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
The aar-group setting on device pool does NOT get pushed to all devices in
the device pool, while the aar-css does.
My strategy is to set aar-css at the dev pool and to manually set the
aar-group on EVERY device (phone, gw, vm or cti port, etc) and EVERY
line/dn. This take no thought while
Hi Justin,
Thanks for your reply i will try that and wll update you on ths...
Thanks
On Mon, Nov 4, 2013 at 4:54 PM, Justin Carney justin.s.car...@gmail.comwrote:
The aar-group setting on device pool does NOT get pushed to all devices in
the device pool, while the aar-css does.
My
:* Wednesday, September 11, 2013 3:56 PM
*To:* Daniel Pagan
*Cc:* ccie_voice@onlinestudylist.com; Ovidiu Popa
*Subject:* Re: [OSL | CCIE_Voice] AAR and SNR
** **
The behavior should not be any different if a shared line is configured
between two devices but only one device rings because
...@onlinestudylist.com] On Behalf Of Ovidiu Popa
Sent: Sunday, September 08, 2013 11:26 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR and SNR
Hello everyone
Can anyone confirm the following results ?
A - If Mobile Connect is enabled for a User then AAR calls will ring only
this helps.
- Dan
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Daniel Pagan
Sent: Wednesday, September 11, 2013 2:57 PM
To: Ovidiu Popa; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR and SNR
Confirmed
To: Daniel Pagan
Cc: ccie_voice@onlinestudylist.com; Ovidiu Popa
Subject: Re: [OSL | CCIE_Voice] AAR and SNR
The behavior should not be any different if a shared line is configured
between two devices but only one device rings because the other does not have
sufficient bw. AAR would not kick
Hi,
That's correct, as AAR would use PSTN to call the Deskphone, It'd rather
call the RDN instead of Deskpone, you see anyway its a PSTN Call.
Cheers,
Laksh
On Sun, Sep 8, 2013 at 8:56 PM, Ovidiu Popa ovi.p...@gmail.com wrote:
Hello everyone
Can anyone confirm the following results ?
A -
Hello everyone
Can anyone confirm the following results ?
A - If Mobile Connect is enabled for a User then AAR calls will ring only
the remote destination, the AAR to the DN call will NOT be extended
B - If Mobile Connect is disabled for a User, AAR calls will ring the AAR
target (as specified by
Hi Alex,
Have you enabled AAR in CCM service parameters? Not Enough Bandwidth
indicates that it's not enabled. You should receive something like
'Network Congestion - Rerouting' when AAR is invoked.
Marty
On Thu, Aug 22, 2013 at 5:00 PM, Alex Pishko alexpis...@gmail.com wrote:
All,
All,
Hopefully someone can give me an idea on this one. Working on my own
equipment but following the labs and am running into an issue with AAR.
I'm sending a call from HQ to BR 1 using 4 digit dial 5002 --- 1002. When
I attempt to make the call I receive the message Not enough bandwidth,
my android device.
-Original Message-
From: Ramcharan Arya ramcharan.a...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 7:59 AM
Subject: [OSL | CCIE_Voice] AAR and uccx
Hi,
I have a CTI Route point 4000 and two CTI port 410x
Sent from my android device.
-Original Message-
From: Ramcharan Arya ramcharan.a...@gmail.com
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 7:59 AM
Subject: [OSL | CCIE_Voice] AAR and uccx
Hi,
I have a CTI Route point 4000 and two CTI
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tue, 09 Jul 2013 7:59 AM
Subject: [OSL | CCIE_Voice] AAR and uccx
Hi,
I have a CTI Route point 4000 and two CTI port 410x and 410y
SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and
associated with trigger
Hi,
I have a CTI Route point 4000 and two CTI port 410x and 410y
SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and
associated with trigger.
Due to RSVP when call exceed ip rsvp bandwidth call from uccx to site
Phones should use AAR and to over PSTN.
My doubts are .
Dear All,
For the AAR is it permitted to USE SLRG and PLUS routing in the lab.
Thanking you as always,
Ravi.
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Are you a CCNP or CCIE and looking for a
u can use anything it shld only work
AAR is nothing work on new requirements as per lab 8 it is very hard
On Thu, Apr 18, 2013 at 2:21 PM, ie ravindra ieravin...@gmail.com wrote:
Dear All,
For the AAR is it permitted to USE SLRG and PLUS routing in the lab.
Thanking you as always,
AAR should use SLRG and Plus routing.-ikizoo
Date: Thu, 18 Apr 2013 14:21:34 +0530
From: ieravin...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] [AAR : SLRG and Plus Routing]
Dear All,
For the AAR is it permitted to USE SLRG and PLUS routing in the lab.
Thanking
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
: +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
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR HELP
Hi experts,
Can any one help me
| CCIE_Voice] AAR HELP
** **
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
List,
I have myself wondering about the implied requirements of AAR involvement
in the lab. Let's say that AAR is a requirement between Site 1 and Site 2
and system devices such as UCCX and UCON live in Site 2. Is it implied
that you need to provide for AAR services to those UCCX / UCON system
Derek,
That is an excellent question. The answer is the ubiquitous: it depends. It
is possible the question may state something like you are done with this task
when Phone A at Site 2 can call Phone A at Site 1 or something similar that
basically should be read as our script is going to do
Bill,
You are absolutely right it depends and while we all hope the test would
clearly identify the success criteria my experience has told me that
certain things are also implied as opposed to being spelled out. I thought
I would poll the audience. Thank you for the feedback.
Derek
On Thu,
Hugo,
I'm trying to understand as I don't have that lab book. Is HQ mgcp and BR1
mgcp or h323?
Michael Sears
Compucom Systems Western Region
Senior Consultant
Office: +1.720.344.6833
Mobile: +1.303.328.5590
Fax: +1.978.863.0740
On Nov 14, 2012, at 7:08 PM, Barrera, Hugo
A lot of times for me at least AAR would work after I reboot the routers.
Edgar
On Wed, Nov 14, 2012 at 10:05 PM, Bill whl...@gmail.com wrote:
So you can see the call go out the pstn at HQ and arrive at SB but you get
busy?
What is the error or reason for busy?
Normally, I do not put
Hi,
I am doing AAR, Lab 6 Volume 1, but I just can't get it to work correctly. The
following is my programming task list and what I witnessed:
Call flow = HQ Phone 2 x5002 calls BR1 Phone 2 x1002
1. Enabled AAR globally in service parameters
2. Created my AAR-PT and AAR-CSS
o
So you can see the call go out the pstn at HQ and arrive at SB but you get busy?
What is the error or reason for busy?
Normally, I do not put anything but aar pt in aar CSS. I the assign the aar
CSS to the device level. I assign the aar group to the line level. On the
target I set the aar
Have you checked that the BR router is in the same location as the BR
phones?
On 15 Nov 2012 03:10, Bill whl...@gmail.com wrote:
So you can see the call go out the pstn at HQ and arrive at SB but you get
busy?
What is the error or reason for busy?
Normally, I do not put anything but aar pt
Hi Folks - I have a 10 digit internal dial plan. I use translation patterns to
strip 9's when sites that are on the CUCM accidentally call each with a 9 and
send the calls over the WAN. I'm testing AAR and I can't seem to get it to
work,
I'm wondering if the translation patterns that strip the
Ok so I have a weird one for you. I'm in lab 7 vol 2 testing aar from HQ phone
2 to BR2 Phone 2. I bring down the bandwidth to 23. AAR works great to 3001,
and cue 3600. When I call 3002 it sits there for a moment and then just fails.
I look at the HQ gw and its sending number +447976852817. I
Hi Jason,
I've also observed the same behaviour. The only thing I've found in
documentation relating to this is the following from the 8.5 features and
services guide...
*Automatic Alternate Routing ** *
* Prior to the implementation of this interaction, if a desk phone was
configured for
a route pattern or
anything that tells it to do that.
- Original Message -
From: Kevin Spicer
Sent: 04/12/12 01:09 PM
To: Jason Murray
Subject: Re: [OSL | CCIE_Voice] AAR and RDP
Hi Jason,
I've also observed the same behaviour. The only thing I've found in
documentation relating
to
find a route pattern or anything that tells it to do that.
- Original Message -
From: Kevin Spicer
Sent: 04/12/12 01:09 PM
To: Jason Murray
Subject: Re: [OSL | CCIE_Voice] AAR and RDP
Hi Jason,
I've also observed the same behaviour. The only thing I've found
Subject: Re: [OSL | CCIE_Voice] AAR and RDP
Thanks for confirming that! Presumably your rerouting calling search space on
the RDP uses a route pattern that points to a RL containing the Standard Local
Route Group, which would explain it routing via the local gateway.
On Thu, Apr 12, 2012 at 7
.
thanks,
/Baktha
Message: 3
Date: Tue, 28 Feb 2012 06:58:58 +
From: Rynard Coetzee rynard.coet...@bytes.co.za
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR with RSVP
Message-ID:
97D19F256B859A48A6A93E8A9210E52A17AF8FCE@BYTESEXCH2K10N1
Hi All
I have a issue where my AAR is not working from HQ site to BR2 site ,I have
configured the RSVP agent and tested it ,working 100% ,I also have AAR group
,and AAR RP configured and also assigned AAR CSS to phones ,the group is also
configured under the line. I have also enabled AAR under
)
--
Message: 1
Date: Mon, 27 Feb 2012 11:40:53 +
From: Rynard Coetzee rynard.coet...@bytes.co.za
To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR with RSVP
Message-ID:
97D19F256B859A48A6A93E8A9210E52A17AF8F5A@BYTESEXCH2K10N1.bytes.local
Hi
Yes I have it configured ,I also tried configuring the AAR mask under the line
,but still same issue
From: A A [mailto:ccie...@gmail.com]
Sent: 28 February 2012 03:41 AM
To: Rynard Coetzee
Subject: Re: [OSL | CCIE_Voice] AAR with RSVP
do you have external mask configured ?
On Feb 27, 2012 3
: VXML on MGCP Gateway (Ricardo)
--
Message: 1
Date: Mon, 27 Feb 2012 13:12:17 -0500
From: Baktha Muralidharan muralic...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR with RSVP
Message-ID
, January 15, 2012 1:03 AM
*To:* Mohd Baqari
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] AAR for UCCX
** **
AAR is supported with CTI, when it is registered. For instance the CTI
Route Point supports AAR, when it is registered. But as soon as it will
unregister
Hi Datucha,
In your scenario whats confusing me is how aar is triggered. As far as I know
AAR isn't supported with CTI
Regards,
Mohammed Al Baqari
Sent from my iPhone
On Jan 8, 2012, at 9:11 PM, datucha123 datucha123 datucha...@gmail.com wrote:
Hello,
I have been making a test and find
Mobility feature when
users roam across different sites.
Regards,
Mohammed Al Baqari
From: datucha123 datucha123 [mailto:datucha...@gmail.com]
Sent: Sunday, January 15, 2012 1:03 AM
To: Mohd Baqari
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR for UCCX
AAR
Hello,
I have been making a test and find out the following issue:
So, I set the AAR Mask to CTI Route Points Line Settings, and the calls
from Branch1 to UCCX when there was no WAN bandwidth available were routed
through PSTN successfully.
I have not set any AAR Group and AAR CSS for CTI
This behavior usually mean that you have missing AAR group/CSS
somewhere in the call flow you have , also take to the consideration
the Bug of the AAR group will only be applied on the Line Level of the
phones , Btw the CCM trace will give you a very Nice explanation of
why the Call didnt got
@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR with Route Patterns
Hi
Check if the phone devices and have an aar CSS, aar group is on the line . Add
both to your gateways and route list can use local route group, make sure aar
rp is sending the correct number of digits to PSTN
HTH
Randall
Sent
Hi guys
I`m trying to get the following scenario to work ,but can`t seem to get the AAR
to invoke. I have 2 GW`s out to the PSTN ,they are located at HQ site and BR1
site ,what I have a RPattern that sends all calls from HQ/BR1 phones out of the
HQ GW. I then have the same RP`s created in a AAR
Hi
Check if the phone devices and have an aar CSS, aar group is on the line . Add
both to your gateways and route list can use local route group, make sure aar
rp is sending the correct number of digits to PSTN
HTH
Randall
Sent from my iPhone
On Nov 24, 2011, at 2:36 AM, Rynard Coetzee
2011 19:16
To: Geoghegan, Stuart
Cc: OSL
Subject: Re: [OSL | CCIE_Voice] AAR into CUE CFA
Hi Stuart - try checking redirecting number outbound on your HQ gateway - if
still an issue can you post a debug isdn q931 on hq and br2 gateways?
hth
Brian
On Fri, Nov 4, 2011 at 5:59 PM, Geoghegan, Stuart
Hi All
I am working with my lab using a centralized dialplan an my BR2/Site-C phones
have CUE (integrated with CUCM).
Everything works fine under Jtapi control and also using SIP when falling back
to SRST.
If I call from HQ to Site-C when there is not enough bandwidth and AAR kicks
in, again
Hi Stuart - try checking redirecting number outbound on your HQ gateway -
if still an issue can you post a debug isdn q931 on hq and br2 gateways?
hth
Brian
On Fri, Nov 4, 2011 at 5:59 PM, Geoghegan, Stuart
stuart.geoghe...@ngbailey.co.uk wrote:
Hi All
** **
I am working with my lab
Hi,
I am adding the AAR group in CUCM. The name is CUCM. Do I need to put anything
in the prefix digits within CUCM?
Is there a rule as to when you would add something? Why/when would you add
something?
Thanks,
Randall___
For more information
The AAR GRP can be empty if you have the full e164 number in the external
number mask of your internal DN's. You can then create Route Patterns beginning
with the + sign.
If you had say 10 digits External Number Masks on your DNs then you could use
the AAR GRP prefix to insert an access code
Location can always be assigned at the Device Pool level for both the phones
and gateways. It is not necessary to set the Location at the device/gw level.
Vik Malhi – CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto:
: [OSL | CCIE_Voice] AAR
Location can always be assigned at the Device Pool level for both the phones
and gateways. It is not necessary to set the Location at the device/gw level.
Vik Malhi – CCIE #13890
Managing Partner - IPexpert, Inc.
Telephone: +1.810.326.1444 ext 420
Fax: +1.810.454.0130
Mailto
Hello Experts
I have a question, when using AAR group the Location assignment for each one IP
phones and gateways or only is necessary to add in device pool?
Thanks,
La información incluida en este mensaje y sus anexos es CONFIDENCIAL y para USO
EXCLUSIVO
connection greeting was heard,
Kind regards
Stuart
From: CCIE For ME [mailto:cciefo...@hotmail.com]
Sent: 16 September 2011 18:15
To: Geoghegan, Stuart; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR and Redirecting IEs to play correct CUC
usergreeting
make sure the calling party
Hi All,
I am working on Vol 2 Lab4 and specifically the AAR requirement for a calling
arriving from the PSTN to BR1 phone 1002 when there is WAN congestion.
We are asked to redirect the all to voicemail and you should hear the
appropriate subscriber greeting
I am confident that my AAR config
make sure the calling party mask on the hunt pilot is set to “” and set the
AAR group and mask on the voice ports
From: Geoghegan, Stuart
Sent: Friday, September 16, 2011 9:52 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR and Redirecting IEs to play correct CUC
Hi,
I'm testing AAR.
When I call from HQ Phone1(5001) to BR1 Phone1(1001),
it seems that AAR is invoked, but call is not routed through PSTN.
Does anyone know how to resolve this issue?
Following is my troubleshooting.
- On HQ Phone1, Network Congestion. Rerouting. message was shown.
- I
) sfuna...@cisco.com
Date: Thursday, August 11, 2011 9:33 am
Subject: [OSL | CCIE_Voice] AAR Issue
To: OSL Group ccie_voice@onlinestudylist.com
Hi,
I'm testing AAR.
When I call from HQ Phone1(5001) to BR1 Phone1(1001),
it seems that AAR is invoked, but call is not routed through PSTN.
Does
...@sch.bme.hu]
Sent: Thursday, August 11, 2011 4:59 PM
To: Satoshi Funabashi (sfunabas)
Cc: OSL Group
Subject: Re: [OSL | CCIE_Voice] AAR Issue
I think you mean BR1 Phone1's AAR Dest Mask is +1617863, right?
HQ gateway is registered and working properly in all other cases but AAR?
HQ Phone1's
hi ,
- AAR will not work for TEHO ,right ?
( when we in AAR situatuin, will the call come out thru other GW
ie : HQ and SC in AAR mode, and for 900.! -- RL- HQ-SC , when go
thru SC will work or not ? )
___
For more information regarding
HI,
I'm playing with mgcp gw and cuc and AAR. I have some issues every other time.
when I call from pstn to br1 ph2 call goes to vm as expected.
When I lower the locations cac I need the call rerouted to vm. I am getting a
busy signal.
I assigned the aar groups and css's
can someone explain
I worked this out. I rushed my whole setup to work on this.My dialing plan was
jacked.
Stressing my 1st lab is coming up shortly
Randall
From: Randall Crumm rrcr...@yahoo.com
To: Online Study ccie_voice@onlinestudylist.com
Sent: Saturday, June 4, 2011 3:07
You may have two locations in US and one in Europe. They dial differently
Sent from my iPhone
On Apr 21, 2011, at 9:56 PM, Erwan Erwan e_er...@yahoo.com wrote:
hi all,
I am watching Vik recorded on CCIE Voice strategy, dicussing AAR
and one of the gotcha is : Single AAR Group if use
gateway.
I think it should work.
Rgds,
Naoufal
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal Priego
Sent: Monday, April 18, 2011 7:51 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR question
Hello All,
I
:* Monday, April 18, 2011 7:51 AM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] AAR question
Hello All,
I have a question
i was doing the workbook 2 lab 7
and i had a questions that involved AAR
and i got stuck
br1, br2, hq were all ucm sites
when
Don't forget to add external phone number mask to CTI ports and also transcoder
in your BR2 for CUE
From: Cristobal Priego [mailto:cristobalpri...@gmail.com]
Sent: Tuesday, April 19, 2011 7:56 AM
To: Naoufal Kerboute
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] AAR question
Hi,
I see if we have + dialing, we can do AAR with only one group
Only the TEHO calls fail to go through WAN
For example,,
We allow 5-calls for HQ-BR1 WAN
and a 6th call: HQ phone calling BR1 PSTN through WAN
and the WAN bw is allready have the allowed 5-calls
How can I let the TEHO
Hi,
With TEHO follow traditional call routing fail over approach, with route-list
having as a primary RG remote TEHO gw, and as a secondary SLRG or even other
specific RG.
And of course you need locations assigned to gw either directly on a gateway or
through a DP, keep in mind DP location
thanks a lot for your reply
I was thinking we need a redundant gateway in the RL
But this way, AAR isn't triggering this call
Only we found bw limitation, so we route the call through PSTN , not AAR
Also do you see it is good to assign location to DP only and not care for any
other place
...@hotmail.com
Sent: Sun, Apr 17, 2011 12:31:50 Am
To: OSL Questions ccie_voice@onlinestudylist.com
CC:
Subject: [OSL | CCIE_Voice] AAR with TEHO
Hi,
I see if we have + dialing, we can do AAR with only one group
Only the TEHO calls fail to go through WAN
For example,,
We allow 5-calls for HQ
Hello All,
I have a question
i was doing the workbook 2 lab 7
and i had a questions that involved AAR
and i got stuck
br1, br2, hq were all ucm sites
when the bandwidth between br2 and the other 2 sites (Br1, hq) wasn't enough
to make the call AAR should kick in
the br2 phones had a mailbox
hi all,
Just wondering if AAR can be applied to call from UCM to CME (thru
Gatekeeper)
HQ Phone (UCM) WAN (Gatekeeper)--CUCME (BR2 phone)
if yes, how to achieve that ?
I only know AAR can be applied to UCM to UCM ( HQ phone in UCM call to
BR2 phone
Hi Experts,
I have an issue with AAR when a line is associated with a remote destination
profile.
When I call during network congestion from a phone at BR1 to the phone at HQ
which has a RDP associated I receive busy.
Another phone of HQ can be called during network congestion, if I remove the
...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Kalyan iyer
*Sent:* Friday, February 11, 2011 11:24 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] AAR - Voicemail problem
Hi folks,
I was working on the bootcamp- lab2. Here
...@onlinestudylist.com] *On Behalf Of *Kalyan iyer
*Sent:* Friday, February 11, 2011 11:24 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] AAR - Voicemail problem
Hi folks,
I was working on the bootcamp- lab2. Here is the scenario.
HQ is a H323 GW. BR1 is a MGCP GW
Hi folks,
I was working on the bootcamp- lab2. Here is the scenario.
HQ is a H323 GW. BR1 is a MGCP GW.
When AAR is invoked, I call from HQ(5002) to BR1 (1002), call gets to BR1
phone and rolls over to voicemail. I am able to leave a message and MWI is
turned-on.
The problem is when I dial the
Hi,
here is what I think happens (can't check as I'm out of the lab for the
moment)
* AAR inserts 5600 (VM number) as redirecting number because this is how it
works internally
* CUC receives the call and applies rules for forwarded calls - attempt
sign in for VM user with mailbox 5600. it
Please try that and feed us back.
Best regards,
Tamer Ismail
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Kalyan iyer
Sent: Friday, February 11, 2011 11:24 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR
I have UNITY SIP integration with CUCM and I am trying to get AAR working.
When there is network congestion, my call from PSTN still uses WAN link for VM.
As SIP trunk appears to be a off-cluster calls for UCM, I added another RG
(local GW BR1) in RL for VM pilot but it still uses WAN link.
I got a quick question.
Phone A calls phone B and ring no answer to voicemail. Due to insufficient
bandwidth on the WAN, AAR was triggered to get to the voicemail pilot.
In this case, which CSS is taking effect?
A) Phone A's AAR CSS?
B) Phone B's Call Forwarding CSS?
C) Phone B's AAR CSS?
Sent: vendredi 28 janvier 2011 14:48
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR to voicemail?
I got a quick question.
Phone A calls phone B and ring no answer to voicemail. Due to insufficient
bandwidth on the WAN, AAR was triggered to get to the voicemail pilot
Hey Michael,
To me it has got to be Phone A's AAR CSS.
Romain
On Fri, Jan 28, 2011 at 8:48 AM, Michael Luo hout...@gmail.com wrote:
I got a quick question.
Phone A calls phone B and ring no answer to voicemail. Due to insufficient
bandwidth on the WAN, AAR was triggered to get to the
:* ccie_voice@onlinestudylist.com
*Ämne:* Re: [OSL | CCIE_Voice] AAR display
Never mind. It's the redirect IE checkbox on outbound gateway.
Interesting thing was, this only applies to MGCP but not H323 gateway.
On Fri, Jan 21, 2011 at 5:56 PM, Michael Luo hout...@gmail.com wrote:
By the way
--
*Från:* Michael Luo [hout...@gmail.com]
*Skickat:* den 22 januari 2011 01:57
*Till:* ccie_voice@onlinestudylist.com
*Ämne:* Re: [OSL | CCIE_Voice] AAR display
Never mind. It's the redirect IE checkbox on outbound gateway.
Interesting thing was, this only applies to MGCP
I was doing IPExpert volume 1 lab 6 task 6.1 - AAR (Automated Alternate
Routing).
Per the proctor guide, the configuration was pretty simple. I got it worked
but observer strange behaviors.
When HQ Phone1(5001) calls BR1 Phone1(1001) via AAR, the display on BR1
Phone1 was From HQ Phone1, which
By the way, in CCM trace:
01/21/2011 16:48:03.289 CCM|StationD:(009) (2,100,9,20) CallInfo
callingPartyName='BR1 Phone1' callingParty=8631001 cgpnVoiceMailbox=
alternateCallingParty= calledPartyName=''
calledParty=12123945001 cdpnVoiceMailbox=
originalCalledPartyName=''
Never mind. It's the redirect IE checkbox on outbound gateway.
Interesting thing was, this only applies to MGCP but not H323 gateway.
On Fri, Jan 21, 2011 at 5:56 PM, Michael Luo hout...@gmail.com wrote:
By the way, in CCM trace:
01/21/2011 16:48:03.289 CCM|StationD:(009)
Hello,
Ensure that when a user at hq site can call the cue voicemail pilot and leave
also extension 3003 a voicemail when there is a wan congestion on br2
Meaning that br2phn2 has been configured with snr, I observed that AAR didn't
work as I called from hq phone to br2 phone 2.
But when
Hello,
With Unity Connection and UCM SIP integration, what would be the best known
way to reach VM pilot number from Branch Phone when AAR is triggered? Both,
for messages button and call forwarding to VM? Pls refer to a doc if any.
thanks, Natan
___
Hi,
I have no doc reference, but you can do this using a Route List with your
SIP trunk to CUC as a first option and Local Route Group (+digit
manipulation) as a backup. works fine.
AAR is only of on-cluster endpoints, whereas CUC is considered external
system when integrated using SIP.
regards
Yes indeed. Thanks for reminding...
On Fri, Jan 7, 2011 at 6:34 PM, Miron Kobelski findko...@gmail.com wrote:
Hi,
I have no doc reference, but you can do this using a Route List with your
SIP trunk to CUC as a first option and Local Route Group (+digit
manipulation) as a backup. works fine.
Hi Experts,
working on an AAR scenario, my aar was being invoked before i brought down my
locations CAC to 23k, i think i am missing something critical here,
Please if you get what am saying or have experienced it before kindly help me
out with how to resolve this issue.
Pithog oil
I am working on lab 7 and trying to get AAR setup. As far as I can tell
it is working. But when I reduce the location to a low level, the call
reroutes out the pstn and I see it come into br2 hits the correct
dial-peer, the debug ccsip tells me 200 ok, but I hear dead space. Wait
30secs and the
Hi all,
I have an issue with AAR to VM over the PSTN.
Problem is i just get a busy tone after 21 seconds when it trys to forward
the call onto VM.
If i press the messages button it dials out to VM so the config seems right.
Config below:
CSS-ALL - pt-internal - \+!
AAR GROUP - VM - no prefix
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] AAR and ANI formatting
Hello,
I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR
correctly kicks in for different types of calls (between HQ and BR1, direct
calls to VM from BR1, for incoming PSTN calls to BR1
for separate
call routing?
Thanks in advance,
Paul (#16842 RS/Sec)
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of kobel
Sent: Sunday, June 20, 2010 11:02 AM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] AAR
Hello,
I'm playing with AAR and VM. When out of bandwidth condition occurs, AAR
correctly kicks in for different types of calls (between HQ and BR1, direct
calls to VM from BR1, for incoming PSTN calls to BR1 forwarded to voicemail
in HQ). It seems that the configuration is ok.
But I've an
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