There is a little restriction for SIP Notify DTMF for CUCM.
Juan is correct - You need to enable the accept unsollicited notify in
sip security profile so that CUCM will be able to receive Notify DMTFs.
But if MTP is checked, then the Notify option will not work.
Also for outbound Notify DTMF
Remove the Authentication Credentials. They are not required for VoiceView.
On Fri, Mar 2, 2012 at 5:52 PM, Baktha Muralidharan muralic...@gmail.comwrote:
Hi folks
I configured voiceview express on CME (Branch2 router). here is my config
on CME-
url services http://CUE http://%3ccue/
Well, as I remember, the Scipts must not be saved in Repository from Script
Editor.
When you make upload of scripts from UCCX Web interface, it makes the
script saving in repository.
On Thu, Mar 1, 2012 at 4:57 PM, Baktha Muralidharan muralic...@gmail.comwrote:
Hi folks
When I try to save
FXS is used to connect Analog Phones, not the PSTN.
For PSTN connectivity you need FXO Card.
On Sat, Feb 25, 2012 at 5:52 PM, Steven
forum.ccie.onlinestudyl...@nocer.net wrote:
Hi group,
i'm playing with FXS ports right now and encountered a strange behaviour.
First the call-flow to get
My condolences to his friends and family
On Sat, Feb 25, 2012 at 7:59 AM, Emanuel Damasceno aedamasc...@gmail.comwrote:
Hello brothers,
I just like to let you know that my friend and study partner, Jeferson
Guardia CCIE #28157, has passed away yesterday.
He was a skateboarder and day
It should appear as registered, as it depends on the IPMA and CallManager
service, where the IPMA service (Route Point) registeres with CUCM Server
on either Publisher or Subscriber (based on CM Groups).
CTI Route Point registration of IPMA does not depend on the End user
configuration for Manager
Send us the GK configuration. along with CUBE as well.
On Tue, Feb 21, 2012 at 12:30 AM, Chevy chevy.man...@gmail.com wrote:
Anyone ever seen this message before?
Gateway CUBE failed to register with Gatekeeper VIA-ZONE even after 2
retries
___
Maybe you are using H323 gateways, and + sign is added on the gateways?
On Tue, Feb 21, 2012 at 12:24 PM, Guoming Zhang guozhang20...@yahoo.comwrote:
Hi,
When I am doing Vol 1 Lab 5A 5.2 question, the calling number from both HQ
and BR1 to PSTN should be in full E.164 format such as
?
It seems to me not since usage of 'service app-b-acd-aa' command does not
allow us to determine a different service name like flash based BACD
'service aa flash://', where 'aa' is name of the service.
Peter
- Original Message -
From: datucha123 datucha123 datucha...@gmail.com
It is also possible to set the T3 (Reserved) threshold to something else
then 100%.
I do not know whether this goes for this particular case, but in general,
Reserved Threshold can be set to other then 100%.
T3 is non-configurable for Ingress SRR Queues, where is it always set to
100%.
But for
Threshold 3 is kind of the percentage of buffer contributed by this queue
to the overall buffer space and threshold 4 specifies the 3rd threshold
for the queue in question.
Reserved Threshold is considered for Egress Queues as Threshold 3 (T3). Not
the M (maximum) threshold.
On Mon, Feb 20, 2012
Are you using VMWare for you Unity Server?
If so then that behavour is observed when the Unity Connection is installed
in virtual environment.
I am also running to such issue some times, but the reboot of Unity server
helps me.
On Sun, Feb 19, 2012 at 4:16 AM, chase mergenthal
1. Are prompts also embedded in the IOS? Or do they need to be copied
in the router’s flash?
No, the Prompts are not embedded in the IOS, you need to manually add them
into Flash.
2. Does drop through mode work with embedded BACD?
Yes, embedded BACD works for Drop Through Mode very
I have tested, and Multicast MoH is supported in SRST Mode for SCCP IP
Phones, I can hear the Music.
On Tue, Dec 13, 2011 at 2:54 PM, datucha123 datucha123 datucha...@gmail.com
wrote:
I have not fixed it, I have still the ToH for SRST SCCP IP Phones.
It is not possible to get the MoH for SCCP
That is because the MVA number does not match 5999 configured in CUCM.
please make sure that 5999 is also configured as MVA access number in CUCM,
and no Called Party transformation takes place from Incoming POTS dial-peer
up to CUCM.
On Sun, Feb 19, 2012 at 11:17 PM, Emanuel Damasceno
RSVP may not work over True MLPP (I do not know exactly).
But for MLP Over Frame Relay it will work.
On Sat, Feb 18, 2012 at 6:58 PM, Radhesh Naik radheshn...@gmail.com wrote:
Hi,
** **
Came across this statement under SRND.
** **
“RSVP is currently not available on Bundle
No, as I remember, Paging is not supported on SIP Phones, at least in
version 7.0
On Fri, Feb 17, 2012 at 6:33 AM, chase mergenthal cm3_...@hotmail.comwrote:
Is paging supported on SIP phones?
Got it to work on my SCCP phones just fine...
-Chase
Interesting question.
I think that Phones Built in Bridge does support G729 and other codecs,
which are supported by IP Phone.
On Wed, Feb 15, 2012 at 10:30 AM, Juan Lopez lopez.hernandez.j...@gmail.com
wrote:
can anyone confirm the phone built-in bridge does support g729? I'm
running it in
George,
Flash MMoH cannot be traversed through L3.
So that the MMoH is not supported for remote IP Phones on CUCME.
CUCME Admin guide says that. Unfortunately I do not remember the exact
page, but its there. Also I have tested that and it is true, the MMoH From
flash does not go to remote IP
just fine.
VGW - FW - L3 Switch - Phones.
Works just fine...
I will do some wireshark traces and show the packets.
On Wed, Feb 15, 2012 at 10:53 AM, datucha123 datucha123
datucha...@gmail.com wrote:
George,
Flash MMoH cannot be traversed through L3.
So that the MMoH is not supported
:-)
If you can't make it work, that's something wrong with your configuration,
but don't say it won't work! :-)))
And please tell me where in CUCME admin guide it says such thing I'll
eat my hat if you show me the section!
Cheers,
On Wed, Feb 15, 2012 at 1:26 PM, datucha123
:26 PM, datucha123 datucha123
datucha...@gmail.com wrote:
You will not be able to hear the Music itslef.
In my testing the Multicast was also traversed to my Remote Branch Router
- I was able to see the Multicast stream and endpoints joined there with
show ip mroute and show ip igmp groups
You can also configure the Privacy Settings globally, at Telephony-service
configuration. with no privacy command, so that you will not need to
disable it per ephone.
On Wed, Feb 15, 2012 at 11:10 PM, Vik Malhi vma...@ipexpert.com wrote:
I don't see the restriction in OWLE lab #2.
The HA
Well, I think you have to add contacts to you CUPC client.
But here are some restrictions:
1) When you add Contacts from CUPC client directly, those are just
contacts, and they do not support Presence.
2) To support Presence (Chat also will be supported) you need to add
contacts from End User
Zero and One are already pre-recorded (by default) in UCCX, so you can use
them.
On Sun, Feb 12, 2012 at 12:09 AM, Randall Crumm rrcr...@yahoo.com wrote:
HI,
I know the steps I need to do but I am unsure of the prompts I need to
record.
Please let know what I need to record.
I believe I
Hello,
Do we need to set the NTP Server for UCCX Win Server? Or the NTP settings
during the Initial Setup is enough? Well, at least for the exam.
I cannot set the NTP Server for Win2003 (UCCX Server) - net time
command does not work, as it is looking for Win Server NTP, and is not
conencting
Prompt - Create Generated Prompt, where you have to point the
String which you want to play out (in your case it would be String with
value of 1 or 2 or anything you like to announce/play out.
You have to select the Constructor Type as well, as it rules how the Number
is played out
Hello,
Assume that we have G711 and G729 MTP's in first MRG, and then Xcoder in
the next MRG:
MRGL oder:
1) MRG_MTP
2) MRG_Xcoder
So as we know, CUCM can allocate the required MTP's based on codec - if
call is going to be g711 then the G711 MTP will be allocated, and if the
call is going to
As we know, the Multicast MoH is not counted against the CAC bandwidth, but
the Priority Queue does. So when the Branch devices (IP Phones/Gateway) are
using the MMoH sourced from the HQ CUCM Servers, we need to take the MoH
bandwidth into account for LLQ Priority queue, but not for CAC, as
That is the Browser issue.
On Fri, Feb 10, 2012 at 5:27 AM, Jeferson Guardia jefers...@gmail.comwrote:
Hi,
Often when doing labs over ucm 7, sometimes I notice specially when
messing around with CSS/partitions, if you open multiple windows and for
example:
You have 5 firefox tabs open
, 2012, at 9:24 AM, datucha123 datucha123 wrote:
I found the problem, but another one has arise.
I am using RSVP as CAC between site, where the RSVP MTP has only G729
codec enabled. And also the MoH is using Unicast so that it is subject to
CAC.
So now when the call is picked up by Mobile
Al Baqari
Sent from my iPhone
On Feb 9, 2012, at 9:52 PM, datucha123 datucha123 datucha...@gmail.com
wrote:
Here is what I get on HQ Router during the call hung up on the Mobile
Phone:
Feb 10 03:48:06.063: %ISDN-6-DISCONNECT: Interface Serial0/2/0:0
disconnected from 206501 , call
the region relation between HQ MTP RSPV and MoH. It should
be g729. Also same between BR1 MTP RSVP and HQ MTP RSVP.
If this is done then MoH should negotiate g729 with BR1.
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Feb 9, 2012, at 9:52 PM, datucha123 datucha123 datucha...@gmail.com
wrote
.
Vik Malhi – CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Feb 9, 2012, at 9:24 AM, datucha123 datucha123 wrote:
I found the problem, but another one has arise.
I am using RSVP as CAC
:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
datucha123
*Sent:* Wednesday, February 08, 2012 11:28 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] MVA problem
** **
I have the following kind of probem:
I am using SLRG for Mobile Connect calls
-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
datucha123
*Sent:* Wednesday, February 08, 2012 11:28 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] MVA problem
** **
I have the following kind of probem:
I
420
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Feb 9, 2012, at 8:52 AM, datucha123 datucha123 wrote:
G729 Codec is enable in IP Voice Media Streaming Application Service
Parameter. And when HQ Phone calls BR1 Phone and places BR1 on hold, the
MoH is played with G729 to BR1 phone
Pool is the MOH server in? Place it inside the HQ Device
Pool / HQ Region.
Vik Malhi – CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto: vma...@ipexpert.com
On Feb 9, 2012, at 9:27 AM, datucha123 datucha123 wrote:
Also as Vik
debug mgcp packets
Trace in CUCM
On Wed, Feb 8, 2012 at 3:35 PM, Jeferson Guardia jefers...@gmail.comwrote:
Hi,
What are the techniques most used to perform MGCP troubleshooting?
Yesterday I was doing a lab and had a router with pstn integration, it was
set for a CSS where my phones had
I have the following kind of probem:
I am using SLRG for Mobile Connect calls, so that that calls to users
mobiles are done through local gateway (this is just for test).
Now, the HQ phone has the RDP assinged with RD of his mobile phone.
Now when the BR1 phone calls this HQ phone, so that the
You are right Emanuel.
I am a native neglish speaker, and sometimes it is hard for me to
understand some task, while rereading them several times
On Tue, Feb 7, 2012 at 8:12 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:
Hey Vega,
Sorry for hearing you didn't make it. I've been there and
But that won't go for HQ router, as it is a Hub Router -- multiple PVCs
per physical interfaces via subinterfaces.
For instance, if you have a T1 frame relay link, and already shaped one PVC
to 500kbps, you cannot shape another PVC to 1536. You have to shape another
PVC to 1536 - 500 = 1036
Outvia is more accurate.
Invia, in most cases, is used for incoming LRQs.
On Tue, Feb 7, 2012 at 11:13 PM, mercy forall mercy_for_...@hotmail.comwrote:
Hi All
now in outvia and invia ,,
Are is it deference if i use it in local zone or remote zone ?
As per Doc, outvia for any traffic
the
following CLI command: show timezone list.
Regards,
Bill
On Feb 6, 2012, at 11:20 AM, datucha123 datucha123 wrote:
Hello,
If we are told to synchronize CUCM Pub server with some NTP, do we need to
set the correct Timezone for CUCM OS as well? Or just Date/Time Groups
stop routing on unallocated number flag - in this particular case, this
parameter has nothing to do with the actual problem. This parameter defines
the rerouting option as William has already mentioned.
Ricardo, try to set the Digit Analysis Complexity to Translation and
Alternate Pattern
You can also leave the Urgent Priority for Patterns, but use the SIP
Dial-rules for SIP phones, with \+! and interdigit timeout with 0
On Sun, Feb 5, 2012 at 12:53 AM, Ashwani ash_r...@hotmail.com wrote:
Thanks Vik. Yes now I am seeing inter-digit timeout dialing from missed
and received
I have the same issue in my own LAB, and as soon as I restart my
CUC server, the MWI and Message start to work from PSTN for a while. but
then again stops.
And I make restart of CUC server every time.
Thus I was using SCCP integration.
On Sun, Feb 5, 2012 at 1:00 AM, Edgar Feliz
You have press 44 (twise digit 4) and it will dial out.
On Sun, Feb 5, 2012 at 1:50 AM, John McGaughey (jomcgaug)
jomcg...@cisco.com wrote:
Hello,
** **
I’m in lab 4 of the new 5 labs. Question 6.4. It’s asking to configure
Live Reply. The DSG says to just click the check box
No. it is supported. The destination Phone will just ring a bit later
through PSTN.
On Sun, Feb 5, 2012 at 3:16 AM, John McGaughey (jomcgaug)
jomcg...@cisco.com wrote:
Hi Vik/All
** **
I’m working on Lab #4 of the new 5 labs. Quesiton 9.2. They are asking
you to configure CFUR on
First of all check if the IP Phones are put in correct Locations.
Then for AAR to work, you need to create an AAR Group and assing it to
endpoints.
On Sun, Feb 5, 2012 at 11:03 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:
Hello Experts,
I am trying to set up an AAR scenario for my
What are you trying to achieve?
Write it in more details, and we will try to help you
On Sat, Feb 4, 2012 at 7:19 AM, Seifeddine Tlili
seifeddine.tl...@lvs1.comwrote:
Hi Everyone
** **
Is there a different workground then creating PT/CSS/RP for Callfowarding
when using SLRG with
Yes, you can.
You have to use the DHCP Binding file (put it in flash), so that if the
Client identifier is found in the file, that it will give the static IP,
otherwise it will give the IP addres out of the pool.
Or you can use the DHCP Classes.
You can read about them in DHCP admin guide.
you want to say that we DO NOT NEED tftp on CME for our Phones on the LAB
Exam?
On Fri, Feb 3, 2012 at 10:45 PM, Vik Malhi vma...@ipexpert.com wrote:
Correct.
In addition you should be careful NOT to erase factory defaults on the CME
phone too- this would mean you need the load command in
for just the audio, not including any
headers...
If anyone sees it differently please let me know, i would hate to walk
into the test and have this whole concept wrong...
On Wed, Feb 1, 2012 at 2:05 PM, datucha123 datucha123
datucha...@gmail.com wrote:
G711 64kbps and G729 8kbps
That is because of IOS.
IOS detects the US Dialplan, and sets the Types accordingly in H323
gateway.
It is not possible to disable that feature. So you have to use voice
translation rules to change the ANI Type.
On Wed, Feb 1, 2012 at 2:01 PM, Juan Lopez
lopez.hernandez.j...@gmail.comwrote:
Hi
G711 64kbps and G729 8kbps are not the L3 Bandwidth for those codecs. These
are the Payload Bandwidths for those codecs.
G711 and G729 on L3 are using 80 and 24 kbps respectavely.
Here is the easy way to calcualte the Codec bandwidth:
(Payload_Size + L3_Header + L2_Header) x PPS x 8
Where
Congratulations.
On Thu, Jan 26, 2012 at 10:05 PM, Bill Lake whl...@gmail.com wrote:
Congratulations, take a well earned break before your work picks up and
your busier than those of us studying.
On Thu, Jan 26, 2012 at 10:11 AM, Thomas Koch koch1...@comcast.netwrote:
If you configure ccm-manager config, it will try to configure all 24/32 PRI
ports, and if you are not using full PRI, then you have to manually disable
unused PRI channels, and also configure the B channel maintenance in CUCM,
so that after every reload the Router will not try to configure all PRI
I have tested, and Telecaster user does not need any special User Groups
(except Standard CTI Enabled).
I can tell, that even the Telecaster User is not necessary at all.
On Tue, Jan 24, 2012 at 1:28 PM, Anthony Alba ascanio.al...@gmail.comwrote:
The script populates the variable at runtime
I have the same issue,
That's because of VMware - well I got to this conclusion.
On Mon, Jan 23, 2012 at 6:24 AM, Jurassic Labs jurassicl...@gmail.comwrote:
I've noticed this lately while loading up the newer 5-lab self study
vRacks sessions. Once a phone (say Ext 2002) is defined in Unity
Check the Inhibit restarts at PRI initialization on MGCP gateway in
CUCM configuration page.
That might help
On Mon, Jan 23, 2012 at 6:31 AM, Jeferson Guardia jefers...@gmail.comwrote:
Getting that weird message with my PRI every 30 seconds, I googled, found
people having the same issue but I
I cannot understand, how the Agent (IPPA or CAD) can show manually
configured Varibales values, until the call comes to Agents phone.
You can use the Get Reporting Statistics Step to gather info about the
Queue - But I do not know, how to pass that variables to Agents layout in
real time - the
you everyday I feel the need to tell you , man please don't flood us
like this !!
Have a good day ! ;)
Nic
Le 21 janv. 2012 à 13:33, datucha123 datucha123 datucha...@gmail.com a
écrit :
Sorry I make a little mistake here in calculation (at 30ms Sampling
Rate G729 needs 40kbps for RSVP
Hello,
What is the difference between these two commands:
no h225 timeout keepalive
call preserve limit-media-detection
Well, both commands (separately) can be used for SRST, when the CUCM goes
down, and active calls should be left active.
Also I have read some documentation, and the final
Hello,
When we are using default sampling rate for codecs, for example G729 of
20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
of all calls:
10 G729 calls - 24 x 10 + 16 = 256 kbps
But when using non-default Sampling Rate for codecs, for instance G729 at
30ms, then
PM, datucha123 datucha123 datucha...@gmail.com
wrote:
Hello,
When we are using default sampling rate for codecs, for example G729 of
20ms, then during the RSVP CAC caluclation we need to add 16 bkps for sum
of all calls:
10 G729 calls - 24 x 10 + 16 = 256 kbps
But when using non
Hello,
H323 Router by default (in IOS) sets the Calling Type and Plan for outgoing
calls to PRI. - I think there is something put in IOS so that it recognizes
the US dialplan, and set the Type and Plan automatically.
How can I disable it?
Because when I set the Type and Plan in Callmanager, and
please correct if wrong
G711 and G729 variants (g722, iLBC are not part for this email)
Cisco IP phones default codecs - G711Ulaw, and G729ar8.
Cisco IP phone supported codecs (SCCP) - G711u, G711a, G729r8, G729ar8,
G729br8 and G729abr8
Cisco IP Phone supported codecs (SIP) - almost the
try to reload the Router.
On Wed, Jan 18, 2012 at 3:36 PM, The Masterplan winmasterp...@gmail.comwrote:
The same thing. The only difference is that now I don't see anymore the
phone in show run.
On Wed, Jan 18, 2012 at 12:34 PM, Mohd Baqari baqari.voic...@gmail.comwrote:
Ok ... Try
well, I do not know the very exact version of CUCM on the LAB Exam, but I
think it is better to study with the one that proctollabs offer. After you
will know the bugs and issues for that UCM/
On Wed, Jan 18, 2012 at 9:03 PM, Juan Lopez
lopez.hernandez.j...@gmail.comwrote:
all,
I found the
Also as I remember, the IPIPGW tech prefix has to match the Destination
Zone prefix.
On Wed, Jan 18, 2012 at 9:39 PM, Steven
forum.ccie.onlinestudyl...@nocer.net wrote:
@Boris
@Leslie
@Amit
I got some other issues too.
I skipped to check the GK-only functionality (BIG mistake).
After i
Also do not forget to configure timer trying - 150/200 ms is OK
On Wed, Jan 18, 2012 at 12:46 AM, Juan Lopez lopez.hernandez.j...@gmail.com
wrote:
When configuring 2 SIP dialpeers for redundancy, together with:
sip-ua
retry invite 2
This should generate in total 3 INVITES sent to the
.
Cheers,
On Mon, Jan 16, 2012 at 4:01 PM, George Goglidze gogli...@gmail.comwrote:
correctisimo :-)
On Mon, Jan 16, 2012 at 3:25 PM, datucha123 datucha123
datucha...@gmail.com wrote:
When configuring the simple BLF Speed Dial, we need to configure the
Subscribe CSS for watching Device
Also check the DB Replication Status
On Tue, Jan 17, 2012 at 6:17 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hello Errol ,
This issue usually mean you have incorrect Host/ processNode files in
your PUB or the Sub and so the communication is broken between the PUB
and the SUB and you have
Hello,
Based on my testings, CUPC support Presence Status change in its Contacts
only when they are imported from AD.
Manually created contacts does not support Presence Status Change.
Is it correct?
___
For more information regarding industry leading
That parameter actually does not effect anything.
On Tue, Jan 17, 2012 at 9:22 PM, Hough, Earl
earl.ho...@pcmallservices.comwrote:
I guess it would determine what your requirements are. If, for example,
the global requirements were that your subscriber were to be the primary
server for all
You can set it to any server you like initially, but when you go to Server
configuration in Port Group page, there you have to set them as necessary
and required.
On Wed, Jan 18, 2012 at 12:18 AM, datucha123 datucha123
datucha...@gmail.com wrote:
That parameter actually does not effect
First of all ensure that the Authentication URL is set correctly.
Also you can try to reset the CUE module - just in case.
On Tue, Jan 17, 2012 at 6:30 PM, study buddy studybudd...@gmail.com wrote:
Hi
While accessing my voicemail via VoiceView, I get the following error when
I click on
, please edit your Subject line so it is more specific
than Re: Contents of CCIE_Voice digest...
Today's Topics:
1. Re: AAR for UCCX (Mohammed Al Baqari)
2. BAT csv template for CUC 7 (donny f)
3. MGCP Registration (mercy forall)
4. Re: AAR for UCCX (datucha123 datucha123)
5. Re: MGCP
When configuring the simple BLF Speed Dial, we need to configure the
Subscribe CSS for watching Device. So that it could the the Watched DN.
But, the Owner User ID, Line Association with End User and other kinds of
associations are not required for this BLF, right?
Even for Call List Presence.
route points as the origin or the destination
of calls. Also, AAR is incompatible with the Extension Mobility feature
when users roam across different sites.”
** **
Regards,
Mohammed Al Baqari
** **
*From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
*Sent:* Sunday
first of all make sure that the L2 is up for the PRI, otherwise the MGCP
gateway in CUCM will show unregistere status.
On Sun, Jan 15, 2012 at 12:16 PM, mercy forall mercy_for_...@hotmail.comwrote:
Hi,
i have 3845 with VWIC-1MFT-G703
i configured mgcp but it is bending in registering on
Calling Number will be visible to all other Sharing users, until you answer
the call. As soon as you will pickup the call, the other IP phone users wil
not be able to see active call on you phone any more.
On Sun, Jan 15, 2012 at 5:49 PM, Ken Wyan kew...@gmail.com wrote:
According to cisco docs
First of all you can look at the CUCM Traces filtered by particular Phone.
Go to Trace, then enable Device Based Tracing and choose the necessary
Phone/device
After ssh into CUCM Server and issue the command file tail activelog
/cm/trace/ccm/sdi recent, this will give you the real time traces.
For incoming calls, the 23548 is more specific match for Called Number of
235482345 then 235. And that is why the Dial-peer 2 is matched.
For outgoing calls, if you place a call to the same number (235482345) and
the destination patterns are the same (235 and 23548) then the dial-peer 2
would be
Hello,
Please correct me if I am wrong.
CUE has three types for MWI notification:
1) Outcall
2) Sub-notify (Subscribe)
3) Unsolicited
CME does support all three MWI methods:
When using Subscribe option in CME, we have to configure the mwi sip
commands on SCCP Ephone-DNs, and MWI on SIP Voice
at 4:31 PM, Emanuel Damasceno aedamasc...@gmail.comwrote:
Wouldn't the command preference X work in that situation?
*Emanuel Damasceno*
CCNP Voice
On Fri, Jan 13, 2012 at 6:36 AM, datucha123 datucha123
datucha...@gmail.com wrote:
For incoming calls, the 23548 is more specific match
That is a great question.
Based on my knowledge, the Set DSCP command is executed first, because
otherwise the exceeding traffic will become EF.
You can also refer to this Linke, where the Auto QoS is duscussed:
http://ieoc.com/forums/t/17680.aspx?PageIndex=1
On Fri, Jan 13, 2012 at 5:47 PM,
, datucha123 datucha123
datucha...@gmail.com wrote:
First of all Preference command works only for Outgoing calls. It
does
not make any sense for incoming dial-peer matching.
also in that particular case, the preference command will not make
any
sense, because those dial
Hello,
Do we need to configure the CUCME-SRST (or call-manager-fallback) as the
Third Callmanager in JTAPI Subsystem configuration in CUE for SRST fallback?
I think we do not need to configure the CUCME-SRST as the Third
Callmanager, as CUCME-SRST does not use the JTAPI, but is integrated with
Different Route Lists may use the LRG
On Thu, Jan 12, 2012 at 4:26 AM, Spence, Paul paul.spe...@jacobs.comwrote:
Guys,
Need a little advise on LRG, I have it set up in a live environment and
associated with a CUCM group. We recently added a new subscriber to a
remote site and wanted to
No it is not bound to one Route List, it can be used by any Route List in
the system
On Thu, Jan 12, 2012 at 4:21 PM, Spence, Paul paul.spe...@jacobs.comwrote:
Ok that’s good to know I though it may have been bound to one RL only
Many Thanks
** **
*From:* datucha123 datucha123
I have tested today, and Boris is right. The multicast MoH can travers
through RSVP MTPs fine.
While Mutlicast does not work for H323 and SIP gateway, when they are using
MTP
On Thu, Jan 12, 2012 at 12:14 AM, datucha123 datucha123
datucha...@gmail.com wrote:
I will test that tomorrow. Because
Hello,
I have noticed, that the Phone Line Label is not passed with SCCP Phone,
when it goes to SRST.
But when the SIP Phones goes to SRST, then the Line Label is passed on over
the SRST.
Is it normal?
___
For more information regarding industry
Hello,
Can you please confirm the following?
When Using Auto Provision DN (for CME SRST), then we can manually create
the DNs, so that the CME will try to match the manually created DN with
learned DN, and if it matches, then assing to that Ephone.
When using Auto Provision ALL, then we can
Hello,
When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH should
not work for Branch 1 IP Phones from UCM Server, as the Muticast cannot
traverse the MTP, right?
But when sourcing Mutlicast MoH from the Branch Router, it will work.
.
---
Regards
Boris
On 12/01/2012, at 5:31, datucha123 datucha123 datucha...@gmail.com
wrote:
Hello,
When using RSVP CAC between HQ and Branch 1 Sites, the Multicast MoH
should not work for Branch 1 IP Phones from UCM Server, as the Muticast
cannot traverse the MTP, right?
But when sourcing
Hello,
I have a question about the CAC:
Imagine that we are asked to configure the UCM standard Locations CAC
between HQ, BR1 and BR2 Sites in the following way (this is just an example)
BR1 to BR2 - allow up to 10 calls
BR2 to HQ - allow up to 5 calls.
Well if using UCM Standard
Hello,
Can you please explain the idea of User External Phone Number Mask
checking for RP, RL, or GW/Trunk Level Calling party Transformations, when
the Translation pattern is already using that checkbox, and making the
translation?
As I know, the Translation Pattern translation takes effect
I do not have such problems in my own lab.
SIP phones Transform the ANI always.
On Tue, Jan 10, 2012 at 12:19 PM, Guoming Zhang guozhang20...@yahoo.comwrote:
Hi,
When I tried lab 5.3, it works on SCCP phones, but not on SIP phones,
though they use the same device pool with calling party
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