Ya
MRGL_FS Contains MRG_FS which contains SB_Xcode and SB_MTP (G711)
dspfarm profile 1 mtp
associate application sccp
codec g711ulaw
maximum session software 4
no shut
dspfarm profile 2 transcode
associate application sccp
codec g729r8
maximum session 2
no shut
MRGL_FS is applied to
the IP Voice Media Streaming App to G729 only. Then reset the
> IP Voice Media Streaming App service.
>
>
> On Sat, Jul 13, 2013 at 1:02 PM, IE Target wrote:
>
>> If we dont Use Transcoder
>>
>> HQ to SB (TEHO) calls will be unsuccessful (dropped due to unavailability
I have tested it without CUP trunk . It just works
So the real question is what is the use of SIP Publish Trunk
If the CUPS (Soft and Desk Phone) works without SIP Publish Trunk
Should we create it
I know a guy who got marks even though he did't create the SIP Publish trunk
He said it is not req
The message which indicates that an MGCP call is tearing down is
MDCX or DLCX.??
Some guys say that it is DLCX no MDCX
May be some one who got marks in this section can clarify
Thanks
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May be you missed
dtmf-realy sip-notify
Under the dial-peer to CUE
And hence DTMF is not sent to CUE and you are not authenticated
Hope That helps
Thanks
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Hi GUYS
Can AnyOne explain the purpose of these command
i.e what happens when we add them/remove them
isdn bchan-negotiat
isdn sending-complete
isdn send-alerting
Thanks
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order .
Whether i add or not , Alerting message is always send before connect in my
lab.
So i am confused.
Thanks
On Sun, Jul 21, 2013 at 6:43 AM, Bill Lake wrote:
> Can you use google?
>
> Sent from my iPhone
>
> On Jul 20, 2013, at 7:22 PM, IE Target wrote:
>
> > Hi GU
Hi Singh
uset the command
debug ip dhcp server packets
debug ip dhcp server events
Thanks
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ww
And the tab and question mark on router will work as normal
there is 100% no doubt on that.
We are not supposed to remember every command We can use the tab.
These are all doubts OR things people used to worry about in V2 of CCIE-V.
Thanks
On Thu, Jul 25, 2013 at 9:30 PM, wrote:
> Send CCIE_Vo
Facing this issue plz help
When sb phone fall back to SRSt(call-manager fallback)
They show registered in show ephone
BUT
they do not get any dn
I tried reloading router, factory reset and every thing
What is the solution . Has any one faced problem like this
plz reply
_
ver to SRST?
> >
> > --Todd
> >
> > On Jul 28, 2013, at 5:56, IE Target wrote:
> >
> >> Facing this issue plz help
> >>
> >> When sb phone fall back to SRSt(call-manager fallback)
> >>
> >> They show registered in show ep
Ya we must add the roles are groups
This is allowed
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I think that is the bug
Only resolution is to relaod the router
Any comments on it ??
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Ya that is the parameter to limit queue length to 2
param queue-len 1
*RSVP*
"only assign to phones for Location.. DO not need to put to DP or others
port.
I will create problem"
What do you mean by that ? means you will assign Hub_None on Device Pools
and assgin HQ_LOC
on the phone page ??
*N
Hi karen
No need to change 525 to 9525
Use full Match to capture all digits
Ya set Busy trigger to 1
There is a mystery about calling name display when call is made to 911 via
MVA
it display calling number but not name. I did't tested it but there is some
discussion about it
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Hi EveryOne need help in this
Question says Use Standared Local ROute group for both the calls mentioned
above and it mention HQ-local and HQ-teho calls
So Should we sue SLRG for EMG call OR Not OR it Doesn't matter. Will we
loose marks for using SLRG in EMG. Is it a violation as question says US
I am curious to know this mystery.
As we are told that MVA caller should be able to call internal calls
and it should appear as if it is coming form internal Phone.
So if internal Phone calls it display caller id and calling name
So MVA calls should also be displaying calling name.
I heard it h
ccm-manager music-on-hold
ephone-hunt 1 longest-idle
pilot 4500
list 4101,4102
timeout 10,10
auto logout 2 dynamic *<*
* ===> But if agent gets logged out then subsequent calls will
directly go to Voice mail even though there are zero calls in the queue ??*
a
Great Observation.
I will try some diff solutions and let you know the results
Mostly it think now that Calling Name is not supported
On Sun, Aug 11, 2013 at 7:38 AM, Somphol Boonjing wrote:
>
> On Sun, Aug 11, 2013 at 11:17 AM, Somphol Boonjing wrote:
>
>> Reading through two thread origin
That is something even i am not clear with
One call = two call leg
Practically if we configure
maximum session software 1
the call works?
some where i read that two sessions will be used pre call ??
Any comments
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Hi
I am getting "Not Enough Bandwidth" for some RSVP calls
it is like it works mostly but some time for some call attempt i get Not
enough BW?
Did anyone face this issue?
How to resolve it
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What proves TCP
1)Outgoing SIP TCP
or
2)transport = tcp
or
3) VIA
What proves
Early Offer
1)content-length
or
2)content type
or
3)SDP itself
Outgoing SIP TCP message to 157.26.1.253 on port 5060 index 1
INVITE sip:321234567890@157.26.1.253:5060 SIP/2.0
Date: Tue, 12 Nov 2013 18:36:17 GMT
ength because the other two are more
> specific, just my opinion (hopefully this works out in my next lab).
>
> -Justin
>
> (Sent from my phone, please excuse and/or laugh at any typos.)
> On Nov 13, 2013 12:08 AM, "IE Target" wrote:
>
>> What proves TCP
>&g
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