folks,
When we order "Full Install / Upgrade media with License " thru PUT (Product
Upgrade Tool).
- Does it means our existing cluster license will no longer valid once Cisco
send us the Media and license?
- Can we download UCM , CUC, UCCX "fresh install media" thru eDelivery PUT ?
-I assu
ver a lot of things didn?t work like Ringlists, Extension
>> Mobility, and Corporate Directory. TAC had to gain root access to CCM
>> and PCD was the primary account listed for all the services. TAC had to
>> change each entry to CCM and then things started working. I spent ab
try to
>>CCM and then things started working. I spent about 8hrs with Cisco TAC
>>working on various issues listed with PCD. I think PCD is a great concept
>>but Cisco needs to work out the bugs. I would use caution if you are going
>>to use PCD. Hope this helps.
>&
folks,
we have plan to upgrade UCM, CUC, Presence from version 8.6 to version 10.5
(around 1000 users)
And we are using new server upgrade.
- I am looking for advice on detailed steps ?
-what is the best scenario to minimize the downtime ?
- do I need to use PCD ?
tks,
K
folks,
How is the dial behaviour in SCCP and SIP phone that register to same UCM.
- SCCP phones dial following digits to other SCCP phone : 2 0 0 << 0 3
Assumption : UCM will see separate SCCP message 2 0 0 << 0 3 (one by one)
- SIP type B dial following digits to other SCCP phone (SIP
folks,
we have following 2 IP phones registered to same CALL Manager.
And there is "active call" between Phone A <> Phone B
Below are the "Line Level " setting of both Phones
--
1. Phone A
User Hold MOH: Audio 2
Network Ho
folks,
what is the GUI for CUE to access mailbox ?
http://x.x.x.x/userorhttp://x.x.x.x/inbox
tks,
___
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tks, since what year Cisco require CCIE-voice for UC specialization?
From: Sam Wilson
To: Karen Johnson ; "ccie_voice@onlinestudylist.com"
Sent: Monday, December 2, 2013 2:53:21 AM
Subject: RE: [OSL | CCIE_Voice] ccie voice for UC specialization
Hi Karen
Per the Master UC Spec
folks
does anyone know when cisco start to require ccie voice as partner Master UC
specialization ?
K
___
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in all attempts I couldn't sleep the night before, drink wine will help.
also no study at all at night before exam, just watch movie or spoil yourself.
wake up early in morning on the exam day and review quickly the notes
do not wait for version 9 , I am pretty sure very less will pass on first
s any CCIE Written Exam, and hold a complete CCIE for
any track, its automatically recertified for next 2 years as well as any other
Certs one is already holding.
Regards
From:ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Karen Johnson
Se
but how CISCO verified if our skills is up to date if only re-certified one ?
From: Ramcharan Arya
To: Karen Johnson
Cc: Josh Petro ; "ccie_voice@onlinestudylist.com"
; sanity insanity
Sent: Monday, October 7, 2013 11:32:50 AM
Subject: Re: ccie written
Hi Karen,
When I pass
So I assume if you hold full CCIE RS and VOICE, in the 2 years time, you only
need to write either one.
From: Ramcharan Arya
To: Karen Johnson
Cc: Josh Petro ; "ccie_voice@onlinestudylist.com"
; sanity insanity
Sent: Monday, October 7, 2013 11:32:50 AM
Subject: Re: ccie wri
hi Arya and all,
when you have 2 ccie specialization, do you need to write WRITTEN exam for both
or just one ?
K
From: Ramcharan Arya
To: Josh Petro
Cc: "ccie_voice@onlinestudylist.com" ; sanity
insanity
Sent: Wednesday, October 2, 2013 6:59:08 PM
Subject: Re: [OSL | CCIE_Voice] Presenc
oh , what workaround?
From: Edgar Feliz
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Sunday, September 15, 2013 6:03:47 PM
Subject: Re: [OSL | CCIE_Voice] OWLE lab 4 - Voicemail issue
If you are using the IPX rack there is a known bug you are hitting.
Edgar
hi folks.
in OWLE lab 4, i can't get the VM when I left the messasge from PSTN phones to
HQ and SB phones.
any parameter to check ?
K
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Are you a CCN
how about for MVA ?
From: LorenzLGRC
To: Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Sent: Monday, September 9, 2013 5:14:09 AM
Subject: Re: [OSL | CCIE_Voice] Associate user in DN level
Yes for presence on cupc
Il giorno 23/ago/2013 23:59, "
2702
After I run CET , i can not create the same CTI again, it will create CTI 2703
and 2704
Do you know how to make same CTI after we run CET ?
K
From: Anthony Holloway
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Saturday, S
Multiple Call/Call Waiting Settings on Device SB1
Note:The range to select the Max Number of calls is: 1-2
Maximum Number of Calls
Busy Trigger (Less than or equal to Max. Calls)
what is the use "busy trigger " in RDP here? cause it always override by
Phone DB "busy trigger" . Even t
all,
does anyone know why login using "uccxadmin' always failed in UCCX server. ( I
load all OWLE lab from proctorlabs)
I have tried with password : c1sc0123, cisco, ccievoice.
"Security exception thrown authenticating User uccxadmin. Authentication
failed:
Cannot authenticate user: uccxa
all,
what is the cause call using "csim start 911" success, however call using phone
in correct DP and CSS failed and got busy tone.
- debug isdn q931 saying "unllocated number" (but it is there)
- sh isdn stat = Frame relay established
- int vlan 302 also UP and GW registered in UCM as H323
folks.
what could be the cause, if HQ never regiser to PSTN
PSNT : ntp master 2 (i chage to 6 or whatever still same)
HQ : ntp server (point to lo of PSTN)
K___
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www.
folks,
in exam,where to find CUPC port quickly to config the QOS lan ?
K___
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all.
when i do OWLE lab 4 from Proctorlabs, I need to use Voicemail Mask ,
otherwise pressing VM will go to General Greeting.
Do you know which service paramaeter or setting they put that caused this ?
K
___
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all,
when in srst mode, i left voice message to 4001 SC ,and checl the message and
delete the message. MWI is disappeared from 4001. However still show "message
Envelope " blinking on 4101.
is that expected? and how to get rid of it?
K___
For m
build home lab Amit
From: Amit Sharma
To: CCIE Study
Sent: Monday, August 26, 2013 6:44:55 AM
Subject: [OSL | CCIE_Voice] cuc vm issue with hq user..
guys...when i working on ipx racks...
i am facing issue...
i can access cuc work on it...
when leave vm from pstn or end user to nay cuc use
yes you need Xfrom,
From: "Barrera, Hugo"
To: Bill ; Edgar Feliz
Cc: "ccie_voice@onlinestudylist.com"
Sent: Sunday, August 25, 2013 4:01:05 PM
Subject: Re: [OSL | CCIE_Voice] Mailbox Transfer
What I’m trying to understand is what if the proctor dials from anyone of the
lines from the PST
is it mandatory to assign user in DN level for all phones?
K___
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all,
i am just curious, my MOH SB for multicast always work without "ip
multicast-routing" and "ip pim dense-mode"
do we really need enable MCAST for moh to work ? And why without Mcats it
also works ?
K___
For more information regarding in
hi Justion,
then I after modify and before save, i also need to remove "srst auto mode prov
all" under telephony service, right ?
K
From: Justin Carney
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylis
Thanks Justin very comprehensive, yes I noticed also that isnd bind is missing
if save.
I will choose to use SRST all then modify ,save me some time in labs.
K
From: Justin Carney
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com, (ccie_voice@onlinestudylist.com)"
Sent:
folks,
when we have to use "srst auto mode provision all" and I need to change
"Desctiption " of ephone-dn
After come back to Normal mode. do we need to save the config or just leave
it?
K___
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hi Ragu,
I also confused about this if CIsco want to see use 1 or not. However not to
conflict it is very easy, u just need to use Translation and change your Remote
destination#
Other who got 100 score on this, can pls advice?
K
From: Ragulan Sathasivan
To: "ccie_voice@onlinestudylist.co
hi all.,
anyone face this before, call from 4001 to 3001 thru GK in SB SRST mode/Cfur,
most of time show as 4001 (instead of +19723033001).
But sometime, i will show as +19723033001.
And looks like this is caused by Phone 3001 is ringing, however CFUR is not
kick in, that is why only sho
folks,
I try to configure MOH using multicast for SB site.
here is what i did :
=
SB router:
--
call-manager-fallback
moh music-on-hold.au
multicast moh 239.1.1.1 po 16384 route 142.1.65.254 142.102.65.254
max-dn 20
max-ephone 10
ip so
Karen,
To stop the live record session you should press the live record softkey again
and it will end the recording and send to voicemail. If you just disconnect
the recording will continue.
--Michael
Message: 6
Date: Fri, 9 Aug 2013 07:27:47 -0700 (PDT)
From: Karen Johnson
To: "ccie_
l
Message: 6
Date: Fri, 9 Aug 2013 07:27:47 -0700 (PDT)
From: Karen Johnson
To: "ccie_voice@onlinestudylist.com"
Subject: [OSL | CCIE_Voice] LiveRecord
Message-ID:
<1376058467.46146.yahoomail...@web163906.mail.gq1.yahoo.com>
Content-Type: text/plain; charset="us-ascii&
yes i also question about the "greeting time" , not sure if they do anything.
From: IE Target
To: heshamcentr...@gmail.com; Karen Johnson ;
"ccie_voice@onlinestudylist.com"
Sent: Saturday, August 10, 2013 10:39:03 AM
Subject: BACD limit max 2 call
ccm-manager music-o
when we shut Gatekeeper, it always take time to go back up.
is there any command to speed it up ?
K___
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thanks, yes this work
From: Hesham Abdelkereem
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Friday, August 9, 2013 6:22:38 PM
Subject: Re: [OSL | CCIE_Voice] BACD limit max 2 call
Hi,
Thats the way you do it to f
all,
is there a way to limit so BACD can only accept 2 call ?
i have used
-max-conn under dial-peer
-param queue-len under sript app-b-acd
however it still play " Thanks for calling " then reject the call.
Can we achieve rejecting call right away, without play "Thanks for calling" ?
K
__
hi folks.
After I press Live Record and press disconnected to end conversation , why the
Live Record session still stay?
is this expected or any configuration we need?
K
___
For more information regarding industry lea
cool stuff , i will try that
From: Vik Malhi
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Wednesday, August 7, 2013 10:26:34 PM
Subject: Re: [OSL | CCIE_Voice] Calling Search Space
The simplest way is the best way- I would set the CSS on the Device Pool und
IOS mtp, i understand the explicit mtp from ucm.
but this must used IOS mtp , because if I do not put MGRL that only contacin
IOS MTP and not check MTP in trunk, i will not work as Early.
K
From: Rafael Chavantes
To: Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Sent: Wednesday
all,
when i check MTP in SIP trunk and use MRGL, call become SIP Early offer.
However why if I do " sh sccp connection " , I did not see SIP Early offer use
my MTP resource at all?
K___
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to display MASK +1972303 in this case.
Advice pls ?
K
From: IE Target
To: Karen Johnson ; "ccie_voice@onlinestudylist.com"
Sent: Sunday, August 4, 2013 8:26:39 AM
Subject: Remote Dest number MVA
Hi karen
No need to change 525 to 9525
Use full Match to capture all
thanks,
yes MVA thru PSTN only show mask of Route pattern (not even 3001) , for ie:
call 911 will show as 9723033001 , call to Local show as 3033003
is there exam requirement to make it show as NAME or 3001 ?
K
From: IE Target
To: Karen Johnson ; "ccie_voice@onlinestudylis
folks,
in exam , is it safe to just use CSS on DN level ? I can't think of why we
need CSS for Phone level, except AAR CSS
K___
For more information regarding industry leading CCIE Lab training, please visit
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Are you a CCNP or CC
hi folks,
can anyone share best practice for configuring RDP 3001 and 525 PSTN in
exam ?
- do we need to create Translation rule to change 525 to 9525 or just
use Full match to capture all digit we get from PSTN ?
- Busy triiger for 3001 , do we need to set to 1 ?
- any other
yes RSVP only in phones,
i will assign for UCCX, i never assign in exam, but always got 100
From: IE Target
To: Karen Johnson ; "ccie_voice@onlinestudylist.com"
; ieravin...@gmail.com; "datapack...@gmail.com"
Sent: Friday, August 2, 2013 8:46:06 PM
Subject: bacd --- 2
all,
do we need to set ntp for uccx server in exam ? and how to do it ?
K___
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all,
only assign to phones for Location.. DO not need to put to DP or others port.
I will create problem
K
From: ie ravindra
To: Amdo Ngawa
Cc: "ccie_voice@onlinestudylist.com"
Sent: Friday, August 2, 2013 7:29:25 AM
Subject: Re: [OSL | CCIE_Voice] CAC RSVP Not enough bandwidth tip
Tha
all,
is this the parameter for limiting 2 call in queue ?
service app-b-acd
param queue-len 1 > need to change to 2 for max two call in queue
param aa-hunt1 4899 ! ephone-hunt DN
param queue-manager-debugs 1
param number-of-hunt-grps 1
Also , for dial-peer VOIP that point to this
ok thanks for info
From: Anthony Holloway
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Wednesday, July 31, 2013 2:14:11 PM
Subject: Re: [OSL | CCIE_Voice] UCCX agent ready state
I would say never to use it. It's a t
Thank you, so in exam better to use this or not ?
From: Anthony Holloway
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Wednesday, July 31, 2013 1:29:01 PM
Subject: Re: [OSL | CCIE_Voice] UCCX agent ready state
There i
folks,
in lab , what is the option aside from typing this "cue-vm-license_12mbx.pkg"
to "software clean install "
can we copy paste the file name so we not mis-type here? Do they have ftp
client in the student PC?
K
___
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ah ok great thanks
From: "sbar...@mystictraveler.net"
To: Karen Johnson ; "ccie_voice@onlinestudylist.com"
Sent: Wednesday, July 31, 2013 12:18:50 PM
Subject: RE: [OSL | CCIE_Voice] UCCX agent ready state
That is expected behaviou
all,
when i call 4000, and it will ring agent 4101. if I did not pick up the call,
call will go to "Not Readt" state
However it never come back to "Ready state"
I have config CSQ and Resource Section in UCCX to "Automatic Work = Enable"
but still it won't go to Ready state until we manu
hi all,
when i call MVA number 3300 and press 1 to dial internal number, it show as
3001.
however when i dial 911, it show as 9723033001 (not as 3001).
how to achieve so it show as 3001 (RDP) ?
K___
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hi Mike,
I upload to Pub as I do not have Sub. I tried to troubleshoot but no luck, so
just reset all my session and do from scratch, now it works.
But the thing that confused me, file is in the TFTP server, i verified by "file
view tftp Ringlist.xml" . I have restart TFTP and UCM few times.
hi Bill,
thanks, yes i saw that. Accurate mean? i copy from existing ringlist and
modify.
I only use PUB as TFTP and UCM (no SUB)
K
From: William Bell
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Tuesday, July 23, 2013 11:42:12 AM
Subject: Re: [OSL |
i have added Ringlist to PUB tftp and restart the TFTP service, even PUB
callmanager
and when i show " file view tftp Ringlist.xml" file is there
but when i press thru phone, it said can't find it .
any missing ?
K___
For more information regarding
all,
i have config "Dir services" in UCCX and it is accessible from web by type:
http:x.x.x.x/service.xml
however when i go to phone and press the Dir button, it said : Http 404,page
not found.
any idea ? i have restart IIS and add permission as well.
K_
hmm but did work for me , i used 7965 phone. the time still same like
without press speaker
From: Nadeem Ahmed
To: Karen Johnson ; "ccie_voice@onlinestudylist.com"
Sent: Thursday, July 18, 2013 9:32:06 PM
Subject: SRST connection monito
ccie_voice-ow...@onlinestudylist.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of CCIE_Voice digest..."
>
>
> Today's Topics:
>
> 1. SRST connection monitor duration (Karen Johnson)
> 2. IPPM service
yes i did this , but still same. really strange.
From: Randall Crumm
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Wednesday, July 17, 2013 9:10:58 PM
Subject: Re: [OSL | CCIE_Voice] IPPM service failed
You need to create an a
CTI Allow
control of All Devices
I did above, but still same error
K
From: Randall Crumm
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Wednesday, July 17, 2013 6:34:48 PM
Subject: Re: [OSL | CCIE_Voice] IPPM service failed
Do
folks,
i got strange issue in my IPPM service.
I verified using "System troubleshoot" in CUPS all looks good and i have config
as per cisco doc. Also the url is verified correct.
- When i press IPPM service thru phone at HQ and SB sites, it gave me error:
"You were trying to access IP Ph
all,
if we need to speed up srst testing (switchback between normal and srst).
- how low can we change "Connection Monitor duration" in Device Pool
- is there any other better way or IOS command ?
K___
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all,
when i press VM button in SC phone, it ask for pin. And i press 12345 .
this work fine in normal mode (CUE register to UCM)
but if i switch to SRST mode, it also prompt for pin, just when i press 12345,
they never detected it.
any config i missed in CUE ?
K___
thanks all, work great now
From: Ashok Boinpally
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Wednesday, July 17, 2013 12:42:35 AM
Subject: Re: [OSL | CCIE_Voice] srst ephone-dn
Hi Karen,
"srst autoprovision none" c
all,
when i configure "srst autoprovision none" and create "ephone-dn 4" and
"ephone-dn 5"
it saying : "ephone-dn 4 has been allocated for srst virtual dn, and it can not
be modified"
however there is no ephone-dn configured for 4 and 5 and no ephone registered.
- which resource that use m
all,
anyone know how to config the RSVP CAC to allow max call, first time right ?
Is there any steps sequence that i can follow to avoid unnecessary
troubleshoting?
K___
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yes doesnt work
From: ccie2k12
To: 'Karen Johnson' ; michael.se...@compucom.com;
ccie_voice@onlinestudylist.com
Sent: Monday, July 15, 2013 3:09:00 PM
Subject: RE: [OSL | CCIE_Voice] Conf Meetme did not disconnect.
have u tried it?
"Drop Ad
hi Mike and Tom,
Thanks, I will, just prepare if they ask me to drop.
K
From: "michael.se...@compucom.com"
To: ccie_voice@onlinestudylist.com
Cc: karen.johnson...@yahoo.ca
Sent: Monday, July 15, 2013 7:10:03 AM
Subject: Re: Conf Meetme did not disconnect.
you mean we have to disconnect ourselves ?
From: ccie2k12
To: 'Karen Johnson' ; ccie_voice@onlinestudylist.com
Sent: Sunday, July 14, 2013 6:21:41 PM
Subject: RE: [OSL | CCIE_Voice] Conf Meetme did not disconnect
don't think thats possible,
ah tks make sense ,will try tomorrow
From: Tom E
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Sunday, July 14, 2013 8:33:49 PM
Subject: Re: [OSL | CCIE_Voice] Conf Meetme did not disconnect
Go to CUCM service parameter &q
folks,
when the Host disconnect from Meetme 2500, other phones who joined did not
disconnect.
is there any service parameter/config need to be done ,so it can also
disconnect?
K___
For more information regarding industry leading CCIE Lab training, p
.
From: Bill
To: Karen Johnson
Cc: ccie2k12 ; ikizoo hello ; William
Bell ; "ccie_voice@onlinestudylist.com"
Sent: Sunday, July 14, 2013 4:32:38 PM
Subject: Re: [OSL | CCIE_Voice] cups best practice
Ok so you are willing to risk a 0 score to avoid little work configuring a SIP
yes i heard rumors some get score without it , but also lots said needed trunk.
maybe variation
it works without sip trunk.
From: IE Target
To: ikiz...@hotmail.com; b...@ucguerrilla.com; Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Sent: Sunday, July
just try it first :) it just work without it
From: ccie2k12
To: 'Karen Johnson' ; 'ikizoo hello'
; 'William Bell'
Cc: ccie_voice@onlinestudylist.com
Sent: Sunday, July 14, 2013 4:34:05 AM
Subject: RE: [OSL | CCIE_Voice] cups be
hi ikizo,
i heard there is variation that asked not to use SIP trunk for Cups, maybe you
missed that.
Can someone confirm ?
K
From: ikizoo hello
To: William Bell ; Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Thursday, July 11,
Yes date group and dp is set. But hq and sb show same time, which Hq is pst and
sb is cst___
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Mike,
What is best practice for GW HQ, SB,SC in exam ? is this configuration below
enough for all 3 GW?
interface Serial0/1/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn bind-l3 ccm-manager
isdn send-alerting
isdn sending-complete
isdn outgo
hi all,
UCM did not even trigger the HW MTP or Xcode in my HQ router, when I make SIP
call using Early offer.
SIP trace showing sdp inside of INVITE and 200/OK
is that expected ? I thought Early offer need MRG ?
K
From: FAISAL AL-EMAD
To: Bill Lake ;
Transcoder only, I thought he said MRG-SB contain below :
SB-Xcode
SB-mtp
From: Edgar Feliz
To: Karen Johnson
Cc: myfriend myfriend ; "ccie_voice@onlinestudylist.com"
Sent: Friday, July 12, 2013 2:03:44 PM
Subject: Re: [OSL | CCIE_Voice]
when we do : file tail activelog /cm
- what command to do HELP, if we forgot what is the next directory ?
K___
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Are you a CCNP or CCIE and looking for
hi friend,
may i see your IOS config for this ?
and also what in inside MRG-SB and MRGL-SB
you only assign MRGL SB to GW h323, right ?
K
From: myfriend myfriend
To: Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Sent: Friday, July 12, 2013 12:45:35
it is SCCP phones
From: Martin Sloan
To: Karen Johnson
Cc: "ccie_voice@onlinestudylist.com"
Sent: Friday, July 12, 2013 6:49:54 AM
Subject: Re: [OSL | CCIE_Voice] clock on HQ and SB Phone
Just a guess but if they are SIP phones and you don
tks ,
so what is the Transcoder use for Fast start here?
K
From: myfriend myfriend
To: Karen Johnson
Cc: ccie_voice@onlinestudylist.com
Sent: Friday, July 12, 2013 12:45:35 AM
Subject: Fast Start
These are my finding
No need to enable fast start on
folks,
I have enable "fast start" in router SB as H323 GW.
sh call acti voice --> Fastconnect=true
-I checked MTP, Inbound Fast, Outbound Fast in H323 GW on UCM
- assign MRGL to SB DP and H323 GW.
However when I do "h225 trace" from UCM, it only gave me "Outbound Fast Start".
Questions
=
folks,
my HQ and SB phone keep showing same clock.
HQ:PST
SB:CST
both assign to DP and in routers, i can see HQ as PST time and SB as CST from
show clock.
what i missed here?
K___
For more information regarding industry leading CCIE Lab trai
folks,
I spent hours just to find out issue with my VM Callmanager Pub.
Here is what happen :
--
- Mgcp GW show not register in UCM, but "sh isdn stat" and "sh ccm-manager"
looks good in GW
(debug isdn q931 show "Bearer capability error" and "SETUP T303 error"
- I res
folks,
i need advice what is best to use between this two in exam?
- rule 1 /.*\(3...$\)/ /\1/
or
- rule 1 /.*\(3...\)$/ /\1/
tks
K___
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Are you a
Folks,
Can we achieve this with One call handler only ?
"Meet Me When joins meet me conference it should announce the participant name
and when somebody dials meet me number it should ask "who may I say its
calling". and when Meetme not avail , go to HQ1 voicemail"
Each time I try to use
yes thanks, i missed that and now work
From: Rrcrumm
To: Karen Johnson
Cc: sanity insanity ;
"ccie_voice@onlinestudylist.com"
Sent: Monday, July 8, 2013 12:10:45 AM
Subject: Re: [OSL | CCIE_Voice] Cue call failed in srst mode
Do you have a sip t
dial-peer yes, trigger mean ?
From: Rrcrumm
To: Karen Johnson
Cc: sanity insanity ;
"ccie_voice@onlinestudylist.com"
Sent: Sunday, July 7, 2013 8:08:42 PM
Subject: Re: [OSL | CCIE_Voice] Cue call failed in srst mode
Dial peer and trigger
On J
Yes___
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Hi folks,
My cue is working fine in normal mode. If i switch to srst mode it failed and
busy tone. Dialpeer and codec g711 , sip-ua ip show correct.
any idea what to tshoot and command to check?
K
___
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hesham,
i have similar issue, it was caused by failed moh files, do following:
- audio-prompt load flash:en_bacd_welcome.au
if moh failed, then download from Cisco and copy to flash.
run "audio prompt " again and files will load successfully , then it will
works as expected.
_
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