It means that the invite SDP did not match with the incoming dial peers sdp.
i.e. there is a codec mismatch. So look at the incoming dial peer and check
the codec in the invite and match it approprately
On Tue, Jan 26, 2010 at 9:52 PM, Wayne Lawson wrote:
> Matthew - How are the Vol 1 Video Solut
since that device is assigned to a RP directly remove that RP and add the
device in a RG check the dependacy of the device so u will know on which RP
u assigned it
On Thu, Jan 21, 2010 at 9:59 AM, Arun Kumar wrote:
> Hi
>
> I'm working on Lab5 in Vol1.
>
> I noticed wired issue, when I tried to
Hi group,
wanted to get everybodys thought on this. if the exam asks
call control of phone using cups what r the things to be configured?
i m thinking voicemail , deskphone no soft mode and be able to change
presence status is there anything else i m missing ?
thnx in advance
__
THis is a general issue you have to decode the script to see what the issue
is. plus check which script you are referecing and all the parameters csq
etc are in line
On Wed, Jan 20, 2010 at 8:45 PM, vccie2010 wrote:
> I have UCCX on vmware. I am able to make calls succesfully to UCCX when
> ther
t; name full control there,
>
> HTH,
>
> On Sun, Jan 17, 2010 at 4:54 PM, kill mill wrote:
>
>> Wanted to run this by you guys. If BR2 is h323 and calling name
>> requirements are mentioned Then do we also take care of the calling name
>> requirement in SRST. Sin
HI appreciate if anyone can point me to the right answer for this. My
proposed answer leads to lot of excess RP's
thnx
On Mon, Jan 18, 2010 at 11:29 PM, kill mill wrote:
> I am facing the same issue did anyone find a solution. Basically I want HQ
> user to log in the BR1 site b
I am facing the same issue did anyone find a solution. Basically I want HQ
user to log in the BR1 site but still be making the same calls like he used
to when he was in HQ. Keep in mind HQ Device pool as well as BR Device pool
have LRG so what i am seeing is if a pattern matches a LRG the LRG is ta
As long as you take care of the advertise g722 codec u should be fine in the
real lab.
On Mon, Jan 11, 2010 at 8:05 PM, Scott ODonnell wrote:
> I have a couple 7962's but no 7965's.
>
> Looking at the technical specs, they look the same other than the color
> display.
>
> Is this correct?
>
>
>
>
Just checked and it seems to work please check the css or RP report to see
if you dont have another 7800 somewhere lurking around
On Sun, Jan 17, 2010 at 11:16 PM, kavi ten wrote:
> Hi Guys,
>
> Did anybody manage to test this feature from CUCM Features Guide 7.0.1 Pg
> 4-20.
> If not yet can so
Wanted to run this by you guys. If BR2 is h323 and calling name requirements
are mentioned Then do we also take care of the calling name requirement in
SRST. Since in srst
the CME would send the External number MAsk as calling name so you might end
up with calling name as External MAsk and calling
Hi,
Had a question. If the question asks guarantee 10 k for sccp traffic
and mark excess to dscp 8 then i configured the following
class-map match-all signal
match access-group name signal
ip access-list extended signal
permit tcp any any eq 1720
permit tcp any any range 11000 11999
perm
party number type
> desired,
>
> On Wed, Jan 13, 2010 at 11:40 PM, kill mill wrote:
>
>> Hi.
>>
>> I ran into this problem the requirement is to send international type in
>> the setup and 14088532345' from BR2 to the PSTN to call hq pstn numbers but
>>
Hi.
I ran into this problem the requirement is to send international type in the
setup and 14088532345' from BR2 to the PSTN to call hq pstn numbers but I
see that gw is overidding the plan is there a workaround for this. I thought
of isdn map address but then
i will have to make it for local as w
-boun...@onlinestudylist.com] *On Behalf Of *kill mill
> *Sent:* Wednesday, January 06, 2010 11:56 PM
> *To:* ccie_voice@onlinestudylist.com
> *Subject:* [OSL | CCIE_Voice] +Teho behaviour not consistent
>
>
>
> Hi,
> I am getting an issue in this scenerio and I w
Hi,
I am getting an issue in this scenerio and I wanted to know what the
exact sequence is of change from Translation-pattern to the RP to the RG. I
am trying to dial a global
number from BR1 +14088532345 but the requirement is to show 8532345 on the
BR1 phone. I did the following
Created a t
On Tue, Dec 22, 2009 at 8:32 AM, kill mill wrote:
> Hi,
> I was testing a script to play a prompt and send a disconnect to the ccm
> and hang up gracefully in other words i hit the script the ICD plays a
> prompt and I want to hang up without the system prompt stating we are having
&
Hi,
I was testing a script to play a prompt and send a disconnect to the ccm and
hang up gracefully in other words i hit the script the ICD plays a prompt
and I want to hang up without the system prompt stating we are having
problems. I dont seem to be able to
get a graceful disconnect unless I red
Hi Aamir,
How do you go to VM do u create a dial peer for 3050
in CME for SRST. ?
On Wed, Aug 12, 2009 at 6:32 AM, Jonathan Charles wrote:
> Regular SRST... Cisco says MWI is supported, but no doco on how.
>
> J
>
> On Tue, Aug 11, 2009 at 8:39 PM, Aamir
> Panjwani wrote:
>
hi,
Did any one face this issue ? If i enable the EM on a phone then the
mobility softkey for voice moblle connect states I am "You are not a valid
Mobile Phone User"
If i disable it it works. Is there something i am missing ?
Thnx
___
For more inf
t off if
> they didn't specifically make an assignment that requires it.
>
> Cliff
>
>
> - Original Message -
> *From:* kill mill
> *To:* ccie_voice@onlinestudylist.com
> *Sent:* Monday, May 04, 2009 11:01 PM
> *Subject:* [OSL | CCIE_Voice] Puttin VM pilot
Guys,
I wanted to get some opinion. Is there any flaws in putting the VM pilot
/ Ports and the MWI DN's in internal partition accessible to everybody ?
Will the procter cut marks on that ?
Thnx
Guys I wanted to know if we use VOice vlan to source the SRST would
they deduct any points in the lab as supposed to loopback . Technically
voice vlan will be up all the time on the BR router because of the phones so
if i got 0 in SRST then its prolly because of something else correct ?
when i say bind remove the mgcp bind control and then bounce the mgcp
On Mon, Apr 27, 2009 at 11:15 PM, kill mill wrote:
> Can you remove the the bind and then bounce the mgcp and get the following
> debug
> deb ccm-manager back-haul events
> deb isdn q921
>
> Thnx
>
>
Can you remove the the bind and then bounce the mgcp and get the following
debug
deb ccm-manager back-haul events
deb isdn q921
Thnx
On Mon, Apr 27, 2009 at 8:38 PM, Frank Costeira wrote:
> Hi,
>
> I ran into the same issue. It drove me nuts for about 1/2 hour.
>
> In my case I reapplied the mgc
Guys,
I made both the methods work when in SRST to get to the right voice
mail box as a workaround for the bug. Both these methods rely on the fact
that the number we choose to redirect to VM is routable in the PSTN back to
the HQ router. Can any one confirm which number should be used in t
Any suggestions on this please ? Wanted to know how to decide the VM pilot
number ?
On Sun, Apr 26, 2009 at 10:48 PM, kill mill wrote:
> Guys had a quick question on this. When we use the CTI RP workaround I
> see you guys send the call to some known number for example 1641 but how do
r and hope it
gets routed to the HQ site ? Is there some thing specific you look when you
decide what number to route back to the HQ ?
Thnx
On Sun, Apr 26, 2009 at 10:47 PM, kill mill wrote:
> Guys had a quick question on this. When we use the CTI RP workaround I see
> you guys send the c
Guys,
Can some one tell me how this can be configured. If I configure
CUE and Unity with DNS names all works fine including broadcast but once i
change the unity's domain name to ip address in CUE CUE stops accepting
messages nor sends any. Can some one put the exact steps to get this wor
The param voice mail will send the call to voice mail after the max call
retry timer has expired. The calling number will be the hunt pilot number
and Unity will play u the fantastic greeting that this mail box does not
exist and transfer u to the AA.
//ankur
On Thu, Jan 29, 2009 at 5:07 AM, kama
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