> importantly I don't revisit config pahes just to setup aar.
>
> Hope this helps...
>
> -Justin
>
> On Nov 4, 2013 6:57 AM, "CISCO CCIE VOICE" wrote:
>
>> Hi Guys,
>>
>> i am trying to configure AAR between 2 sites, when i assign AAR-GROUP a
Hi Guys,
i am trying to configure AAR between 2 sites, when i assign AAR-GROUP and
AAR-CSS on device pool it does not take effect rather i have to apply it
each phone device and GW inorder for it to work.Is there any thing i am
missing
Thanks
___
For mo
Hi Experts,
I am not able to restrict max-call-per-button 2 and busy-trigger-per-button
1 using ephone-template
ephone-dn-template 1
call-forward busy 2220
call-forward noan 2220 timeout 20
huntstop channel 1
ephone-template 1
softkeys remote-in-use Newcall CBarge
max-calls-per-button 2
hannel selection order is not bound to your telco
> configuration.
> You can set top-down or bottom-up with no issue at all.
>
> hth
> lorenz
>
>
> On Wed, Jul 3, 2013 at 10:58 AM, CISCO CCIE VOICE
> wrote:
>
>> Hi Guys,
>>
>> I am trying to understand the
Hi Guys,
I am trying to understand the "Channel selection order", i know that if
TELCo is using bottom Up then i have to use opposite Direction i.e Top
Down .My Telco is uses last channel for IN-bound calls and set Top down on
my side for OUT-bound calls but still i can see that its taking bottom
38404
>
> “Designing and Implementing Cisco Unified Communications on Unified
> Computing Systems”
>
> ** **
>
> *From:* CISCO CCIE VOICE [mailto:ccievoic...@gmail.com]
> *Sent:* Monday, July 01, 2013 12:58 PM
> *To:* Sears, Michael (msears)
> *Cc:* ccie_voice@
Thank You Michael, Question asked about L1 should me NETWORK SIDE and L2
USER SIDE in that case
controller t1 0/0/0
clock source line -L1 Network Side (This is by default
enable no need to add it)
and for L2: USER SIDE do we need to add any additional commands under
serial int
Hi Michal,
As i can see that Clock source line command which is by default enable
further can u give an example for both "User Side" and "Network Side"
means can u past the commands that indicates both user and Network Side
Thanks
On Mon, Jul 1, 2013 at 7:00 PM, wrote:
> Send CCIE_Voice mai
Hi Amit,
Remove the TCL application and add it again i will work
Thanks
On Sun, Jun 30, 2013 at 9:00 AM, wrote:
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://onlinestudylist
Hi Todd,
Please add the below commands and let me know the result...
voice service voip
allow-connections h323 to h323
h323
emptycapability
h225 id-passthru
h225 connect-passthru
Thanks
On Wed, Jun 26, 2013 at 5:38 AM, wrote:
> Send CCIE_Voice mailing list submissions to
> cc
ning: parameter number-of-hunt-grps has not been registered under
> app-b-acd namespace
> Branch2(config-app-param)#do show run | begin application
> associate application SCCP
> !
> dspfarm profile 5 conference
> codec g711ulaw
> codec g711alaw
> codec g729ar8
> codec g7
t; !
>>
>> service app-b-acd-aa <-- you can't change the name of the embedded
>> BACD AA script
>>. (detail remove for brevity).
>>param service-name app-b-acd <-- refer to the embedded BACD Queue
>> script
>>param handoff-
ueue script
>param handoff-string app-b-acd-aa
>. (detail remove for brevity).
> !
>
> dial-peer voice 222 voip
> service app-b-acd-aa <-- refer to the name of the embedded BACD AA
> script
>. (detail remove for brevity)...
> !
>
>
at, Jun 22, 2013 at 1:00 AM, CISCO CCIE VOICE
> wrote:
>
>> Hi,
>>
>> I am trying to Remove B-ACD configuration but still showing in the
>> running configuration i have restarted the router but no look any guess?
>>
>>
>> thanks
>>
>>
>>
Hi,
I am trying to Remove B-ACD configuration but still showing in the running
configuration i have restarted the router but no look any guess?
thanks
___
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HI experts,
I am trying to remove B-ACD cnofigs from the router but still show in
running config,i have reloaded the router but still no luck any guess ?
Thks
___
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an stay in a call queue and
> retry to connect with a hunt group before the call is sent to an alternate
> destination number.
>
> •*seconds*—Maximum period of time, in seconds. The range is from 30 to
> 3600. The default is 600.
>
>
> On Mon, Jun 17, 2013 at 5:11 PM, CISCO
hi experts,
I am trying to understand timer for "*param* *max-time-call-retry" can
anyone share there knowledge about how does it effect the bacd script and
the purpose of this field*
*
*
*Thnks*
* *
___
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fairly quickly. Good luck.
>
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> On Jun 16, 2013, at 10:06 AM, "CISCO CCIE VOICE"
> wrote:
>
> I have Branch 2 site which contains Transcoder moreover all my
ctions. Otherwise you would get a busy signal if
> region said g729 and CCX was set for g711.
>
>
>
>
>
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> On Jun 16, 2013, at 9:41 AM, "CISCO CCIE VOICE"
> w
ebook<http://www.facebook.com/pages/Alexander-Open-Systems-AOS/109484829074064>
> | LinkedIn <http://www.linkedin.com/company/aos>
>
>
>
>
> Sent from an Apple iOS device with very tiny touchscreen input keys.
> Please excude my typtos.
>
> On Jun 16,
t;
> I think the easiest way is to check UCCX Service Parameters under System
> menu.
>
> --Somphol.
>
>
> --Somphol
>
>
> On Sun, Jun 16, 2013 at 7:36 PM, CISCO CCIE VOICE
> wrote:
>
>> Hi,
>>
>> How ca
Sent from Mailbox <https://www.dropbox.com/mailbox> for iPad
>
>
> On Sun, Jun 16, 2013 at 3:37 AM, CISCO CCIE VOICE
> wrote:
>
>> Hi,
>>
>> How can i verify that UCCX is using G729 codec native
>>
>> Thnajs
>>
>
>
___
Hi,
How can i verify that UCCX is using G729 codec native
Thanks
___
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Hi,
How can i verify that UCCX is using G729 codec native
Thnajs
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It will be a better option to cancel or reschedule CCIE voice lab exam and
take it after Feb 2014 ,its wast of money,time and effort on CCIE Voice
On Wed, May 29, 2013 at 10:27 AM, wrote:
> Send CCIE_Voice mailing list submissions to
> ccie_voice@onlinestudylist.com
>
> To subscribe or u
Hi experts,
i am also searching for that solution thanks i will also try that and let
u know, which command i need to run on SB gateway to get the output,do i
need call start fast under voice service voip in SB gateway ?
Thanks
On Tue, May 21, 2013 at 11:41 AM, wrote:
> Send CCIE_Voice mail
, 2013 at 10:21 PM, CISCO CCIE VOICE
> wrote:
>
>> HI experts,
>>
>> When the Phones are in CME SRST its does not shows Extension on Phone
>> display this happens when i change the name +86223033001 to S
HI experts,
When the Phones are in CME SRST its does not shows Extension on Phone
display this happens when i change the name +86223033001 to SiteC Phone 1
under Ephone-dn's
Thnks
___
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Hi Experts,
When i dial UCCX Trigger i am not able to play custom welcome prompt rather
its playing MOH.
thanks
___
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Are you a CCNP or CCIE and looking fo
Hi Experts,
How can i send Calling name to PSTN over MGCP and H323 Gateway?
thanks
___
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I am facing the same problem when is use default icd.aef script
___
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ly
> validated
> 4-Then go to UCCX Admin > Script Management -> Upload the Script
> --> Overwrite the Script --> Then Refresh The Script
> 5-Refresh the Application Script
>
> Then you should be good to go
>
> Thanks,
> Hesham
>
>
> On 29 Ap
Hi Experts,
I am getting an error "i am sorry we r currently facing system problm and r
unable t process ur call" during CSQ
Thanks
___
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Are you a CCNP or
OK, The difference is the variation
> between the INVITE and the ACK.
>
> This link has further description.
> http://www.iplogos.fr/English-Resources/Focus/sip-early-media-early-offer-en.html
>
>
>
> On Sun, Apr 28, 2013 at 2:51 AM, CISCO CCIE VOICE
> wrote:
>
>> Thank you
Thank you suresh i will check ths on my pod...
On Sun, Apr 28, 2013 at 11:26 AM, Suresh Bhandari wrote:
> Content Length 0 means, there will be an OK followed with SDP So
> basically both are same.
>
>
> On Sun, Apr 28, 2013 at 1:56 PM, CISCO CCIE VOICE
> wrote:
>
>
, Suresh Bhandari wrote:
> If you are busying out the channels in CUCM using automatic mgcp
> configuration, it will not be Fractional PRI... Experts, please correct me
> if I am wrong!
>
>
> On Sun, Apr 28, 2013 at 2:01 PM, CISCO CCIE VOICE
> wrote:
>
>> HI Thanks Again,
>
> Notice the boldface text that represents your gateway.
>
> Further, you have to check Enable status Poll in your gateway page.
>
> Hope this helps.
>
>
> On Sun, Apr 28, 2013 at 12:45 PM, CISCO CCIE VOICE
> wrote:
>
>> HI Thanks,
>>
>> but if suppo
including SDP message and so its an early offer.
>
> HTH
>
>
> On Sun, Apr 28, 2013 at 1:15 PM, CISCO CCIE VOICE
> wrote:
>
>> HI Experts,
>>
>> Which SIP Message contains Delay Offer and Early Offer message ? i i am
>> bit confuse as some document
HI Experts,
Which SIP Message contains Delay Offer and Early Offer message ? i i am bit
confuse as some documents says SIP INVITE message contains both Delay and
Early Offer and some documents say its SIP 200 OK message
Thanks
___
For more information
HI Thanks,
but if suppose i want to use 6 channel from T1 PRI on HQ Router and 12
channel of E1 PRI from Branch-2 router,so how can i busy out channels under
service parameter ? and how does CUCM knows about which router interface to
busy out as i have two site with different fractional PRI
Any
Hi Experts
Which method to be use in CCIE Voice lab exam in order to configure MGCP
Auto or Manually.
Thanks
___
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Are you a CCNP or CCIE and looking for a
HI experts,
When Branch-2 MGCP router is in CME SRST mode ,i am unable to see Alerting
message on inbound call from PSTN,moreover when i call from PSTN my call
is directly connected and there is no ring back tone ,Can any one help to
fix the issue.
ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8
Hi Guys,
Can i group H323 GW and SLRG in one Route List ? it that the best practice
?can any one provide me with the reference for best practice to which
protocol in can group together like h323, mgcp,sip trunk,GK Trunk ect in
to Route List.
Thanks
___
Hi Guys,
Regarding E1 Framing if the question states the below
FOR E1 use the following:
switch type:net5
framing :CRC
linecode :HDB3
then do i need to use the NO-CRC or CRC ?
___
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ISPLAY IE in outgoing ISDN messages is allowed
>>
>> and
>> *isdn outgoing ie ?
>> *
>>...
>>displayDISPLAY IE in outgoing ISDN messages is allowed
>>...
>>
>> so this may be otherwise option for the same display-ie.
>>
>>
Hi guys,
what is the difference between IE display and display IE command
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bind-l3 ccm-manager
isdn outgoing *display-ie VS ie dislay*
isdn outgoing ie redirecting-number
Hi Experts,
Can any one share there knowledge and experience on how to troubleshoot
one-way audio when the call is answer from PSTN phone which messages do i
need to look at on RTMT and which traces do i need to enable on CUCM to
check the One way audio problem ..
Thanks
_
Hi experts,
i have been trying to test the Call forward unregister whn the call arrive
on destination phone its show on the display as FORWARD,FOR and BY on the
phone screen so is there a way to display "FROM" on the destination phone
screen
thanks
___
Hi experts,
i am facing a problem with CUCM DHCP ,my phones are taking an ip address as
first IP from starting range and so on actually it has to take last ip
first and so on
eg: CUCM DHCP range :177.1.11.10 -177.1.11.30
MY phones getting an ip as 177.1.11.10
thank
iguration/guide/b_cflash.html#wp23144
>
>
> ** **
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE
> *Sent:* Tuesday, March 12, 2013 10:51 AM
> *To:* ccie_voice@onlin
Hi Experts,
can any one help me I want to create directory folder on router without
formatting flash when i use mkdir command its saying that invalid input
thnks
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HI experts,
If the question ask to configure H323 Trunk to backbone router then in that
do i need to configure it as H323 Gateway or H323 Non -GK controlled trunk?
Thnks
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Hi Experts,
Can anyone explain me how to collect h323 trunk debugs from RTMT ...
Thanks
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HI Guys,
Can any one Share H323 and SIP Trunk configuration that need to be done on
PSTN Router in order for it to work properly ...
Thanks
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Are you a
HI All
When the Branch-1 is in SRST Proper Mode (call-manager-fallback) When i am
trying t press voice mail button its playing system greeting not the user
greeting ,its there any thing i need do on CUCM in order to play user
greeting when the phones are in SRST..
Thanks
___
>3. Re: Phone NTP Reference Vs NTP Reference (Cory Gray)
>4. Re: Phone NTP Reference Vs NTP Reference (CCIEing)
>5. AAR HELP (CISCO CCIE VOICE)
>6. Re: AAR HELP (Jamie Parr (jamparr))
>7. Re: AAR HELP (CISCO CCIE VOICE)
>
>
> --
*
> Engineer - IT
> jamp...@cisco.com
> Phone: *+44 20 8824 2641*
> Mobile: *+44 7590622049*
>
> ** **
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CISCO CCIE VOICE
> *Sent:* 05 March 2013 08:13
&g
Hi experts,
Can any one help me to understand the below Question
Call from HQ to Branch-2 should invoke AAR base on RSVP bandwidth,If
HQPH1 calls BR2PH1 then it should display E.164 number and Network
congestion Re-routing Message on Phone
So ,my doubt is that do i need to use AAR and l
n error, please contact the sender and delete the material from any
> computer.
>
> __
>
> ** **
>
> From: CISCO CCIE VOICE
> Date: Tuesday, February 26, 2013 2:44 PM
> To: "ccie_voice@onlinestudylist.com&q
Hi,
Is it safe to use BAT in real lab exam?
thanks
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, 2013, at 6:45 AM, "CISCO CCIE VOICE"
> wrote:
>
> Hi
>
> Can any one help me with Dial Plan consideration when calling from HQ Site
> to Branch 1 Site,following what i have configure.But the problem is that on
> B1PH1 screen its showing as below
>
&g
Hi
Can any one help me with Dial Plan consideration when calling from HQ Site
to Branch 1 Site,following what i have configure.But the problem is that on
B1PH1 screen its showing as below
*From +14082021001
Forward by:2001*
*HQ SITE:*
*Extension Range*:1XXX
*Partition:*Branch_1_SRST_PT
*CSS
HI,
Does MWI works on SRST Proper mode (call-manager-fallback).?
Thanks
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HI
When in SRST either SRST Proper or CME as SRST ,when i call from HQ to
branch office do i need to assign unregistered DN in order to have ext to
ext calling working case of WAN fails
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Anyone else interested? few left
On Wed, Aug 11, 2010 at 8:21 PM, Cisco CCIE wrote:
> Sorry for an offtopic post but I purchase a bundle of IPX PL voice rack
> rentals but ended up building my home voice lab so these are up for grabs.
> You can unicast me for details.
&
Got a few more left and I am done
On Wed, Aug 11, 2010 at 8:21 PM, Cisco CCIE wrote:
> Sorry for an offtopic post but I purchase a bundle of IPX PL voice rack
> rentals but ended up building my home voice lab so these are up for grabs.
> You can unicast me for details.
&
t happens with your debug tftp events? Is
> the phone looking for the path and files you have uploaded? Sometime you
> may need to adjust your tftp-server settings with the alias command
>
> cheers
>
>
> On Sun, Aug 22, 2010 at 8:01 AM, Cisco CCIE wrote:
>
>> Yup th
large.png
> May 23 15:44:39.087: TFTP: Opened flash:Desktops/320x196x4/large.png, fd 8,
> size 73628 for process 294
> May 23 15:45:00.323: TFTP: Finished flash:Desktops/320x196x4/large.png, time
> 00:00:21 for process 294
>
> http://tinyurl.com/39cu8eq
>
>
> Ash>
>
OK so it appears that this has been happening with others as well. I did a
search but none had it resolved. I had this working and then decided to redo
the scenario but now it just won't work. Here are all the configurations
just incase someone asks for it. Is there a bug with CME that makes this
h
wow a bunch sold in 16 hours.. few more left and I am good :) Thanks all.
gotta love this list
On Wed, Aug 11, 2010 at 8:21 PM, Cisco CCIE wrote:
> Sorry for an offtopic post but I purchase a bundle of IPX PL voice rack
> rentals but ended up building my home voice lab so these are up for
Sorry for an offtopic post but I purchase a bundle of IPX PL voice rack
rentals but ended up building my home voice lab so these are up for grabs.
You can unicast me for details.
Thanks
SoontobeTripleCCIE
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Great feeling when you figure it out on your own
I figured out the problem.
And as usual, I'm the architect of my own demise :)
I had redundant tftp-server statements for the term62.default.loads file.
(along with others as well)
One for SCCP and one for SIP.
tftp-server flash:/SCCP.8-3-3s/term6
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