marketing</gname> <user> <uname>firstname.lastname@example.org</uname> <fname>Nancy Fox</fname> </user> </group> </buddylist>
***************************************************************************************************** Second option is thru CUPS. From CUPS 8.x , you can import contact list with the help of BAT file.
... View more
Introduction This document covers the overview of SIP OPTIONS Ping feature overview and the steps for enabling the features. SIP OPTIONS Ping The SIP OPTIONS Ping feature can be enabled on the SIP Profile associated with a SIP trunk to dynamically track the state of the trunk's destination(s). When this feature is enabled, each node running the trunk's SIP daemon will periodically send an OPTIONS Request to each of the trunk's destination IP addresses to determine its reachability and will send calls only to reachable nodes. A destination address is considered to be "out of service" if it fails to respond to an OPTIONS Request, if it sends a Service Unavailable (503) response or Request Timeout (408) response, or if a TCP connection cannot be established. The overall trunk state is considered to be "in service" when at least one node receives a response (other than a 408 or 503) from a least one destination address. SIP trunk nodes can send OPTIONS Requests to the trunk's configured destination IP addresses or to the resolved IP addresses of the trunk's DNS SRV entry. Feature overview With OPTIONS Ping feature UCM periodically sends SIP OPTIONS to every remote destination peer, to detect its availability. If the remote destination peer is unavailable (no response or it responds “408 Request Timeout” or “503 Service Unavailable”), UCM will mark this peer as unavailable. If the remote destination peer is available (any other responses rather than “503” or “408”), UCM will mark this peer as available. UCM will send new INVITE only to the “available” remote destination peers. Only SIP trunk with default type, “type None(default)”, supports this feature This feature is introducted from CUCM 8.5 onwards.. SIP OPTIONS Ping feature has been introduced to allow quick fail over and status report OPTIONS message is sent to the remote destination peer periodically Response from the remote destination peer decides if new INVITE is sent to it In SIP Profile, SIP OPTIONS Ping section has the configurations to control this feature Enabling SIP OPTIONS Ping is recommended for all SIP trunks because it allows Unified CM to track trunk state dynamically rather than determining trunk destination state on a per-node, per-call, and time-out basis. OPTIONS Ping Feature Configuration Here are the steps to enable OPTIONS Ping feature. Create a SIP Profile with OPTIONS Ping enabled. Create a SIP Trunk Security Profile (OPTIONAL, if you want to change the listening port or transport type). Create a SIP Trunk to the remote destination peer and apply the SIP Profile and SIP Trunk Security Profile created above to the Trunk. Go to Device -> Device Settings -> SIP Profile and create a SIP Profile with SIP OPTIONS Ping Enabled Checked and enter appropriate values for the corresponding parameters. Related Information Cisco Collaboration System 10.x Solution Reference Network Designs (SRND)
... View more
This document covers the overview of features offered on Cisco Paging server/ InformaCast.
What features do I get from InformaCast?
Basic Paging vs Advanced Notification
Basic Paging- Free, not licensed
Point to Point and Group Live Audio Paging to/from Cisco IP Phones
Unlimited Groups/Zones of endpoints as configured by admin
Maximum of 50 endpoint devices per group
Paging between sites is supported (Multicast on WAN required)
Advanced Notification- Optional and licensed
Pre-recorded/scheduled broadcasts (school bells/shift changes)
Notification to Jabber IM
Notification to Social Media (key for higher ed)
Communication with mobile and remote users
Triggered notification to/from other systems- M2M input/output (panic buttons, door locks, lights, etc.)
Integration to existing overhead paging systems
Text and Audio to Cisco IP Phones and other endpoints
Broadcasts to IP Speakers
911/emergency call monitoring/alerting/recording
Weather Alerting with CAP
Dynamically-triggered emergency conference calls
CUCM Integration with Cisco Paging Server/InformaCast Configuration Example
... View more
You can look at collecting logs thru RTMT. I would recommend you to download them and then login to RTMT > Double click on the Trace and log central and look up for the event viewer - Application log and Event viewer-System log. you can troubleshoot the IP Phones and IP communicators being unregistered.
Once you have collected them you can see them in Notepad++ using a searching criteria equal to the device name and also the timeframe when the problem happened. You can also collect them from the SysLog Viewer> Expand Application Logs > then Cisco Syslog, you can see it from there if there's a specific timeframe.Still you can download them when you click on "Save"..
Once you got the information you will see message similar to this - with Device name and Device unregistered status.
Apr 25 13:01:22, JAMES-CCM-1, Error, Cisco CallManager, ccm: 9889781: Apr 25 17:01:22.231 UTC : %CCM_CALLMANAGER-CALLMANAGER-3-DeviceUnregistered: Device unregistered. Device name.:SEP0024975A58DE Device IP address [Optional].:10.34.192.135 Protocol.:SCCP Device type. [Optional]:365 Device description [Optional].:8WDN 10 Reason Code [Optional].:13 Device MAC address [Optional].:0024975A58DE IPAddressAttributes [Optional].:0 Cluster ID:StandAloneCluster Node ID:JAMES-CCM-1, 14
Also for troubleshooting IP Phone Registration, using Status Messages and CUCM/Phone Logs collection and Trace/Logs reading
I would recommend you to go thru the below videos which are really helpful https://supportforums.cisco.com/video/12212291/troubleshooting-ip-phone-registration https://supportforums.cisco.com/blog/12088286/troubleshooting-ip-phone-registration-cucm
... View more