Hello everyone,
I have a problem with SRST to Gatekeeper calls.
When the IP Phone are registered with CUCM calls to CUCME through GK works
just fine.
But as soon as I will shutdown the CUCM Service on all servers, so that the
IP Phones will register with SRST Router, the outgoing calls to CUCME
t
ot initating H225 to CUCME.
On Mon, Nov 14, 2011 at 10:18 PM, datucha123 datucha123 <
datucha...@gmail.com> wrote:
>
> Hello everyone,
>
> I have a problem with SRST to Gatekeeper calls.
>
> When the IP Phone are registered with CUCM calls to CUCME through GK works
> just
Are you using UCCX on Win 2003? (Not Cisco OS)?
If so, then the UCCX is unable to download the JTAPI client from CUCM and
is not fully synchronized.
To fully synchronize JTAPI on UCCX server that is installed on Win 2003
from Microsoft (Not Cisco Win) you have to create the WINNT folder on C:
Af
dsp service dspfarm - enables the dsp profiles, like transcoder,
conference and etc.
no dspfarm - tells the router to separate the resources for DSP Farms and
E1/T1 interface. If the default value will be set (dspfarm) then the dsp
profiles and E1/T1 interfaces will use the same set of DSP Resou
rect calls from SRST
> registered phones to the gatekeeper? Without that you're not going to get
> very far...the SRST router has no knowledge of the callmanager gatekeeper
> configuration.
>
> -matthew
>
> On Mon, Nov 14, 2011 at 1:18 PM, datucha123 datucha123 <
> da
Hello,
Can anybody please explain the difference between the "Intercluster Trunk
(Gatekepeer-Controlled)" and "H225 Trunk (Gatekepeer-Controlled)".
And which one must be used on the LAB exam for connecting CUCM to
Gatekepeer?
___
For more information r
-- Forwarded message --
From: datucha123 datucha123
Date: Mon, Jul 11, 2011 at 12:45 AM
Subject: B2BUA mailbox
To: ccie_voice@onlinestudylist.com
Hello, everyone
Can anybody please tell me how does the b2bua mailbox command works for SIP
IP Phones registered to CUCME?
I have
be understood by a standard H.323 endpoint. Configure this
>> type of trunk only when interoperability is required with versions of Cisco
>> CallManager prior to Release 3.2.
>>
>> On Wed, Nov 16, 2011 at 12:10 PM, datucha123 datucha123 <
>> datucha...@gmail.com
As I know, there are not Analog Phones any more on the LAB. So why are you
wondering for Analog Phones?
On Wed, Nov 16, 2011 at 10:11 PM, Devakanth Gangavarapu <
devakanth2...@gmail.com> wrote:
> Hi All
>
> Can someone help me understand how the online Rack Rental procedure
> Any videos or docume
Hello everyone.
I have a question about the GK.
I could not understand how does the GK knows where to route call based on
the following GK configuration:
gatekeeper
zone local test test.com 177.1.254.1
gw-type-prefix 1#* default-technology
no shutdown
CUCME and CUCM, both are registered with
nts
> are registered in the same zone.
>
> Thanks
> --
> *From:* datucha123 datucha123
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Thursday, November 17, 2011 4:30 PM
> *Subject:* [OSL | CCIE_Voice] Gatekeeper Call Routing
>
> Hello ev
Try to configure the "*address-hiding*" and "*redirect ip2ip*" commands in
Voice Service VoIP.
On Fri, Nov 18, 2011 at 1:06 PM, Gerence Guan wrote:
> Hi Everyone,
>
> I am working on this issue for couple of days and already pulled my head
> off~
>
> The topology is like this:
>
> We have Ava
Hello everyone,
I have an issue with the Media Resources on CUCM,
So here is the problem:
CUCM is connected to CUCME through ICT trunk. G729 Codec. MTP is checked in
the Trunk.
HQ Phones are assigned the MRG with containing G729 MTP registered with the
HQ Router.
BR IP Phones (not the CUCME IP
interface FastEthernet0/1/0
switchport trunk native vlan X
switchport mode trunk
switchport voice vlan Y
spanning-tree portfast
On Fri, Nov 18, 2011 at 8:47 PM, Cecil Wilson wrote:
>
>
> Hello
> I have a 2811 with the HWIC-4ESW card . how can I access these
> ports to no shut and p
As I remember, Shared Lines between SIP and SCCP Phones in version 7.0 is
not supported.
Correct me if I am wrong please.
On Sat, Nov 19, 2011 at 3:06 PM, Prakash kumar
wrote:
> Hey ,
>
> is it possible to create a shared line between sccp and sip phone ?
>
> itried to config one .but an incoming
You need to go under "frame-relay interface-dlci xxx" and configure the
command over there.
On Sun, Nov 20, 2011 at 3:45 PM, Raees Shaikh wrote:
> Hi All,
>
> Below is the config from my lab HQ router
>
> !
> nterface Serial0/3/0
> no ip address
> encapsulation frame-relay
> clock rate 6400
or J !!! Then we
> can discuss further.
>
> ** **
>
> Regards,****
>
> Mohammed Al Baqari
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
> datucha123
> *Sent:* Friday,
hello, everyone.
I have several questions about the QoS:
1) As I know, the Frame Relay Fragment Size must be always calculated based
on the Interfaces Access Rate? Am I right? And when using the Auto QoS on
Frame Relay Subinteface, it will calculate the Framgent Size based on the
"bandwidth" comm
Yes, Sure,
Mandatory (Video Desired)
But still no success :(
On Sun, Nov 20, 2011 at 9:58 PM, Randall Crumm wrote:
> Is the setting set to mandatory under location in CUCM?
>
> Randall
>
>
> ------
> *From:* datucha123 datucha12
ool
> assigned to the GW?
>
> RC
>
> ------
> *From:* datucha123 datucha123
> *To:* Randall Crumm
> *Cc:* "ccie_voice@onlinestudylist.com"
> *Sent:* Sunday, November 20, 2011 10:00 AM
> *Subject:* Re: [OSL | CCIE_Voice] QoS fo
Do you mean under the DHCP configuration, that such question has been on
the exam?
What do you mean under the SIP early offer? or MGCP TS?
On Sun, Nov 20, 2011 at 8:24 PM, Peter Jeff wrote:
>Hi Guys,
>
> It was my 5th attempt and i got lab 6 frustration is on peak everytime i
> went for
As I know, we have to calculate the FRF.12 size based on the Access Rate,
and not per the PVC Speed.
So in this case we have to find out the actuall Access Rate of the FR link.
So we must NOT calculate the FRF.12 based on the 384, but use the actuall
FR Access Rate.
Am I right? or not?
On Mon,
I have the follwoing issue on GK:
So here is the configuration of my GK
*gatekeeper
zone local SEA ine.com 177.1.254.1
bandwidth total zone SEA 16
no shutdown*
**
*CUCM and CUCME are using g729 codec to call each other.*
CUCM and CUCME are registered with Dynamic Tech Prefixes in the same SEA
39363000
>> *' in class SCPH1 *!!!... The underlined value is the correct option
>> 60 value for SC Phone 1*
>>
>> ** **
>>
>> *STEP 4* Reconfigure the correct option 60 value for phone 1.
>>
>> ** **
>>
>> ip dhcp class SCPH1
&g
Auto QoS will also calculate the FRF.12 based on the PVC speed, and this is
not correct as I know.
On Mon, Nov 21, 2011 at 4:40 PM, Ccie Voice wrote:
> I am not sure also.
>
> but can we use auto qos voip trust?
>
> --
> *From:* datucha123 datuch
n even with a special
> Gatekeeper region that has g729 to everyone.
>
> HTH
> Chris
>
> On Mon, Nov 21, 2011 at 6:14 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I have the follwoing issue on GK:
>>
>> So here is the configuration o
Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Nov 21, 2011, at 6:25 PM, datucha123 datucha123
> wrote:
>
> I am using 7962 IP Phones, and when I start the DHCP debug command, as
> mentioned, I could not see the option 60 in DHCP debug messages for my
> phone.
>
&
I do not have problems on CUCM. It is reserving normally.
I have a problems for calls initiated by CME Router
On Mon, Nov 21, 2011 at 8:09 PM, Bill Lake wrote:
> Try changing under system paramters intraregion codec to g729
>
> On Mon, Nov 21, 2011 at 8:29 AM, datucha123 datucha123 &l
Did you make an MGCP Signaling and Media Bindings?
On Tue, Nov 22, 2011 at 2:16 AM, Peter Jeff wrote:
> MGCP TS lab (6) what we can do for one way audio explanation on debugs
>
> thanks
>
> ___
> For more information regarding industry leading CCIE La
Hello,
Is it possible in PRI ISDN to get the Alerting Name, in the LAB environment?
I mean, when the CUCM IP Phone calls CUCME IP Phone through the PRI, the
CUCM IP Phone will display the Alerting Name and Number of the CUCME IP
Phone? Just like the SIP and H323 trunks do when the CUCM is connect
Hello,
When the CUCM Cluster registers with GK, two endpoints are created in GK -
one for Publisher and another for Subscriber.
And to prioritize those CUCMs in GK, we have to use the "zone prefix xxx
gw-priority" command with appropriate values.
But how to prioritize CUCM Trunk in GK, when I am
Hello,
In some Workbooks, IPExpert and INE, there are some LABs, where the FR
Fragment size is calculated based on the PVC speed. But that is not correct
as I know, as the Fragment size must be calculated based on the actuall
Access Rate of an interface.
Could anybody explain me, why some LABs ar
Try configure "upgrade" and "file text" commands under the Voice Register
Global.
Also if your IP Phone and CUCME Routers are not in the same Broadcast
domain, then configure the Username and Password and set the "authenticate
register" command in Voice Register Global.
On Tue, Nov 22, 2011 at
Ok, but I am registering CUCM with GK, and not vise versa. So the GK has to
know the order of CUCMs.
I think that the DP will not tell the GK, which CUCMs to use as a Primary
one.
On Tue, Nov 22, 2011 at 7:24 PM, Luis Felipe Segnini Salas <
felipe_segn...@hotmail.com> wrote:
> Hello,
>
> You wi
t; *version* 2
> > !IP addressTypeHardware address Lease expiration
> > 142.102.66.21 /24 id 0100.1B54.951C.D9 Infinite
> > 142.102.66.22 /24 id 0100.1AA1.93E0.F5 Infinite
> > *end*
> >
> > BR2-RTR#sh cdp nei det
> > Devi
tween
> them ,
>
> Finally , the GK never register to anyone but the Directory GK .
>
>
> Ash
>
> On Tue, Nov 22, 2011 at 10:18 AM, datucha123 datucha123
> wrote:
> > Ok, but I am registering CUCM with GK, and not vise versa. So the GK has
> to
> > know t
bytes (which is done
> by dividing the result by 8): *
>
> *Fragment Size in Bytes = (PVC Speed in kbps * Maximum Allowed Jitter in
> ms) / 8*
>
> You don't even have to do the calculations ... table 3-1 has it done for
> you ... Page 205. Likewise, auto qos will do the same.
>
>
Sent from my iPhone
>
> On Nov 22, 2011, at 2:39 PM, datucha123 datucha123
> wrote:
>
> > Hello,
> >
> > In some Workbooks, IPExpert and INE, there are some LABs, where the FR
> Fragment size is calculated based on the PVC speed. But that is not correct
> as I
at 11:22 PM, Mohd Baqari wrote:
> You need to enable display-ie on your pri links and in cucm.
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Nov 22, 2011, at 2:34 PM, datucha123 datucha123
> wrote:
>
>
> Hello,
>
> Is it possible in PRI
r config. Also post the output of debug ccsip message
>
> Regards,
> Mohammed Al Baqari
>
> Sent from my iPhone
>
> On Nov 22, 2011, at 3:12 PM, datucha123 datucha123
> wrote:
>
> Try configure "upgrade" and "file text" commands under the Voice
&g
re I sent u email yesterday.
>
>
> Regards
> Amit
>
> Sent from my iPad
>
> On 22/11/2011, at 1:14 AM, datucha123 datucha123
> wrote:
>
> I have the follwoing issue on GK:
>
> So here is the configuration of my GK
>
> *gatekeeper
> zone local SEA
STN router have the commands to send the display-ie on the serial
> interface for the D channel?
>
>
> On Tue, Nov 22, 2011 at 3:37 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I jave enabled the Dislpay IE on PRI and in CUCM.
>>
>> But t
; with priority = 5 for all of them and the GK will load balance between
> > them ,
> >
> > Finally , the GK never register to anyone but the Directory GK .
> >
> >
> > Ash
> >
> > On Tue, Nov 22, 2011 at 10:18 AM, datucha123 datucha123
> >
Predot means that the digits before "dot" will be stripped.
On Wed, Nov 23, 2011 at 7:38 AM, Emanuel Damasceno wrote:
> Hello Experts,
>
> As I practice more and more for my CCIE exam, I came across DDI
> NANP:Predot. I have no experience whatsoever with the North American
> Numbering Plan. All m
gt;
>
>
>
> On Wed, Nov 23, 2011 at 8:38 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Predot means that the digits before "dot" will be stripped.
>>
>> On Wed, Nov 23, 2011 at 7:38 AM, Emanuel Damasceno <
>> aedamasc.
> “voice service voip
>
> qsig decode”
>
> ** **
>
> Regards,
>
> Mohammed Al Baqari
>
> ** **
>
> *From:* datucha123 datucha123 [mailto:datucha...@gmail.com]
> *Sent:* Wednesday, November 23, 2011 2:19 PM
> *To:* Abel ...
> *C
I am trying to create the Voice Mail Box for the User from the CUCM
directly, meaning that I am going into Line configuration and then
selecting the "Create Voice Mail Box" from the upper right corner.
But when I choose the User Template (it is visible in CUCM when adding the
VM Box) and click SAV
Hello,
I have configured the Voice Mail Box Mask for (As I know, this will
transform the Redricting Number to last 4 digits).
but somehow it does not work :(
So here what is happening:
I have configure the User and VM Box in CUC for CUCME IP Phone, (ext 3012).
I have configure the Call For
ld point me to
> any documents that explain this, I'd appreciate it.
>
> *Emanuel Damasceno*
>
>
>
>
>
> On Wed, Nov 23, 2011 at 9:45 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I do not know what does that mean, but do not wo
Hello,
How can I force CUCME IP Phones to display the Numbers in Missed/Received
calls directory with + sign?
When the call comes to CUCME with Caller ID with +, the CUCME IP Phone
dislpays that + sign in lower place of the screen. But in the Directories
(Missed/Received) calls, that number is li
; On Wed, Nov 23, 2011 at 11:38 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Hello,
>>
>> I have configured the Voice Mail Box Mask for (As I know, this will
>> transform the Redricting Number to last 4 digits).
>>
>> but somehow
CM, but I
> thought you had to set the application user to an axl enabled user that is
> also set in unity connection phone system. IE: admin.
>
> I may be wrong.. Has been a while since I tried that and I don't have
> access to my lab right now.
>
> Chris
>
>
IS to 4 digits, then associate
> this to a dial-peer going to voicemail.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:14 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, sure, it is enabled on the Incoming Gateway.
>>
>> I have also
I just had
> time to run through it and worked fine for me. When I had both users I got
> the same error you are.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:15 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, I have that user "admin&quo
What do you mean under the No Supplementary service?
So you mean *no supplementary-service h225-notify cid-update*?
On Thu, Nov 24, 2011 at 4:00 AM, Gurpreet Singh Kukreja <
tycoononway1...@gmail.com> wrote:
> Hi,
>
> This depends on what is asked in the Lab. If the lab tells you to display
> 10
t;>
>> Have you tried:
>> translation rule XX
>> rule 1 /^1/ /+1/
>>
>> translation-profile 1
>> translate calling 1
>>
>> voice-port x/x/x:x
>> translation-profile incoming 1
>>
>> If you did, and didn't work, I am gonna start
I think you can do both as nothing will be ruined after the synch.
On Thu, Nov 24, 2011 at 10:01 AM, Ken Wyan wrote:
> Hi Guys,
>
> I have a very basic question with manually created users in CUCM. ( *No*AD or
> LDAP integration)
>
> When integrating CUCM with CUC in the exam , is it sufficient
One option would be a
> voice-translation rule that strips the RDNIS to 4 digits, then associate
> this to a dial-peer going to voicemail.
>
> Chris
>
>
> On Wed, Nov 23, 2011 at 3:14 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> Yes, sure,
simple crossover Cable is OK
On Thu, Nov 24, 2011 at 3:07 PM, muhammad nouman wrote:
> Hi
>
> Quick Quesion to connect E1 back to back do I need any specail cable or
> simple crossover cable with RJ 45 connector is OK.
>
> Thanks
>
> Nomi
>
> ___
> Fo
f H.225 messages with caller-ID updates. You can enter the
> command on both, under the dial-peer & under the global config mode. I do
> this under the global config mode;
>
> *voice service voip*
> *no supplementary-service h225-notify cid-update*
>
>
> Regards
>
om CUCM End user page.
>
> AXL user should have Super User rights. Try restarting the Cisco AXL Web
> Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
> removing LDAP config from the CUCM and try re-adding it into CUCM again.
>
>
> HTH
>
>
>
om CUCM End user page.
>>
>> AXL user should have Super User rights. Try restarting the Cisco AXL Web
>> Service in both CUC and CUCM. If the CUCM is integrated with LDAP, try
>> removing LDAP config from the CUCM and try re-adding it into CUCM again.
>>
>>
>>
Very interesting thing.
I am not at my Lab side now, but tommorow will get there and test it also.
On Thu, Nov 24, 2011 at 9:50 PM, Priyank Kiran wrote:
> Experts,
>
> Need help understanding the following behavior conceptually -
>
> Have the subscriber as dedicated MOH multicast server incremen
What codec are you using for normal Voice calls? is it G729?
If yes, then I have the following on my mind:
Look, when we use G729 for Voice Calls, and then try to hold the PSTN
phone, then the gateway need to connect to MoH with G711, as the Routers
Flash MoH file is G711 only.
And in your case,
Can you please post the Solution for that task Amit?
Or point to the IP Expert LAB number, where the solution is?
On Thu, Nov 24, 2011 at 11:19 PM, Amit Singh wrote:
> Mate.
>
> Ipexperts labs have covered all these topics.
>
> If u have prepared for some other labs. Can't say anything.
>
> If
;
> Priyank
>
> On Thu, Nov 24, 2011 at 3:06 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> What codec are you using for normal Voice calls? is it G729?
>>
>> If yes, then I have the following on my mind:
>>
>> Look, when we us
Hello,
I have a problem with CUC.
When the external (PSTN) caller leaves a VM for CUCM IP phones, the MWI
does not lights up, and even the recorded message is not present in the
users mailbox. WHile the CUC tells the PSTN caller that the message has
been received. But actually, I cannot have it i
How to enable CUC to transfer to external destinations?
By default, when the CUC answers the call we can diall only the Extensions
that are presend in CUC users.
But how to enable CUC to transfer to any destination?
___
For more information regarding in
Hello,
I have a Route Pattern like - 9.011!# for international calls.
When the international call comes to CUCM, I am using the Normalization
Rules (somethimes Translation Pattern) to add the 9011 to calling party, so
that it will be possible to dial back from the Missed/Received Calls.
But th
herwise you can
> configure alternative number on the CUC .
>
>
> Ash
>
> On Thu, Nov 24, 2011 at 3:29 AM, datucha123 datucha123
> wrote:
> > I have tryed the same with SRST, but that Mask in VM Profile did not take
> > effect :(
> >
> > Can anybody tell me, wh
t for off-net
> incoming call not so I had to insert an transcoder resource in mrgl of
> gateway.
> You can do another test without transcoder but be sure you are using g711
> from pstn gateway --> ccm--> ip phone.
>
>
>
>
> Ion
>
>
> ------
Nov 25, 2011 at 9:01 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> As I know, the CUC has the G729 Support as well, So I do not think that
>> the issue is because of Tarsncoder.
>>
>> My local Phones with G729 can leave a voice mail without a p
That is a very easy
On Fri, Nov 25, 2011 at 6:43 PM, andrew margis wrote:
> Hi,
>
> http://www. certknowledge
> .com/forum/index.php?topic=39.0 (It's certknowledge(.)com/forum)
>
> I m working on early offer solution for sip.
>
> Regards
>
Hello,
Is it possible to integrate CUCM with AD (For users), so that CUCM will
import users from AD, but make CUC to import users from CUCM (those that
are imported from AD)?
___
For more information regarding industry leading CCIE Lab training, please
ranslation.
Also the MWI does not work in such case.
Am I correct for this post Ashraf ?
On Fri, Nov 25, 2011 at 9:40 PM, Ashraf Ayyash wrote:
> Can you give the calling number Translation try ?
>
> let me know how it goes
>
> Ash
>
> On Fri, Nov 25, 2011 at 7:07 AM,
e CCM make sure you will have CSS of the VM port contain the
> right PT to make the call hit whatever RP you meant to send to the
> call to
>
> Ash
>
> On Fri, Nov 25, 2011 at 5:36 AM, datucha123 datucha123
> wrote:
> >
> > How to enable CUC to transfer to extern
Hello,
My CUE does not Turn Off the MWI for Broadcast Message when the User will
listen and delete the message.
Only after the "mwi refresh all" command, the MWI will turn off on IP
Phones, for the Broadcast message.
As for normal messages and GDM, the MWI works fine.
___
>
> E
>
>
> On Fri, Nov 25, 2011 at 9:01 AM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> As I know, the CUC has the G729 Support as well, So I do not think that
>> the issue is because of Tarsncoder.
>>
>> My local Phones with G729 can le
rom my iPad
>
> On 26/11/2011, at 7:57 AM, datucha123 datucha123
> wrote:
>
> Which one I have to edit:
>
> Default
> Transfer<https://172.16.4.124:8443/cuadmin/restriction-table.do?op=read&objectId=5570ed8d-0631-4bf8-b8ad-1a2b077bd320>
> Default System
> Transfer
at a time and test. And let everyone know
> your findings.
> It will take less than 5 minutes. In the real lab u do not have access to
> this forum. Just keep this in mind.
>
> Regards
> Amit
>
> Sent from my iPad
>
> On 26/11/2011, at 7:57 AM, datucha123 datucha
races yet? What do you see in them?
>>> Does this work fine internally if you dial from an IP Phone registered with
>>> the CM?
>>>
>>>
>>> - Gurpreet
>>>
>>>
>>>
>>>
>>> On Fri, Nov 25, 2011 at 2:21 PM, datucha
CCX Doc
>> ICD + 1button = Support > voice & CUCM > Contact Center Express
>>
>> MVA = CUCM > help > search > mobile voice access > look for H323/PRI
>>
>> CME Feature and configuration = Documentation > CME> tech notes and
>> config guide
That's because of Codec Problem between HQ router and the CUCM.
Now look, as we know, when the CUCM sends the H323 SETUP message to GW, the
GW, by default, begins the H245 Address announcement on Alert Message, and
at that point the CUCM and GW have exchanged the H245 message, and the GW
provides
On Sat, Nov 26, 2011 at 1:59 PM, datucha123 datucha123 <
> datucha...@gmail.com> wrote:
>
>> I had several CUC installations in Production Environment, and did not
>> have any problems like that.
>>
>> I will try to collect the Traces for that call from CUCM an
sue:
>
> Is your voice-port in an active (not shut down) state?
>
> Do you have the proper DSP resources for your configuration?
>
> Check these and post your configuration if this is not the issue.
>
>
>
> On Sat, Nov 26, 2011 at 1:24 PM, datucha123 datucha123 <
> d
You can also use the "File List" command to navigate to the actual file
directory. And after use the "File View" command to see the file content.
You can also use the "File Tail" command (but in such case, but for Log
files for instance) to view the File content in real time
On Sun, Nov 27, 2011 a
First you need to integrate UCCX with CUCM fully, meaning that the JTAPI
and RMCM users must be present in CUCM after successful integration.
Please verify that you have configured the JTAPI and RMCM in UCCX, so that
it will be replicated in CUCM automatically.
Then make JTAPI resynch.
When all
Do you mean IP Expert LABs?
On Sun, Nov 27, 2011 at 9:12 PM, Edgar Feliz wrote:
> Guys I have been working on the new (5) labs mostly 1-3 since they have
> the guide. Here is what i have found to be things to watch for and fixes,
>
> Please note i have been working on these for over 10 days and
Hello,
I have been working in the following issue for several days but could not
get it fixed:
When the CUE send a Broadcast message to its users, the MWI is turned on
all IP Phones normally.
But when this broadcast message listened and deleted by the Users, the MWI
does not go away, the Lights
Hello,
Is it possible to upload the custom prompts to Unity Connection?
I know, that CUC has the integrated JAVA Applet, which allows you to record
the prompts either through the PC or IP Phone.
But what if I have a prompt file already located at my PC, and I want to
use it in Unity Connection
Thats because the CUE can still communicating with CUCM, hence is sending
the SIP Notify messages for MWI with CUCM IP address as "To" header, which
cannot be identified by SRST.
As for initial booting, the MWI is sending normally, because the JTAPI is
not activated in CUE yet, and the SIP notify
I have tested it, and the MWI works indeed in SRST.
For instance you have a Branch site with BR Router with CUE module inside,
and HQ site with CUCM.
Now when the WAN between the BR and HQ sites is down, the IP Phones will
re-register to their local BR SRST.
Just do not forget to configure the "
Also you have to check the "Display IE" checkbox in CUCM H323 gateway
configuration.
On Tue, Nov 29, 2011 at 7:05 AM, Rrcrumm wrote:
> Do you have "isdn outgoing display-ie" under serial 0/0/0:23 or 15
> interface?
>
> Randall
>
>
>
> Sent from my iPhone
>
> On Nov 28, 2011, at 5:52 PM, ccielabr
Hello,
Is there a topic requiring SIP integration of Unity Connection with CUCM on
the LAB? Or only SCCP is asked always?
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and lo
gt; cid-update", it seems to me this would ENABLE
> the CID to be updated.
> It doesn't make sense to me why DISABLING this with "NO
> supplementary-service h225-notify cid-update" actually allows the gateway
> to trigger an updated display on the phone.
>
&g
e
>
> ! Under PRI Interface
> !
> interface Serialx/x:23
> isdn supp-service name calling
>
>
>
> Thanks
> --- On *Tue, 11/29/11, datucha123 datucha123 *wrote:
>
>
> From: datucha123 datucha123
> Subject: Re: [OSL | CCIE_Voice] Calling Name with
Hello,
I am facing a Problem with MoH over the SIP/H323 Trunk with MTP.
I know, that the multicast MoH is not supported when the MTP is check in
the H323 gateway, but I have a problem with Unicast MoH with transcoding.
But I have the following configuration.
Only G711U is activated for MoH (def
is you should be able to hear MoH instead of ToH.
>
> ** **
>
> Regards,
>
> Mohammed Al Baqari
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *datucha123
> datucha1
When you have Voice Class Codec for incoming Dial-peer on CUCME, with G711
and G729, the calls to CUE are not transcoded, and the calls are failing.
That's because CUE uses only G711 and as the G711 is also found in the
Voice Class codec, the call is trying to take G711, whereas CUCM IP Phones
are
phone does not hear the MOH.
>
> Symptom 2 is observed if the CCM endpoint does a conference, hold, or
> transfer.
>
> Workaround: Use an MTP.****
>
> Regards,****
>
> Mohammed Al Baqari
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
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