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
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
___
For more information regarding industry leading CCIE Lab
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
___
For more information
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 somp...@gmail.com wrote:
On Sun, Aug 11, 2013 at 11:17 AM, Somphol Boonjing somp...@gmail.comwrote:
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
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 ??*
application
service app-b-acd
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
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
Ya we must add the roles are groups
This is allowed
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
www.PlatinumPlacement.com
I think that is the bug
Only resolution is to relaod the router
Any comments on it ??
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
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 ??
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
over to SRST?
--Todd
On Jul 28, 2013, at 5:56, IE Target myfrnd...@gmail.com wrote:
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 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,
Hi Singh
uset the command
debug ip dhcp server packets
debug ip dhcp server events
Thanks
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a job? Check out
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
___
For more information regarding industry leading CCIE Lab training, please
.
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 whl...@gmail.com wrote:
Can you use google?
Sent from my iPhone
On Jul 20, 2013, at 7:22 PM, IE Target myfrnd...@gmail.com wrote:
Hi GUYS
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
___
For more information regarding industry leading CCIE Lab
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
___
For more information regarding industry leading CCIE Lab training, please visit
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
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 myfrnd...@gmail.com wrote:
If we dont Use Transcoder
HQ to SB (TEHO) calls will be unsuccessful (dropped due to unavailability
of Xcoder)
Thats why
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
22 matches
Mail list logo
