CUCM 7.0 Domain Placement

I am in the process of desiging a new CUCM 7.0 system and wanted to get some feedback on where within our corporate domain the system should be placed.
My end goal is to have LDAP integration so that I can pull user information from my active directory. I will also have a Unity Connection server and UCCX server.
What is the best practice on putting these servers within a domain?
Should they be placed in the ROOT or a Child domain?

Hi!
Please be aware that CUCMis an appliance that runs on a customized embedded operating system, and it cannot currently be added to a
Microsoft Active Directory domain. However, you can integrate the server with your LDAP directory. For more information, please refer to the
SRND:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/7x/directry.html
HTH, if it does, please rate.
Regards,
Teresa.

Similar Messages

Asking for advice for Jabber deployment - multi CUCM cluster\AD domains

I would like some design advice for deploying Jabber and CUPS in our company. We have 2 locations, west coast (SiteA) and east coast (SiteB). Each site have their own CUCM 7.15 clusters, Unity clusters, AD domains (trusted, but not in the same forest).
At SiteA I have setup CUPS (8.6.3.10000-20) and jabber and have it working great.
I would like to setup CUPS\Jabber for SiteB, but they need to be able to IM\call\etc to SiteA (And vice-versa).
SiteA and SiteB both have CUCM LDAP sync turned on, and LDAP directory synced with both domains (although SiteA cannot authenticate to CUCM at SiteB, and vice-versa due to the fact you can only LDAP sync authentication with one domain, CUCM user database contain users from SiteA and SiteB).
We have SIP trucks setup to pass internal calls and line status(BLF) between the trunks, and can communicate via internal extensions just fine.
The problem I’m running into is my jabber-config files uses the EDI directory – which can only look at one domain, so I cannot search the other domain. I believe changing to UDS fixes this, but I understand it would require me to upgrade both CUCM clusters to 8.6.2 - unless I’m mistaken.
I’m aware the desktop sharing will not work until CUCM is upgraded to 8.6.1 or 8.6.2.
I’m wondering if anyone has any advice, or can confirm I’m on the right track. Thanks in advance!

The thing that's important to understand is how CUP and Jabber build the XMPP URI. The URI has a left- and right-hand side; the left is the username while the right is the XMPP domain. CUP uses the LDAP attribute specified in CUCM's LDAP System page, sAMAccountName by default, for the left-hand-side. The right-hand side is the FQDN of the CUP cluster. Jabber must use the same values as CUP when displaying search results. Take note that nowhere in this process does the entire XMPP URI originate from the directory source.
In your case you have two separate CUP clusters in two separate domains. This won't work because when a user searches for a contact in the directory using Jabber, the client will build the XMPP URI as [email protected]. Even if you got the other domain's user objects into the search results the right-hand-side of the URI would be wrong and the presence subscription would never succeed since the other cluster is in another domain. As such your first task must be to move the CUP clusters into the exact same fully-qualified DNS domain. Once this is done you can use Inter-Cluster Peering to build a larger XMPP network in which all users have the same presence domain. If you intend to do Inter-Domain Federation in the future this must be your public DNS domain, not your internal active directory domain. If you use a non-public DNS domain TLS handshake will never succeed for inter-domain federation requests.
Once you have Inter-Cluster Peering in place you can use Active Directory Lightweight Directory Services (the new name for ADAM) to front-end both forests. Both CUCM clusters would need to import the full list of users representing both domains and the sAMAccountNames must be unique across both domains.
Finally, you can instruct Jabber to use UDS and query it's local CUCM cluster which will be able to return a search result from both domains. Since the CUP clusters are peered in the same domain the XMPP URI can be built properly, the presence subscription can be routed to the correct cluster, and life will be good.
By this point hopefully it's clear that EDI won't cut it since it would be limited to only returning search results from the local forest.
Please remember to rate helpful responses and identify helpful or correct answers.

Cross Domain Placement

I seem to have a complete crossdomain.xml that should work
fine. However, I am in a situation that hopefully can be fixed.
Here at the U we share the www2.med.umich.edu with other
departments. Our IT department is throwing a fit about putting the
crossdomain into the web root.
However, there is supposedly a web root set for one of our
areas 'prmc'. I added the crossdomain to that directory since that
is where the rss files (xml) are located. It still seems to try to
pull the crossdomain from www2.med.umich.edu and not
www2.med.umich.edu/prmc.
Is there anywhere to specify the path to the cross
domain?

Figured it out.
In case anyone else has this problem:
Have to import flash.system.Security;
Then in you init add:
Security.loadPolicyFile("
http://www.yourwebsite.com/anotherfolder/crossdomain.xml");

Finesse silent monitoring vs CUCM whisper coaching

Hi all,
My understanding is both of these function uses the same resources from CUCM which is the BIB. And the only way to do whisper coaching is through CUCM. So for all the supervisory function (silent monitor, barge, intercept), they will be able to use Finesse Supervisor but when they want to do whisper coaching they will need to use the IP Phone directly by doing silent monitoring first then do whisper coaching. Is that the correct flow?
If setup in such a way, will there be any conflict between Finesse & CUCM?
Thanks!
-JT-

You can check out the 3rd Party solution "MARS Silent Monitoring & Whisper App for CUCM" in Cisco Market Place
The solution requires only CUCM.It does not require Cisco UCCX/UCCE.

How to set the caller tune on PRI-Line/CUCM 8.X

Hi Team,
If passible to set the caller tune songs or advertising in CUCM 8.x -( in place of ringing).
How to set the caller tune for incoming calls( advertising), including 4 digit extension. This feature is available or not (cucm 8.0.3) Kindly verify and confirm
regards,
Johnson

Dear Team,
Now i have configured the Call Queuing, Its working fine, But how to verify to incoming calls or Missed call are in queuing, how to showing.
Connected call are showing, but unable show missed call like who ever in queue.
Regards,
Johnson.

CUCM 10

Hi,
Can anyone tell me where I can find CUCM 10?
The place where it should be (Cisco Unified Communications Manager Version 10.0 / Unified Communications Manager / Cisco Unity Connection Updates)
has only the page server?
Where is the CIUCM 10.0 iso?
Thanks,
Jan

Hi,
if you want to install it on a lab before deploying it to your customer, being a partner, you can order an NFR kit through CCW with a considerable discount.
HTH
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Carlo
Please rate all helpful posts
"The more you help the more you learn"

Frustrated....it has limited chapter markers...is there other dvd authoring

Look like DVD SP is not comptiable with education projects. My projects are usually have alot menus and alot of chapter markers.... I had to split video tracks into few because of too many chapters and actions. Do you know if that DVD SP is possible to pass limited chapteer markers? Or there is good DVD authoring software for Macitnosh? Someone suggested me to use Sceneraic (PC software) and it is way too expensive. Hope there I will get a help.

Oh, Guys,
While we're at it, There are other serious limitations DVD SP has when it comes to the larger projects requiring lots of end actions (over the years, every time I read the DVD-Video Specifications, I find one or more additional cool capabilities DVD SP left out):
1) Cells. DVD SP only allows creating 256 cells in the *entire Video Title Set*. The DVD-Video Specifications provides for thousands in the VOB, grouped in thousands of possible PGCs which DVD SP also has no way to construct (see below).
The paltry few cells that DVD SP does allow placing per VTS, are not allowed to have end actions associated to them, or be targeted by the Connections. Only cells flagged as Markers can have end actions - and then only 52 of the Markers in a Video Title Set (Track) can have end actions (I had a post around here about a year ago explaining why); and only cells designated as Markers show up as navigation targets.
That's a serious limitation for larger projects - and for many smaller projects.
Cells in DVD SP are good to bound buttons on video, and set as the layer break, is about all.
2) PGCs. DVD SP's method of allowing extra one sequential title PGCs (Program Chains) is by creating Stories. But all the Stories in the VTS can only be constructed from the same Markers (same video segments) in the Track. In Scenarist, you can create thousands of PGCs and cells to have all the end actions you need - Check out the logical element and asset domain placement scenario in the "Scene It" DVD games, authored in Scenarist. Thousands of all the logical constructs allowed by the Specifications are there to check out. Note: Those discs can take quite a while to load in the DVDAfterEdit demo, or other IFO Editor browser.
But even though DVD SP allows the creation of Stories (one sequential Title PGCs in the VTS), it does not allow the creation of non sequential PGCs (except for one or two non sequential PGCs in one limited in usefulness circumstance) - which is where the heavy duty arrangement of thousands of PTT's (chapters), Programs, and Cells take place. Gotta be able to create bunches of those non sequential PGCs for those large interactive projects.
I'm preparing an article that goes over the 'regular' limitations of DVD SP and how it has set up some less than optimal functioning for what it does allow. For instance, large projects can slow to a crawl and drop in player compatibility percentages during navigation of the DVD SP generated disc, simply due to the default structures users often create. For instance, the biggie - too many Tracks; There are penalties for trying to use *an entire Video Title Set*, complete with full VTSM menu domain structure (DVD SP hides this from the user), in the place of a non or sequential Title PGC.
There's more to it, of course. But I've spent the last 11 years working on ways to get around these DVD SP limitations. And for most project types, I've been successful helping my authoring studio clients. But one thing all serious DVD SP authors with good paying clients need to consider, is getting licensed with Scenarist Professional ($5,000), which I did several years ago. When you see how blazing fast even simple and complex DVDs output by Scenarist navigate, compared to the same project output by DVD SP, it will be become very clear which project types should be done in which app; and your client will be best served.
Take care,
Trai
New Constellation Technologies / TFDVD Research Labs

Creating multiple mailboxes using Powershell for users who already exist in AD

Hi there,
Wondering if anyone can help, I am wanting to use Powershell to run a command which references a CSV file to create a mailbox for a user who already exists in AD.
It needs to reference the username, to ensure the correct user has a mailbox created (as mentioned they exist already in the domain), place them on a specific database (e.g DB1) and assign them an email address as well as possibly an alias email address
too.
Can anyone provide me the headings of the CSV file which would be needed as well as the powershell command to run. It would be great too if I could also set the quota sizes for the user at the same time of creating the mailbox, but if not possible or complicated
I can run a script afterwards to do this. I would only want to warn them when their mailbox is 1.5GB in size and stop them sending when it reaches 2GB.
Thanks

So create a csv file with the following headers and data
UserID, Alias
UserID should be the user SamAccountName, and the Alias is the Mailbox Alias that you want to set. Supposing that the csv file is named users.csv and located under C:\ , open the Exchange Management Console and run:
$users = Import-Csv c:\users.csv
Foreach ($user in $users) {
Enable-Mailbox -Identity $user.UserID -alias $user.alias -database 'DB1'
Set-Mailbox -Identity $user.UserID -IssueWarningQuota 1.5gb -ProhibitSendQuota 2gb
Please Mark As Answer if this helps
ammarhasayen

WLST offline writeDomain

I've been running WLST in off-line mode and when I try to write the modified domain, I get an exception which doesn't help me very much. I just wonder why this method is called: "SecurityInfo.getAllUsersInAdminRole"
[wlst] com.bea.plateng.domain.script.jython.WLSTException: java.lang.NullPointerException
[wlst] at com.bea.plateng.domain.script.jython.CommandExceptionHandler.handleException(CommandExceptionHandler.java:51)
[wlst] at com.bea.plateng.domain.script.jython.WLScriptContext.handleException(WLScriptContext.java:1373)
[wlst] at com.bea.plateng.domain.script.jython.WLScriptContext.writeDomain(WLScriptContext.java:712)
[wlst] at jrockit.reflect.VirtualNativeMethodInvoker.invoke(Ljava.lang.Object;[Ljava.lang.Object;)Ljava.lang.Object;(Unknown Source)
     [wlst] at java.lang.reflect.Method.invoke(Ljava.lang.Object;[Ljava.lang.Object;I)Ljava.lang.Object;(Unknown Source)
     [wlst] at org.python.core.PyReflectedFunction.__call__(PyReflectedFunction.java:160)
[wlst] at org.python.core.PyMethod.__call__(PyMethod.java:96)
[wlst] at org.python.core.PyObject.__call__(PyObject.java:270)
[wlst] at org.python.core.PyInstance.invoke(PyInstance.java:261)
[wlst] at org.python.pycode._pyx5.writeDomain$14(/tmp/WLSTOfflineIni7092.py:70)
[wlst] at org.python.pycode._pyx5.call_function(ILorg.python.core.PyFrame;)Lorg.python.core.PyObject;(/tmp/WLSTOfflineIni7092.py:???)
[wlst] at org.python.core.PyTableCode.call(PyTableCode.java:208)
[wlst] at org.python.core.PyTableCode.call(PyTableCode.java:267)
[wlst] at org.python.core.PyFunction.__call__(PyFunction.java:172)
[wlst] at org.python.pycode._pyx15.f$0(/home/xxxx/projects/tromskraft/customerportal/devenv/portal/build/filtered.py:112)
[wlst] at org.python.pycode._pyx15.call_function(ILorg.python.core.PyFrame;)Lorg.python.core.PyObject;(/home/xxx/projects/tromskraft/customerportal/devenv/portal/build/filtered.py:???)
[wlst] at org.python.core.PyTableCode.call(PyTableCode.java:208)
[wlst] at org.python.core.PyCode.call(PyCode.java:14)
[wlst] at org.python.core.Py.runCode(Py.java:1135)
[wlst] at org.python.core.__builtin__.execfile_flags(__builtin__.java:308)
[wlst] at org.python.util.PythonInterpreter.execfile(PythonInterpreter.java:158)
[wlst] at weblogic.management.scripting.utils.WLSTInterpreter.execfile(WLSTInterpreter.java:305)
[wlst] at weblogic.ant.taskdefs.management.WLSTTask.execute(WLSTTask.java:108)
[wlst] at org.apache.tools.ant.UnknownElement.execute(UnknownElement.java:275)
[wlst] Caused by: java.lang.NullPointerException
[wlst] at com.bea.plateng.domain.security.SecurityInfo.getAllUsersInAdminRole(SecurityInfo.java:587)
[wlst] at com.bea.plateng.domain.security.SecurityInfo.getDefaultAdminID(SecurityInfo.java:73)
[wlst] at com.bea.plateng.domain.script.ScriptExecutor.writeDomain(ScriptExecutor.java:572)
[wlst] at com.bea.plateng.domain.script.jython.WLScriptContext.writeDomain(WLScriptContext.java:704)
[wlst] at jrockit.reflect.VirtualNativeMethodInvoker.invoke(Ljava.lang.Object;[Ljava.lang.Object;)Ljava.lang.Object;(Unknown Source)
     [wlst] at java.lang.reflect.Method.invoke(Ljava.lang.Object;[Ljava.lang.Object;I)Ljava.lang.Object;(Unknown Source)
     [wlst] at org.python.core.PyReflectedFunction.__call__(PyReflectedFunction.java:160)
[wlst] at org.python.core.PyMethod.__call__(PyMethod.java:96)
[wlst] at org.python.core.PyObject.__call__(PyObject.java:270)
[wlst] at org.python.core.PyInstance.invoke(PyInstance.java:261)
[wlst] at org.python.pycode._pyx5.writeDomain$14(/tmp/WLSTOfflineIni7092.py:70)
[wlst] at org.python.pycode._pyx5.call_function(ILorg.python.core.PyFrame;)Lorg.python.core.PyObject;(/tmp/WLSTOfflineIni7092.py:???)
[wlst] at org.python.core.PyTableCode.call(PyTableCode.java:208)
[wlst] at org.python.core.PyTableCode.call(PyTableCode.java:267)
[wlst] at org.python.core.PyFunction.__call__(PyFunction.java:172)
[wlst] at org.python.pycode._pyx15.f$0(/home/xxxxx/projects/tromskraft/customerportal/devenv/portal/build/filtered.py:112)
[wlst] at org.python.pycode._pyx15.call_function(ILorg.python.core.PyFrame;)Lorg.python.core.PyObject;(/home/xxxx/projects/tromskraft/customerportal/devenv/portal/build/filtered.py:???)
[wlst] at org.python.core.PyTableCode.call(PyTableCode.java:208)
[wlst] at org.python.core.PyCode.call(PyCode.java:14)
[wlst] at org.python.core.Py.runCode(Py.java:1135)
[wlst] at org.python.core.__builtin__.execfile_flags(__builtin__.java:308)
[wlst] at org.python.util.PythonInterpreter.execfile(PythonInterpreter.java:158)
[wlst] at weblogic.management.scripting.utils.WLSTInterpreter.execfile(WLSTInterpreter.java:305)
[wlst] at weblogic.ant.taskdefs.management.WLSTTask.execute(WLSTTask.java:108)
[wlst] at org.apache.tools.ant.UnknownElement.execute(UnknownElement.java:275)
[wlst] at org.apache.tools.ant.Task.perform(Task.java:364)
[wlst] at org.apache.tools.ant.Target.execute(Target.java:341)
[wlst] at org.apache.tools.ant.Target.performTasks(Target.java:369)
[wlst] at org.apache.tools.ant.Project.executeTarget(Project.java:1214)
Trond Andersen, Invenia AS, http://www.invenia.no

Here how I did it.
1. Read the basic domain template:
<pre>
readTemplate(r'@bea.home@/weblogic92/common/templates/domains/wls.jar')
</pre>
2. Set domain name and possibly username and password. Also alter the server name if there is a new for it.
3. Write the domain to the file system:
<pre>
setOption('OverwriteDomain', 'true')
writeDomain(r'@domain.placement@')
closeTemplate()
</pre>
4. Read the newly written domain
<pre>
readDomain('@domain.placement@')
setOption('ReplaceDuplicates','false')
</pre>
5. Add the required templates. Here is how we did it for the portal setup:
<pre>
addTemplate(r'@bea.home@/weblogic92/common/templates/applications/workshop_wl.jar')
addTemplate(r'@bea.home@/weblogic92/common/templates/applications/p13n.jar')
addTemplate(r'@bea.home@/weblogic92/common/templates/applications/wlp.jar')
</pre>
6. Do all the changes to the portal configuration like setting the portal data sources, jms configuration etc.
7. Update the domain to file:
<pre>
updateDomain()
closeDomain()
</pre>
done!!!
Trond Andersen, Invenia AS, http://www.invenia.no

DX80 making Audio Calls Only

I have just recently acquired a DX80 to test in our environment. I was able to get it registered to CUCM 10.5 and place calls with it. The problem is though that it will only make Audio calls. I cant seem to get it to make any video calls.
All the other devices (SX20s) registered on CUCM can make video and audio calls without any issues.
I have installed the device pack.
cmterm-devicepack10.5.2.12014-1.cop.sgn
and the device
cmterm-dx80.10-2-3-33.cop.sgn
Running with:
CUCM 10.5
VCS 8.5.1
Any suggestions?

Thanks for the response !!
I have put these two all by themselves on a small network and am trying back and forth between the two with no other influence . Yes , they are acting just like a unit without multisite that is already in an active call.
They are both set to 4 meg and they are the only devices on that network. There is one caveat and I havnt mentioned it because I have done it a thousand times and it never mattered . I am not using the Cisco camera with camera control and I do have the error for no camera attached.
I have seen this tons of times in the past and it didnt matter at all. I usually hook a high end camera into the HD/SDI input for what I use these for.
These are older "Tanberg" branded codecs tho and I am wondering if it puts the units into a state where they cant make a video call ? I looked for some API command to disable such a thing if it existed and found nothing. Im dying here.

Linux X11 Login and ARD

Disclaimer -
I have tried getting this answered else where and everyone seems to misunderstand the question. I've tried to provide a decent overview to get help.
I'd like to be able to connect to a Mac using my windows machine. I don't know all the right terms, so I'll give a few examples.
- I can use my windows machine to ssh to a linux box. If I use ssh and have an xserver/xterm? running, I can run gui apps while logged into the linux machine. So if I do this from a command line -
ssh -X [email protected]
I login and get a shell prompt. If I enter nedit or kate on the command line, it will start the application up and give me the gui.
- Using some voodoo magic of cygwin, which I don't understand, I can start a session?? on my windows machine and it looks just like I'm sitting in front of the linux machine. Other people are logged in remotely the same way or using ssh from above. I get a regular gui login window and have a gui desktop as well. This is in essence what I want to do to a mac. Here is the script I run to make this happen -
SET CYGWIN_ROOT=c:\cygwin
SET PATH=.;%CYGWINROOT%bin;%CYGWINROOT%\usr\X11R6\bin;%PATH%
SET XAPPLRESDIR=/usr/X11R6/lib/X11/app-defaults
SET XCMSDB=/usr/X11R6/lib/X11/Xcms.txt
SET XKEYSYMDB=/usr/X11R6/lib/X11/XKeysymDB
SET XNLSPATH=/usr/X11R6/lib/X11/locale
c:\cygwin\usr\X11R6\bin\Xwin :1 -query remotecomputer.domain.place.com
I didn't write it, I just use it.
- I've used VLC, but that only lets me control the active account, not login as above. I know VLC isn't my solution.
- I looked into Apple Remote Desktop, but the Apple site shows all kinds of things you can do to administer the machine. I'm not interested in doing all that. You can enable VLC support under the preferences, but again, thats not what I really need.
Is there a way to do what I'm trying? Does anyone know the name of what I'm doing when I run that script for cygwin?
Someone said I can do what I want with Linux because its using X11. Since Apple uses Quartz, its not possible. Is this possible using OSX Server?
Thanks,
Aaron R>

Got info else where.

UCS Central 1.1.1a, LDAP, UCSM 2.1

Hi All,
I have couple of questions regarding UCS Central deployment.
Customer has mutiple domains with variety of OS being installed per UCSM domain (VMWare, Redhat, Windows, Hyper-V, Oracle) .
1. Should I go ahead and create mutiple Domain Groups in UCS Central? Add each of the domain to respective DG later on? Or Should I just add all the UCSM instances to Org-root DG?
2. Customer wants to deploy LDAP and since there are mutiple UCSM domains here, VMWare Admin should have access only to VMWare blades same follows to rest of the OS types. Does LDAP works with UCS Central 1.1.1a, UCSM 2.1? I have tested on UCS Central 1.1.1a, for some reason the third tab (Domain) doesn't list when I launch UCS Central. Am I missing anything here?
Thanks a lot in advance.
Best Regards,

1. Creating DG's can be done at anytime. And domains can be moved between DG's subsequent to the initial installation --- though you should be aware that doing so may have service impact, since you are potentially changing policy resolution for multiple domains. As a best practice, you do not generally want domains directly under the root DG --- use other DG's (or sub DG's). Another best practice to minimize risk is to minimize domain movement between DG's. So I'd recommend DG creation and domain placement initially.
2. LDAP works on UCSM 2.1 --- but there are some known limitations with UCS Central 1.1.1a. Best Practice is to not put anything under the root DG directly --- use other DG's (or sub DG's). Especially necessary for LDAP. Never put LDAP configs under the root DG. (Not clear what the "third tab(Domain)" is referring to. Can you please post a screenshot or detail what you're expecting?)

Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP

Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#

Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows.

How to change host name and domain name in installed Cisco Meeting Place Express 2.0?

Hi
I have Cisco Meeting Place Express 2.0 installed on server. Meting Place uses Linux (Red Hat) as base operation system. Please tell me how I can change host name and domain name for Meeting Place Express application to integrate it into corporate domain infrastructure.
Thanks in advance!!!
Sincerely yours
Sergey Bondarenko

Hi
Thanks for your attention for my humble problem.
I have performed Meeting Place Express configuration through the “net” command.
But now unfortunately I can't connect to meeting place through web browser. I just get blank page.
Though I can reach the server with help of ssh. I think my DNS works properly because I can resolve the ip address of cumpxvoice .uclab.com and cumpxweb. uclab.com with help of dig command.
Have you any idea?
A lot of thanks in advance!!
Sincerely yours Sergey Bondarenko
Here I have printed the net configuration output of Cisco meeting Place Express:
[root@cumpxvoice root]# net
This is the MeetingPlace Express network configuration utility.
Up to the point where you save changes, it is safe to hit ^C to get
out at any time, in which case nothing will have changed.
1) List current configuration
2) Configure Ethernet port 1
3) Configure Ethernet port 2
4) Configure service bindings
5) Set host names
6) Set domain name
7) Configure DNS service
8) Configure NTP service
9) Configure routing
10) Done
Select: 1
Primary/audio host name: cumpxvoice
HTTP host name:          cumpxvoice
RTMP host name:          cumpxweb
Domain name:             uclab.com
Service bindings:
Audio:                   port 1 (eth0)
Web browsing (HTTP):     port 1 (eth0)
Web conferencing (RTMP): port 2 (eth1)
Port 1 (device eth0):
IP Address:      10.62.60.116
Network mask:    255.255.255.0
Default gateway: 10.62.60.1
Link setting:    autoneg on
Port 2 (device eth1):
IP Address:      10.62.60.117
Network mask:    255.255.255.0
Default gateway: 10.62.60.1
Link setting:    autoneg on
Static routes:
Destination      Gateway          Network mask     Port
0.0.0.0          10.62.60.1       0.0.0.0          1
DNS Servers:
10.62.60.115
10.0.1.4

Need Urgent Help on Meeting Place Integration with CUCM 7.1 and AS5400 PSTN Gateway

Hi,
This is first time I am on this forum.
I have already tried going through a lot of docs on docwiki.cisco.com but couldn't find complete configuration help.
I have to integrate Meeting Place 8.X with an existing CUCM and an E1 gateway (PSTN Gateway) AS5400.
The CUCM is already part of a Telepresence Environment. I need to create a SIP trunk between AS5400 and CUCM 7.1 and then create a Trunk between AS5400 and Cisco Unified MP 8.X and then between CUMP and CUCM.
I need help on AS5400 SIP Configs as well as parameters I need to cover on CUCM (Though I have done some basic dial-peer configs but they haven't been of much help).
Then I also need help on AS5400 SIP configs with CUMP 8.0
Any docs on Integration between CUMP and TP3000 will be of great help too.
Rgds,
Asim

I can get the Ricoh to register as sip endpoint, it answeres then imediatly disconnects. Doing a monitor with Wireshark looks like it attempts to negotiate t38 but fails. Any idea why this fails?
|160.260684000|         INVITE SDP (g711U)            |                   |SIP From:
|         |(5060)   ------------------> (5060)   |                   |
|160.338806000|         INVITE SDP (t38)              |                   |SIP Request
|         |(5060)   <------------------ (63435) |                   |
|160.339545000|         491 Request Pending           |                   |SIP Status
|         |(5060)   ------------------> (5060)   |                   |
|160.547894000|         406 Not Acceptable            |                   |SIP Status
|         |(5060)   <------------------ (63435) |                   |

CUCM 7.0 Domain Placement

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