CallManager/Voip call monitoring tool?

Any one out there there using a 3rd party or Cisco's new monitoring server for voice? Looking for a third party or Cisco's tool for monitoring voice. We have a CAT3 enviroment that we are about 100 percent sure will work with the IP Phones, but if problems come up after deployment from dropped packets or old cables, we want to be able to narrow down where the bad patch cable is or locatio run that is failing.
Thanks for your input

Hi,
There is a great tool from Prognosis - Integrated Research. They have an assessment product and they have a monitoring product for ongoing management.
http://www.prognosis.com/Products_and_Solutions/IP_telephony_monitoring/VoIP_network_assessment/page__1395.aspx
From a troubleshooting point of view. Fluke has a great handheld tool that tests voice from the wall point. Net tool is the entry level one.
http://www.flukenetworks.com/us/Solutions/Handheld+Network+Testers/Products.htm
Fluke also has other full network management tools with voice in mind.
Cheers,
Tim.

Similar Messages

Where can i check in the real time monitoring tool if there are active calls in progress?

where can i check in the real time monitoring tool if there are active calls in progress?

Hi
Selecting Call Manager -->Call Process ---> call activity
HTH
Regards
Carlo

Lync 2013 Stress and performance tool - VoIP calls

I want to test only VoIP calls using the Stress and Performance tool. Does the tool make real SIP calls to the Front End server? It seems like its only simulating a VoIP connection and not doing a real SIP call --I don't see the SIP SDP setup
or RTP media stream. Please confirm this.
I configured the Load Configuration Tool ( UserProfileGenerator.exe ) to only use VoIP and configured for Load Level: medium. I only see between 2 and 4 Voip calls in progress at any given time. How can I increase the calls to 50+ simultaneous
calls?
For the UC/PSTN Gateway test, it says I need to run this on 2 clients. How does this work? I ran the RoutingRules.ps1 script, and it changed the voice routes and broke the current routing. This is a lab setup, so I'm only setup with
4 digit dialing between all clients and an Asterisk trunk. It doesn't seem like this will work unless I change to E.164 numbers for everything. Can someone share more information on configuring this?
Thanks,
Jeff

The tools can only create users, contacts, and distribution lists, and simulate user load.
Please check the Number of Users you specified in Common Configuration, you can check
http://technet.microsoft.com/en-us/library/gg679085(v=ocs.14).aspx
Lisa Zheng
TechNet Community Support

Reporting (Monitoring) Tool for CUCM 8.6

Hi ,
I am looking for a reporting or statistic tool for CUCM 8.6. , better we search a Monitoring tool .
Our main concrn is that we would like monitoring the Lines , the Phones , Calls (In und Out) , Count of Calls etc. not only print but a dashboard also .
Could you recommend me a programm by your experience ?
Thanks for your support
Homayoun

Here's a quick summary of what to do to view the number of registered phones per subscriber. There's lots more detail on all the available parameters in the docs from that link I gave above.
Open RTMT and select the Default profile. From the left hand window, select the "Performance" option.
This will open the Performance screen. Pick one subscriber from the list of nodes, expand it out, and then expand out the "Cisco CallManager" branch. Select and drag the Registered HardwarePhones option to one of the six blank squares.(You can adjust the divider to make reading the labels easier - but selecting the divider isn't always easy) You can repeat for all your other subscribers.
You can add extra tabs by selecting Edit (From the top-level menu bar) -> New Category.
GTG

VOIP call Failed (Error Code 15) in Prime Event log.

I'm seeing the following events in Prime Infrastructure related to Cisco 7925 phones.
VOIP Call Failed for "(Error Code 15) occurred on client "xx:xx:xx:xx:xx:xx' which was associated to AP <APName> on interface 802.11b/g.
I can't seem to find where this error code 15 is defined. The closest I was able to find was error code 15 under gatekeeper codes in the following document which may or may not be relavent.
http://www.cisco.com/en/US/docs/ios/voice/monitor/configuration/guide/vt_voip_err_cds_ps6350_TSD_Products_Configuration_Guide_Chapter.html
Any insight into this would be appreciated.
Jake
P.S. No voip call jokes on 802.11b/g please

It had 18Mbps set as the min phy rate. I also think it has to do with CAC issues at 18Mbps that are pretty well noted and differeing mandatory rates across the 802.11b/g/n and 802.11a/n networks on the controller. When the phone would roam onto a radio with 18Mbps was the min mandatory rate, it would drop and give this error.

Set up VoIP call termination

Hi, my client is send to SIP G711, G729 Voip calls through the internet, how can I use cisco router 2811 for terminate the calls? I do not have a PSTN. Thanks

You just need to take one minute to search CCO for "SIP trunk configuration" to get the docs
Cisco CallManager Express (CME) SIP Trunking Configuration Example
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml
HTH
java
If this helps, please rate
www.cisco.com/go/pdihelpdesk

Notifying PBX if Voip calls can't be completed from WAN

I have 10 2621XMs each connected directly to G3 PBXs. They supply VOIP call lines to the PBX via a direct T1 (NM-HDV-1T1-24). However, when the direct T1 is up between the PBX and the CISCO router, how do I tell the PBX that VOIP calls can't be completed? This, in it's simplest form, would be due to a Frame outage on the WAN side, or perhaps internal call issues in the cloud. Using basic configs, the PBX will only know it when the direct connect T1 is down not the WAN T1 on the other side of the router. Is there any proven ways to track other interfaces or routing and convey that to the voice-ports by seizing the unused ports or taking down the direct connect T1?

Hello there,
YES you can. The feature is known as "Busyout Monitor".
Please check at:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1830/products_feature_guide09186a0080087b50.html
You can find gatekeeper enhancements for that feature at:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps1839/products_feature_guide09186a0080110ba7.html
Also for latest features available into that direction (monitoring packet loss, delay) please check the URL below:
http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps5012/products_feature_guide09186a00800879da.html
Those will surely help you to understand the feature.
REGARDS
Dragos

Monitoring-Tools

Hi,
are there any monitoring tools for web services. How can I observe my web service and tryck the accesses for it. Any suggestions?
Regards,
Mischa

I have written a free Java performance monitoring tool called JAMon that you might be interested in. It couldn't be easier to use and provides a wealth of information including performance and scalability statistics and more. Other features include the ability to monitor production code, and to disable monitoring at runtime.
To see the type of information that JAMon generates look at the following sample
JAMon report:
http://www.javaperformancetuning.com/tools/jamon/JAMonAdmin.html
To learn about performance monitoring, how to use JAMon, and how to
download JAMon go to:
http://www.javaperformancetuning.com/tools/jamon/index.shtml
If you know of anyone that wants to monitor there code, please refer them
to this link. Thanks. Let me know what you think.
Steve - [email protected]
Note:
JAMon is being hosted by Jack Shirazi, author of the O'Reilly book "Java
Performance Tuning" - http://www.oreilly.com/catalog/javapt

WebLogic Monitoring tools

Are there any monitoring tools (say monitoring weblogic server is up or not , finding no of JMS queues and active connections, finding how much java heap is being used etc) available for WebLogic Server? Are there any documents available from Bea detailing the classes available for accomblishing the same.
Thanks
Venkat

Venkat,
There is a company called Precise (www.precise.com) that offers products for
WebLogic, Oracle, EMC, some ERP systems, and CISCO, as well as software that
ties all the information together. AFAIK, their WebLogic monitoring product
is currently only available as part of a "solution" (i.e. it comes in a
package with a consultant and may be bundled with some of their other
products).
Peace,
Cameron Purdy
Tangosol, Inc.
http://www.tangosol.com
+1.617.623.5782
WebLogic Consulting Available
"Venkat" <[email protected]> wrote in message
news:3a313863$[email protected]..
>
Are there any monitoring tools (say monitoring weblogic server is up ornot , finding no of JMS queues and active connections, finding how much java
heap is being used etc) available for WebLogic Server? Are there any
documents available from Bea detailing the classes available for
accomblishing the same.
>
Thanks
Venkat

VoIP / Call Manager Vulnerability

Has anyone performed a vulnerability scan for VoIP/Call Manager? If so what were the tools you used to perform the scans/vulnerability assesment of the system?
What are some things I should look for?
Did you do any kind of penetration tests?

Nessus is the best freeware tool. If you have
the $$$, go with a product from Spirent.

Monitoring tool needed

Hi guys.
My company will be needed for a monitoring tool that will help assist the dbas in their daily activities. Yeah one can think of 10g grid, but can any one suggest for a more sophisticated tool that will help us record downtime details, capactity planning and all the rest? (even those that arent free).
Thanks.
DID

Veritas I3 is a very good product (used to be called Precise for Oracle).
It will monitor everything from your storage server, to your databases (sqealserver and Oracle) to your web servers to your middleware servers and a few other things you probably never heard of.
Not blinking lights or pulsing data flow diagrams like quest. And is freakishly expensive and has incredible hardware requirements so there is a large TCO involved. The data is good, has SLA tracking, as much detailed history as you want. We kept the history at 6 months, so it was about 55 gig of storage for that.
But we liked it. We dropped it because it lost its bang for the buck and the hardware requirements was beginning to become more than the databases and applications it monitored.

New FIOS customer with dropped VOIP calls and Internet connection

I am a new FIOS customer. Got my 50/25 connection a week ago, switching from a TWC 6/1 connection. Ever since the new connection, I've had numerous issues.
My VOIP (Ooma) connection constantly drops and re-connects during conversations
I've had random Internet connection losses, which picks up again after a few minutes
My home alarm starts chirping every once in a while
I've contacted Verizon several times due to these problems and have received varying answers with no resolution of the problem.
The first time I spoke with support, the tech logged into my router and changed the WiFi channel saying that would fix the problem. It didn’t.
The second time I contacted them, the tech ran a bunch of diagnostics and said everything looked fine so it must be an IP address conflict with my devices, because I had a couple devices using static IP addresses. He said everythinf should be DHCP and the last two digits could not be higher than 99 (192.168.1.99). He said FIOS does not support 3-digit numbers at the end.
So I changed all my devices to DHCP and ran some online VOIP tests. It showed a packet loss of 2-5% and MOS score of 1 (which is bad). I was still getting dropped connections, so I disconnected all devices and connected just one computer to the router and tested again. I was still getting packet loss.
Then I called support a third time, this time the tech said there were no 2-digit IP restrictions and that he was detecting there was no UPS baterry backup for the ONT which was probably causing the problem, so he dispatched a field tech to my house.
Today the field tech came (same guy as before), he took one look at the box and said it was too close to my Electric meter and the RF from the meter was causing interference to the FIOS connection and resulting in dropped connection.
He moved the ONT to another location and said that should fix it.
Well, I'm still seeing packet loss and low MOS score when I run the VOIP test.
I don't know how much of what the techs are saying is true and how much is made up stuff.
Has anyone had similar issues and have thoughts on solutions or likely causes for dropped VOIP calls and connections? Could RF be causing this?
I thought going from a 6/1 Cable connection to a 50/25 FIOS connection would be awesome, but this has turned out to be a nightmare, and I may have to switch back to cable if the problem is not resolved.
I would appreciate any help.
Thanks!

Don't know where the packet loss is happening. I ran the VOIP test on myspeed.visualware.com and it shows a packet loss of 2-5% at different times and a MOS score of 1.
The report says MOS should be around 4 for good VOIP calls.
The Verizon tech who came to the house just blamed the electric meter box for RF interference and move the ONT farther away.
My concern is that I'm getting different answers from different techs at Verizon.
Regarding IP addresses. The Router shows a DHCP range from 192.162.1.2 to 192.168.1.254 as available for devices on the network. So, if I need to assign a static IP to a device should I use a number below 99 or above 151?
Thanks!

Oracle EPM 11.1.2.1 Monitoring tools

Gurus,
could you please let me know what are the available system monitoring or performance monitoring tools for oracle EPM 11.1.2.1 or in case licenses are involved what are the best key areas for manual procedures to monitor the system?
Many thanks !

Hello,
Monitoring is a broad term. There are many 3rd party tools available that can provide monitoring, it all depends on what exactly you're looking for.
Oracle has a document on setting up Oracle Enterprise Manager to monitor Hyperion Applications: http://www.oracle.com/technetwork/middleware/bi-foundation/sys9-oem-integration-1-129287.pdf. For performance monitoring, you can also look at this: https://blogs.oracle.com/pa/entry/epm_11_1_2_how3. This will basically connect to the JRockit JVM and look at the performance for the web application.
If you want to do more monitoring (services, servers, processor, CPU), you can look into Hyperic HQ: http://www.hyperic.com/. They have both an open source version and an enterprise version. I've used the open source version myself and it is pretty good and plays well with Windows. You can see many metrics, graphics and information via the web based application. You can also setup email alerts on certain activities (service stop, service start) and also monitor website availability. For being an open source software, I think it offers alot and it is very reliable.
Again, these are just a few suggestions. There are many tools out there that can accomplish what you're looking for.
Regards,
Luis

Transfer VOIP Calls Between Cisco Desk Phone and Cisco Jabber For IPhone 9.5

Does anyone know how to transfer an active voip call from a Cisco IP Desk Phone to Cisco Jabber for IPhone? I can transfer a call from Cisco Jabber for IPhone to my Cisco IP Desk Phone no problem. I put the call on hold and then click "Resume" on my Cisco IP Desk Phone. However I cannot do the same but the other way around. If I put the call on hold on my Cisco IP Desk Phone, I see "no active call" on my Jabber client. The only information I could find slighlty relevant was using the Mobility Key/Remote Destination Profile feature however this defeats the object as this will forward to an external number, e.g. mobile and I just want to transfer the call within the VOIP environment between the two devices that are using the same directory number.
I am using Cisco Call Manager 9.1(2), Cisco Presence 9.1 and Cisco Jabber for IPhone 9.5.
Any help would be greatly appreciated.
Kind Regards,
Paul Parker.

Did you ever find an answer to this ?
I am seeing the same behavior and trying so see if I can put calls on hold and pick them up both ways also.
The only answer I seem to have found is to use park instead
That would/should work but I would just prefer to hold/unhold
Just not sure why we would not be able to hold/unhold on what is essentially a "shared" line
Does anyone have this working for them ?

How to change font size in a call-out tool box?

Hi, the other day i inserted a call-out tool box in a PDF document, however i cannot find anything that can change the font size of the letters i input in that box. could anyone help? Thanks.
Sherman

This is a question for the Acrobat support forums – please post there.
From: Adobe Forums <[email protected]<mailto:[email protected]>>
Reply-To: "[email protected]<mailto:[email protected]>" <[email protected]<mailto:[email protected]>>
Date: Wed, 23 Nov 2011 10:55:07 -0800
To: Leonard Rosenthol <[email protected]<mailto:[email protected]>>
Subject: how to change font size in a call-out tool box?
Re: how to change font size in a call-out tool box?
created by danmarch<http://forums.adobe.com/people/danmarch> in PDF Language and Specifications - View the full discussion<http://forums.adobe.com/message/4043425#4043425

CallManager/Voip call monitoring tool?

Similar Messages

Maybe you are looking for