Call manager and Cisco IP phones

I would like to know if it's possible to use Cisco IP phones in small environments, without having Call manager, or it's mandatory to have always CallManager if one wants to use the IP phones.
Thank you

You can use Call Manager Express, which runs on cisco 1751/60, 2600 and above routers. it can support up to 120 users. Cisco Unity Express will provide voice mail. this is a network module in 2600 and above routers. for more info, see www.cisco.com/go/ccme

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RTP streaming and Cisco IP phones problem

Hello,
I'm trying to write an application that should dial some numbers and play the voice message from the file into the phone line using Cisco JTAPI and Java Media Framework.
I've found some samples, that seems useful for me, but unfortunately they does not work. There are no any errors and no exceptions, I have no idea what to do.
Small brief: I make a call from one Cisco IP phone (7960) to another using Cisco JTAPI, then I catch the CiscoRTPInputStartedEv event, get the IP and port of the IP Phone and call the RTPStreamer class constuctor with them. It gives no any errors or exceptions (just a message shown below), but there is only silence in the phone line. Message:
Should b streamin'...
Encoding ok?: true
streams is [Lcom.sun.media.multiplexer.RawBufferMux$RawBufferSourceStream;@53d : 1
sink: setOutputLocator rtp://192.168.1.22:20794/audio
Please see the RTFStreamer class code below.
I set the packet size to 160 as reccomended for Cisco IP phones, I use the greeting.wav from Cisco example that properties are 8Khz 8bit mono, but it still doesn't work.
Could you help me? Thank you for any advice!
import java.io.* ;
import java.util.* ;
import java.net.* ;
import javax.media.* ;
import javax.media.control.* ;
import javax.media.format.* ;
import javax.media.protocol.* ;
import stream.*;
public class RtpStreamer
     public static int PlayCounter = 0;
     private RtpStreamer()
          // not supported
     public RtpStreamer(String IP, String Port)
          PlayCounter++;
          new RtpStreamer("rtp://" + IP + ":" + Port + "/");
     public RtpStreamer(String CurrentMediaUrl)
          PlayCounter++;
     System.out.println("Should b streamin'...");
     // Create a Processor for the selected file. Exit if the
     // Processor cannot be created.
     Processor processor = null;
     StateHelper sh = null;
     try
               String mediaUrl = "file:\\C:\\greetings.wav";
     processor = Manager.createProcessor( new MediaLocator(mediaUrl));
     sh = new StateHelper(processor);
     catch (IOException e)
     System.out.println("Exception occured (1a): " + e);
     catch (NoProcessorException e)
     System.out.println("Exception occured (1b): " + e);
     // for loggin purpose
     //sh.setContext( getServletContext() );
     // configure the processor
     if (!sh.configure(10000))
     System.out.println("Configuration failed!!");
     // Block until the Processor has been configured
     TrackControl track[] = processor.getTrackControls();
     boolean encodingOk = false;
     // Go through the tracks and try to program one of them to
     // output ulaw data.
     for (int i = 0; i < track.length; i++)
     if (!encodingOk && track[i] instanceof FormatControl)
     if (((FormatControl)track).setFormat( new AudioFormat(AudioFormat.ULAW_RTP,8000,8,1)) == null)
     track[i].setEnabled(false);
     else
     encodingOk = true;
     else
     // we could not set this track to ulaw, so disable it
     track[i].setEnabled(false);
               // set packet size to 160
               try
                    Codec codec[] = new Codec[3];
                    codec[0] = new com.ibm.media.codec.audio.rc.RCModule();
                    codec[1] = new com.ibm.media.codec.audio.ulaw.JavaEncoder();
                    codec[2] = new com.sun.media.codec.audio.ulaw.Packetizer();
                    ((com.sun.media.codec.audio.ulaw.Packetizer)codec[2]).setPacketSize(160);
                    ((TrackControl)track[i]).setCodecChain(codec);
               catch (Exception e)
                    System.out.println("Error setting packet size in 160: " + e + " in " + e.getMessage());
     System.out.println("Encoding ok?: " + encodingOk );
     // At this point, we have determined where we can send out
     // ulaw data or not.
     // realize the processor
     if (encodingOk)
     if (!sh.realize(10000))
     System.out.println("Realization failed!!");
     // block until realized.
     // get the output datasource of the processor and exit
     // if we fail
     DataSource ds = null;
     try
     ds = processor.getDataOutput();
     catch (NotRealizedError e)
     System.out.println("Exception occured(2): "+e);
     // hand this datasource to manager for creating an RTP
     // datasink.
     // our RTP datasink will multicast the audio
     try
     //String mediaUrl= "rtp://192.168.1.12:20002/audio/1"; // it works without errors
                    String mediaUrl= CurrentMediaUrl + "audio";
     MediaLocator m = new MediaLocator(mediaUrl);
     DataSink d = Manager.createDataSink(ds, m);
     d.open();
     d.start();
     catch (Exception e)
     System.out.println("Exception occured(3): "+e);

BTW is there any solution to figure out if the RTP application makes any network activity or not?

Is it possible to take the CDR data from a v4.2 Call Manager and copy it to a separate server where it would be made available for reporting?

Is it possible to take the CDR data from a v4.2 Call Manager and copy it to a separate server where it would be made available for reporting? We are not interested in migrating the CDR data to v6 because of the concerns it introduces to the upgrade process. Is it possible to get the raw data and somehow serve it from a different machine? (knowing it would be 'old' data that stops as of a certain date). If so, what would be the complexity involved in doing so?
It seems like the CDR data lives within MSSQL and the reporting interface is within the web server portion of the Call Manager... that's as far as we've dug so far.

Hi
It is absolutely possible to get the data - anyone you have in your org with basic SQL skills can move the data off to a standalone SQL server. This could be done most simply by backing up and restoring the DB using SQL Enterprise Manager.
Moving the CAR/ART reporting tool would be more difficult... if you do actually use that for reporting (most people find it doesn't do what they need and don't use it for anything but basic troubleshooting, and get a third party package) then the best option may be to keep your publisher (possibly assigning it a new IP) and leave it running for as long as you need reporting.
You would then need a new server to run your upgraded V6 CCM; you may find you need this anyway.
Regards
Aaron
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What is the recommended Delay/Latency between Call manager and The SRST setup

Hi ,
Need to understand the readability between the CUCM and SRST.
What is the recommended Delay/Latency, Bandwidh between Call manager and The SRST gateway setup.
Regards,
Velu S

Hi Manish,
I've been struggling to get this information and what I could understand from the SRND is that this 80ms is just related to the Intra-Cluster communications (Between Servers UCS), there is no relation with SRST Gateways:
"The maximum one-way delay between any two Unified CM servers should not exceed 40 ms, or 80 ms round-trip time."
From that I assume that the RTD between the SRST and the Cluster should be based on the affirmation below:
"If a voice service is hosted across a WAN where the one-way latency is 200 ms, for example, users might experience issues such as delay-to-dialtone or increased media cut-through delays. For other services such as presence, there might be no problem with a 200 ms latency."

Call manager and rme config archive report

Hi,
I know that rme does not support the call manager for config archive. It is clear.
The RME home page show config archive status for failed devices and there are some devices which are not supported by RME config archive.
I've set this devices to suspended state in RME but config archiv is trying with this devices also.
This is a financial costumer and the local Financial supervisory authority would like to see all devices config is saving successfully.
How can i exclude the call manager and similar devices from config archiv process and status reports?
Regards,

Hi,
I would like to qoute from RME help.
Working With Suspended Devices
Suspended device state cannot participate in any RME application flows but all historical data pertaining to the device will continue to be maintained by RME.
Nevertheless the RME is trying to fetch the config from suspended devices!!!
What is the truth in this thing?
Regards,

Cisco call manager and ip phone software.

Hi everybody.
Does Cisco call manager also include software required for ip phone? Or software for ip phone needs to be installed on tftp server and it does not come with cisco call manager.?
thanks and have a great weekend.

Hi Sarah,
by ip phone software, do you mean Cisco IP Phone Agent software?
if yes, then you need to have Customer Response Solution (CRS) and Call Manager together to setup ip phone software (services).
check this link for further info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a00801c5765.shtml
plz Rate if it helped.
Soroush.

What is the best cisco ip phone for call manager and ipcc practicals

Hi i have recently started my training on Cisco Call Manger and CCIE voice from a leading cisco voice training institute (http://networkerszone.com/), and am working on 7900 series phones, is there any other phone that i should use or this is fine.

Naval,
7900 phones are good enough for both CUCM & UCCE. UCCE however doesnt support all models of 7900 series, please refer UCCE Compatibility Matrix for supported phone models.
You may also use CIPC as agent phone.
GP.
Pls rate the post if it helps !!

Call Manager and Avaya Phones

Hello Guys,
I am running Cisco Call Manager 7.0. I ahve 2 Avaya 4620 IP Phones and want to add them to my existing VOIP network. How do I go about doing this? All of my other phones are Cisco 79xx.

I tkink you can add avaya phone to cisco environment as a thirt party sip phones.

Cisco Call Manager and MGCP Question

Hello,
I appreciate if somebody can help.
Scenario:
Site 1 PSTN E1----VG----Call Manager----VG--- PABX---Site 2 PSTN1 E1
I have configured a dialing pattern on Cisco call manager 6.xxxxxx to Send to VG on Site1
Both VG routers are using MGCP with call Manager.
The problem if from Site 2 tries to call 6xxxxxx the call manager is not routing the call to the VG in site 1.
I did debug ccapi inout and on Site 2 VG the call response was the number unassigned. This means that the call Manager is searching the directory for the destination but it is not searching the route patterns.
Any ideas to override this and ask the call manager to check it's destination pattern?
Thanks,

Problem solved. The VG in Site 2 was in a CSS that is not allowed to dial PSTN.
Regards,

Cisco Call Manager and LifeSize Endpoints

Hello dear support community,
last week I asked which version of Cisco Call Manager supports BFCP (https://supportforums.cisco.com/message/3966334).
Unfortunately I asked the wrong question.
As it turns out I just didn't need to know which version of Call Manager supports BFCP, which I know now, I also need to know if Call Manager is compatible with LifeSize systems?
Here is the setting, so you know what I'm dealing with:
The costumer uses Cisco Call Manager (Ver. 9.1) as SIP registrar for his LifeSize endpoints. The problem is, when he is making a video conference (via SIP) and he wants to share a presentation, the other side either sees him or the presentation, but not both.
We tried it with our Call Manager (Ver. 9.0) and some LifeSize endpoints. As soon as the LifeSize endpoint uses CUCM as SIP registrar, the option to share the presentation is completely gone.
I guess the costumer switched sources from cam to pc. That would explain why he just sees either video or presentation. But, as you might have guessed, it's not an acceptable solution for the costumer.
And yes, we made sure that BFCP is enabled for the endpoints.
So, what I'm asking myself and you is:
Is this known? And more important, is there a reliable workaround?
Thanks a lot in advance.
Best regards
Tobias

I've been down this road and when I asked the question Lifesize's stance was that it wasn't supported by THEM. I pointed out that CUCM now supports line-side BFCP but they essentially shrugged their shoulders. Either they are blatently disabling content sharing by virtue of using a SIP Proxy or they have a priorietary SDP/header that CUCM isn't passing through.
Either you can call Lifesize and shake them up a bit or you can wireshark the SIP dialog to see what the codec asks CUCM before disabling the sharing button. Once you know that you might be able to use SIP transparancy to pass the header through.

Call manager and type of Voip phones

Just wanted to thank everyone for the help.
My background is Telepresence and minimal on Call Manager 8.
Just wanted to ask if I can somehow pull a list of all type of phones connected to the call manager - we currently have 7942 and 7975. I need the list which phone number is using which type of phone.

Hi Mark,
Here are 3 suggestions:
1) Goto Device > Phone, add the 'Directory Number' and 'Device Type' as search criteria and click 'Find'. This is limited as it can't be exported.
2) Run the following SQL Query:
SELECT d.name, n.dnorpattern, dn.numplanindex, m.name as Model FROM numplan AS n JOIN devicenumplanmap AS dn ON n.pkid=dn.fknumplan JOIN device AS d ON d.pkid=dn.fkdevice JOIN typemodel AS m ON d.tkmodel=m.enum JOIN typeclass AS c ON d.tkclass=c.enum WHERE c.name="Phone" ORDER BY m.name, d.name, dn.numplanindex
3) Use a 3rd Party product to extract and export the required data such as PhoneView from UnifiedFX
PhoneView is the most advanced endpoint management product available including the ability to gather extensive device/user information and then interact and export that data.
Thanks
Stephen Welsh
CTO
http://www.unifiedfx.com

E164 dialling through Cisco Call Manager and its Gate Keeper

Hello,
one of our customers has succesfully setup a few Polycom VSX Endpoints and they are registered to a Gatekeeper.
Calls can be made manually in both directions but if a IP Communicator/IP Phone calls a VSX the recent call list will only include the IP Address of the Gatekeeper and not the E.164 Number so calls cannot be returned.
Is there a way to configure this ?
I am unaware of the Version of the CCM.
Regards
Steff

configuration has two gatekeeper zones. An internal gatekeeper
zone and an external gatekeeper zone.
Devices registered to the internal zone can call devices on the external
Calls through H.323 address but not by e164 alias. However the devices on external zone
Can call devices on internal zone through e164 alias.
Call Manager is not involved in this problem.

Call manager and IP phones

after the basement remodel i will have a server and computer repair area for my testing. I have a call manager server and a couple of ip phones and would like to setup a phone at each of my families houses. what would i need and how would i set it up.

there might be a way to do it using SIP but I have not done it and I am just giving you an idea of how to do it with SCCP default and in a secure way
Forget about the router model I was talking about.
You would have to assume that your home where the CCM is as the Main site and pretty much have all othere families as remote sites, you would need to setup VPN between the houses and if you want everyone to dial each other you would need DMVPN(Dynamic Multipoint).
You would need a good router at the main site, depending on how many sites you have you may be able to use just a 2801. For the remote sites, I would think you can use an 871 or 831 router which does VPN and also has a few PoE ports switch.
You would have to create tunnels between each other house.
JoeL

Call Manager and Keepalives

Does the call manager send out a keepalive when a phone is registered and if it does how often and how big is the packet?

The Phone sends the TCP Keepalive (TCP KA) to the primary Callmanager every 30 seconds after registering. It also sends TCP connect to the secondary Cisco callmanager so that when it does not get the keepalive from the primary it registers to the secondary callmanager.

Call Manager and HP switches?

I have a scenario of having an infrastructure of HP Procurve switches. I have 802.3af PoE to the access layer. The switches support QoS, VLAN, LLDP (not CDP), etc... I am interested in Cisco Call Manager to replace current Avaya based PBX's. Can this be done? Better yet, do you think Cisco will support it?

Cisco phone support LLDP which will act the same as CDP. I have done this on 796x and 7975 phones and it works great. You just need to make sure you are on the lastest firmware that supports it.
http://h40060.www4.hp.com/procurve/uk/en/pdfs/application-notes/AN-C3_ProCurve-Cisco-IP-phone-7900-final-101308.pdf

Call manager and Cisco IP phones

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