If all else fails you can set up an alternate tftp and try the firmware change
there
We have used this method to recover some older 7936 phones and it worked well
http://www.youtube.com/watch?v=690-4S2CS5Ufeature=youtube_gdata_player
Sent from my iPad
On Feb 4, 2012, at 3:36 PM, Emanuel
As long as your provider does not monitor and charge for overages ;)
Sent from my iPad
On Feb 7, 2012, at 1:42 PM, George Goglidze gogli...@gmail.com wrote:
Frame-Relay is known for it's oversubscription capabilities...
So feel free to oversubscribe... It will be fine to configure one PVC
I believe transcodes do not by default support g729r8 and this must be added
manually
Sent from my iPad
On Feb 9, 2012, at 6:56 PM, Vik Malhi vma...@ipexpert.com wrote:
I think the reason is a codec mismatch between the transcoder and mtp.
The MTP probably has g729r8 support but the
How about some configs
Bill
On Feb 26, 2012, at 1:53 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote:
Hi All,
In the five-lab ebook, lab 2, there's a requirement that HQ/BR1 phones must
receive an IP address from the DHCP server configured on HQ-RTR. BR1 phones
didreceive IP address
On switch you have no cdp adv, this command prevents the phones from getting
vlan set
No dhcp pool on router so your br1 is most likely using cached addressing
Bill
On Feb 26, 2012, at 2:54 PM, Ramy Abdelrahim ramyoth...@hotmail.com wrote:
Kindly find attached the full config of HQ-3750
phone or load?
Bill
On Mar 24, 2012, at 11:23 AM, Justin McIntyre justin.mcint...@blackbox.com
wrote:
I have the same exact problem. I am in Florida and my lab is in Virginia. I
have a VPN setup between an 1861 and a 2821(Virginia). Sccp phones function
just fine. What kind of service
Was any srst provisioning done?
Bill
On Mar 29, 2012, at 7:15 PM, Justin McIntyre justin.mcint...@blackbox.com
wrote:
Can anyone tell me why my learned ephone DNs would stay up even if I’ve
removed telephony-service, saved config, reloaded and then re-entered
telephony service back in. I
Sorry late to the party but was NBAR excluded in this config per task? If not
would it be possible to use NBAR?
Example
class-map match-any NBAR
match protocol MGCP
policy-map mark
class NBAR
set
interface
service-policy input mark
Bill
Set from iPad
On Apr 2, 2012, at 8:04 PM, steven
Let us say they want the local router to be ntp server or back up
Then y
Bill
On Jun 8, 2012, at 2:56 PM, Jason Murray murr...@usa.com wrote:
Do not use ntp master on any of your routers. The only command that is
needed is the ntp server command. Even if you want one of your routers
locally even when
external ntp is working
Bill
On Jun 8, 2012, at 7:00 PM, Jason Murray murr...@usa.com wrote:
You still only need the ntp server command. Lets say the requirement was to
sync the HQ router with the PSTN router and then all other devices sync NTP
with the HQ router
Hello
I have downloaded fileopen on iPad, then put my files on dropbox, then opened
file in dropbox, once opened, I hit the open with, scrolled down to fileopen
and was able to open the file on the iPad.
Bill
On Jun 11, 2012, at 7:41 PM, Rafael Chavantes raf...@chavantes.com wrote:
Hey
Log in you cisco account
Click on products
Click on voice and unified communications
Click on ip telephony
Now find your product, warning cisco is changing the site some so might be
different
Bill
On Jun 16, 2012, at 11:26 PM, Vikas Singhal vikising...@yahoo.co.in wrote:
Hi Dan,
I
Can you post you list of how everything is connected and the config of you hq
switch?
So say hq router fa 0/0 to hq sw port 1/0/1 and pstn fa0/0 to hq sw 1/0/24
Bill
On Jul 16, 2012, at 12:19 PM, Wilson dwil1...@insightbb.com wrote:
I have never been able to successfully ping the PSTN
Well if your phones are 7960 they do not support it
If you are calling from an h323 gw then it strips the +
need more details to he more
Bill
On Aug 31, 2012, at 1:44 PM, abollgoog ccie ccieg...@gmail.com wrote:
Hi Bill/All
I am calling from site to site; the plus show up in the status
Most Cisco UCM servers only require a 72 or 80 GB HDD so 400 should cover you 2
UCM (pub sub) CUCx, CUPS, and UCCX.
Bill
On Oct 22, 2012, at 1:22 PM, Craig Hill (crahill) crah...@cisco.com wrote:
You are a little light on HD space, to run everything. I would go with ESXi 5
and use thin
Do show inv and maybe show diag | inc PVDM
also try show voice dscp group all
Or at least I think that might help see if the dscp resources are available
Bill
On Oct 26, 2012, at 12:02 PM, Nicolas MICHEL mcl.nico...@gmail.com wrote:
Same result :(
Le Friday, October 26, 2012 6:57:30 PM
number, uncheck the box and put aar as its CSS.
Bill
On Nov 14, 2012, at 7:08 PM, Barrera, Hugo hugo.barr...@nexusis.com wrote:
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
and by trying
them you will know which one fits which need
In general you would use auto qos VoIP trust for frf12 if dscp is trusted and
without trust if not.
Bill
On Dec 24, 2012, at 11:26 PM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Bill,
Thanks for for the link
I would not use voice class codec as cue only supports g711 better to just set
codec to g711 directly. Otherwise if say a remote phone only supports g729
and you have voice class it will not invoke transcode
Bill
On Dec 28, 2012, at 7:37 PM, Amaia Lesta amaia.le...@gmail.com wrote:
Hello
So when you assign the service to the phone add those fields for each user,
also when you create the service you can set defaults so one user works right
away and you just edit the rest
Bill
On Dec 31, 2012, at 5:19 AM, singh singh8...@in.com wrote:
hi Guys,
I have configured 1 button
Directed call park in this case works only if you dial the retrieval code of
*2456 then the park number 2456
So try *24562456 or just change the retrieval code to * and dial *2456
Bill
On Jan 2, 2013, at 9:53 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote:
Hi,
Task requirement
because right now all we know is
pri, but most likely hq is a gateway for CUCM and we don't know what type. We
don't know if this is your lab, rack rental or real world. It is very helpful
to have this information to be able to give you help.
Bill
On Jan 9, 2013, at 3:38 AM, Heath Williams
are ready to go
Bill
On Jan 9, 2013, at 4:49 AM, Suresh Bhandari bring...@gmail.com wrote:
In Vol 1 Lab 11A, running in last two hours of the session.
The CUE has CCM license, and no licnese file for CME is there in the Contact
Center Remote Desktop as displayed by Vik in VoD.
I contacted
or use a softphone for now. You can also hardware VPN into
proctor labs and I find that to be the best solution for them to give me more
of a true lab experience.
Bill
On Jan 8, 2013, at 11:32 PM, Piotr Puchalski p256...@gmail.com wrote:
Nic,
I’ve been battling a somewhat similar problem
.
Hope this helps but when building a lab, I don't think the extra expense of the
e1 or e1/t1 card is justified if you already have what you require to just use
t1 cards.
Bill
On Jan 23, 2013, at 3:27 PM, Allen Chen alnc...@hotmail.com wrote:
Hi Todd,
The PSTN router has a T1 connection
Depends on what they ask
Is it system wide or just one phone?
Bill
On Jan 27, 2013, at 2:34 AM, ie ravindra ieravin...@gmail.com wrote:
Hi My Friends,
is it correct if I stripped the caller ID from the router when it asks for
hide the caller ID ? for lab exam or either what
the user to the selected resource.
Bill
On Jan 29, 2013, at 1:17 AM, sanity insanity
networksanitytoinsan...@gmail.com wrote:
hi Guys,
I am trying to create a script for the following . How achieve it?
-- when users call they should hear
“Thank you for calling” and immediately”
“After
Great walk through but just one note, this is for g729 calls which surely you
will be doing but just to be sure you don't use these numbers with g711 ;)
Bill
On Jan 30, 2013, at 7:03 AM, Justin Carney justin.s.car...@gmail.com wrote:
The two types of CAC are locations-based and RSVP-based
Sounds like you need to troubleshoot your FTP software server.
Bill
On Feb 8, 2013, at 7:49 PM, Ben John benjoh...@hotmail.com wrote:
Guys,
i am trying to do a software install for my CUE model. i can ping my ftp
server but when i tried to download the software i get the following error
Bill there is no preference to method just results so either method will work.
I prefer voice translations myself
Bill
On Feb 15, 2013, at 10:38 PM, William Bell b...@ucguerrilla.com wrote:
In OWLE Lab 4 there is a requirement to allow 4-digit dialing to Site A and
Site B from Site C
Intermittent problems are tough without the full config but have you tried
making sure it is not a hardware issue? Can you match the failed calls to
failed pings? Have you some debugs of your sip messages? The study list needs
more information to try to help with the issue.
Bill
On Feb 16
phone to its own
file on the same server. Play with it, it is the best part of learning this.
It will give a great sense of satisfaction when you can do whatever you want
with this method.
If you want more details on this look back to the discussion around 17
November 2012.
Bill
Sent from
have an answer for you. However, I can confirm that I have noticed
the same behavior. When I have associated custom tones for join/leave
events, I only hear the tone on join. Nada on leave. I haven't figured it
out yet.
-Bill
--
William Bell
blog: http://ucguerrilla.com
twitter
Adding the voice class codec in your sample in a situation you only want g729
(over the WAN) can cause this issue.
Sent from my iPad
On Feb 18, 2013, at 6:11 PM, Jason Lee jas7...@gmail.com wrote:
All,
Given a CME with Gatekeeper lab scenario, I've been configuring my inbound
dialpeer on
Hi Bill
Read the question carefully but if you can control the config it is better than
trusting something you don't trust
Bill
Sent from my iPad
On Feb 21, 2013, at 12:29 AM, William Bell b...@ucguerrilla.com wrote:
Leslie/Steve/Jason,
What are your thoughts on pre-configuring ephone
So your getting to the tutorial so that means you are getting to cue but it
doesn't see you created a pin
Are you doing the cue config by cli with username scph1 pin 12345 and voicemail
mailbox owner scph1 and enable then no tutorial? Or by web page config?
Sent from my iPad
On Feb 23,
http://docwiki.cisco.com/wiki/Cisco_Unified_Survivable_Remote_Site_Voicemail
This is not lab worthy at this time since the lab is on version 7 but something
to tuck away for later.
Bill
Sent from my iPad
On Feb 25, 2013, at 10:24 PM, Andy Thanh tanthanh2...@gmail.com wrote:
It is impossible
Does it help you, and by this I mean you specifically, reach your goal of
making your dial plan work just the way you want.
I know lots of people don't use css at the gw but I felt more comfortable using
them. Could I pass without it, yeah as long as everything works like it is
supposed to
Why don't you try creating a separate xml file store it on a web server under
wwwroot and have it list your requirements. Look back at about 17 August 2012
for a lively discussion on how to complete this.
Sent from my iPad
On Feb 28, 2013, at 10:27 AM, Hesham Abdelkereem
not sure that applies in this scenario... the signaling
port was changed on the CME router. The phones I am using to call into BACD
with are also registered to the CME site.
On Thu, Feb 28, 2013 at 11:43 AM, Bill Lake whl...@gmail.com wrote:
Right there Cory, the default is 1720 but if changed
My preference is to do as much as possible in the h323 gateway instead of on
CUCM but there is no right way as long as you meet the requirements of the
task. Remember you have to meet the demands of calling and caller info in SRST
as well.
Sent from my iPad
On Mar 2, 2013, at 11:47 PM,
and testing things you might see.
Sent from my iPad
On Mar 3, 2013, at 6:20 AM, sanity insanity networksanitytoinsan...@gmail.com
wrote:
hi Bill,
Thanks for your reply.
However with this method using local route groups for HQ( MgcP gateway ) and
Site B ( h323 gateway) with the same route
Well it looks like it is picking up the RTP header compression, shaping and
even the service policy but not doing it.
Have you tried rebooting the router or copy, delete then reapply ?
Sent from my iPad
On Mar 3, 2013, at 8:43 PM, Jason Lee jas7...@gmail.com wrote:
All,
Was wondering if
Try to use
http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html
Sent from my iPad
On Mar 10, 2013, at 7:20 AM, Josh Petro josh.pe...@gmail.com wrote:
Unplug the FXO card. That's the only way Im aware of that works every time.
Josh
On Mar 10, 2013 7:09 AM, CCIEing aboaz...@gmail.com
Remember that even if you set this on the edge it is possible to remark at the
dial peer or by policy map. This could be quite useful, say ITSP #1 uses one
setting and ITSP#2 uses another. Then you could set them differently as they
are sent. This gives you flexibility.
In this case the
First step is to ensure calls are going out the PSTN and coming in the VG. If
it. Is your lab then you can easily do this if not then SP has to do it on the
PSTN side.
You should then verify at your VG the number as it arrives. Use debug isdn
q931 to verify it is hitting your GW. Then
?
Jamie Parr
Engineer - IT
jamp...@cisco.com
Phone: +44 20 8824 2641
Mobile: +44 7590622049
From: Bill [mailto:whl...@gmail.com]
Sent: 10 March 2013 14:50
To: Jamie Parr (jamparr)
Cc: Pixar Perfect; CCIEing; Amp; CCIE Voice OSL
Subject: Re: [OSL | CCIE_Voice] H323 VG dial-peers
I believe facility ie is enabled by default and is not seen on PRI (remember
MGCP is back hauled and config is in CUCM) but you can always add it to be sure
you must also use the isdn supp-serv name calling to support name display
This can be determined in the debug isdn q931 when you see
So the number comes in a MGCP GW and you want to change it from 555-1234 to
9555-1234 is that correct?
If so, then use debug isdn q931 to verify the incoming numbers ANI and isdn
type. If it is say local/isdn then under the GW page drill down near the
bottom and prefix a 9 to the local
Can you try making the block pattern a urgent priority pattern
Sent from my iPad
On Mar 16, 2013, at 1:05 PM, vignesh sethuraman sethuvign...@yahoo.co.in
wrote:
Hello Experts,
I am working on Task 5.7 from Vol1. Question is to block the 91900? numbers.
I have configured a Route pattern
spaces.
I wouldn't think it necessary to do this. Did you select the radio button
for 'Do not route this pattern' on the 900 RP?
Thanks
On Saturday, March 16, 2013, wrote:
I have already set that and tried but no luck Bill.
Sent from Yahoo! Mail for iPhone
Can you try
dial-peer voice 102 voip
preference 2
destination-pattern 3300
session target ipv4:ip address of the CUCM Pub
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
Is site to site g711? Or is your CUCM local?
application
service cmm http://ip address of the CUCM
Ip ospf network point-to-point
And he said he can ping and session to it, web interface is not working though?
Sent from my iPad
On Mar 19, 2013, at 6:31 PM, Andy Thanh tanthanh2...@gmail.com wrote:
Did you advertise this network through ospf? This is for simple command:
router ospf 1
No Wireshark on the lab so this won't be tested for anything you can't capture
on CUCM traces in this case since you are using RTMT but my suggestion is to do
the captures with your own testing to see what appears. That is the best way
to learn and find the output.
Sent from my iPad
On Mar
Traditionally you would use the alternate extension or a on the pilot. So
if you we're denied the ability to use alternate extension for this task but
had to use it for another, say allowing easy voicemail access to a user at
home, then I think you are looking at a very specific inbound
Is your gateway registered in CUCM?
Are you getting the proper output of your show commands? Show isdn status,
show ccm
Do you have
int seri x/x/x
Isdn bind-l3 ccm
Did you try no MGCP MGCP?
Can you post more of your config?
Sent from my iPad
On Mar 22, 2013, at 4:52 PM, CCIEing
In versions 12.4(15) or newer it will have the embedded version so you should
have the embedded version on the lab.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html
Sent from my iPad
On Mar 30, 2013, at 10:29 PM, virajith vir...@rediffmail.com wrote:
Look for CCIE VOICE Alchemy by Kevin Wallace it would be very helpful and I
took this with ipexpert last year but don't see it now.
Sent from my iPad
On Apr 2, 2013, at 6:06 AM, Josh Petro josh.pe...@gmail.com wrote:
Hi everyone
Im just now near the end of lab 5 and Im going to attempt to do
I like to use ntp server ip burst iburst as it helps converge faster.
I also like to tie it to the loopback
Sent from my iPad
On Apr 2, 2013, at 5:47 AM, Vikky Kumar vikkyne...@gmail.com wrote:
hi,
If hq-router is ntp reference for branches AND hq router has some external
ntp reference.
I believe either way will give you the same results. I did not assign a
preference to my primary dial peer but I believe this to be a preference.
You can manipulate how the dial peer is used by using the command
Dial-peer hunt ?
This will show you ways to change how your dial peer is picked.
You need to verify as in the past this did not always work. Not sure how the
new scheduling works
Sent from my iPad
On Apr 15, 2013, at 10:27 PM, Mohamed Gazzaz mgaz...@hotmail.com wrote:
The two sessions are linked automatically when you book B2B sessions and your
configuration stays.
Or change the port in CUCM if not already done :)
Sent from my iPad
On Apr 23, 2013, at 10:38 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote:
Please remove port 1820 from VoIP service it will work
Sent from my iPhone
On Apr 23, 2013, at 9:54 PM, Martin Sloan martinsloa...@gmail.com
What language did you pick?
Sent from my iPad
On Apr 27, 2013, at 11:10 PM, Hesham Abdelkereem heshamcentr...@gmail.com
wrote:
Dear Experts,
The current version of CUE is 2.1.3 something like that and I was trying to
install CUE 7.0.3
However , I have downloaded all the package from
If they announce a new version in June you will have about 6 months from then
to pass your lab.
My recommendation from there is if you have plenty of study/lab time then go
for it. If you don't and have to squeeze it in then you might be better off
waiting to see.
My thought is that 600 to
Did you assign them capabilities under CUCM?
This is path. System-licensing-capabilities assignment
Sent from my iPad
On Apr 28, 2013, at 11:12 PM, Hesham Abdelkereem heshamcentr...@gmail.com
wrote:
Dear Experts,
I'd like to add contacts to CUPC Client.
However i go to
I think David Blair (proctor in RTP) is a very good proctor but more important
is that he is a good person. That means more as we all know passing a test is
important but life with those around you is more important.
Sent from my iPad
On May 1, 2013, at 9:41 PM, Pavan K pav.c...@gmail.com
Why not just create dial peers in CME to match your existing translations and
send them over to CUCM?
Sent from my iPad
On May 10, 2013, at 2:46 AM, Dharambir kumar varma dharambi...@gmail.com
wrote:
On May 10, 2013 1:02 AM, Dharambir kumar varma dharambi...@gmail.com
wrote:
Hi sir
I think it is an old version of secure CRT and not one easily found on the web.
I think something like version 3 or 4 but I really did not worry about that, I
use the current version and it works similar but don't expect much more that
very basic interface
Sent from my iPad
On May 14, 2013,
dial-peer voice 9911 pots
translation-profile outgoing 9911
destination-pattern 9911$
port 0/0/0:23
forward-digits 3
Sent from my iPad
On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote:
dial-peer voice 9911 pots
translation-profile outgoing 9911
voice translation-rule 8
rule 1 // // type any unknown plan any isdn
rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn
Sent from my iPad
On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote:
In the dial peer configure no digit strip. :)
On Sun, May 19, 2013 at
, Bill. Any thoughts on why the gw is only sending '11' to the
PSTN? If the dial-peer is stripping explicitly matched digits it should
strip all of the digits. It just doesn't make any sense to me that the
voice translation debug and test shows that the digit manipulation happens
correctly
My plan was to practice cups until I could do it as fast as possible, when I
did this, I would use the build in tools to verify everything was working.
So to start use the integration guide on Cisco's web page and repeatedly
integrate and make everything work.
Use diagnostics, especially
Did you try using this?
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/service/8_6_1/admin/sasnmpv1.html
Or just google it to get videos like this
http://m.youtube.com/#/watch?v=fxV2SP3Xmtwdesktop_uri=%2Fwatch%3Fv%3DfxV2SP3Xmtw
Sent from my iPad
On May 25, 2013, at 5:04 AM, Dharambir
If you are trying to match traffic from a single server wouldn't you want to
match on ip?
You might try a class map that does match all and have it match sip and the IP
address of the server in question.
Sent from my iPad
On May 27, 2013, at 3:59 AM, Ken Wyan kew...@gmail.com wrote:
Agreed, I passed my lab early this year. I spent a lot of time and energy
passing this lab and to have it retire and devalue is truly a disappointment.
My plan originally was to renew the first time with CCIE RS then second by
passing my RS lab. Looks like plans will change again.
Bill
?
From: Bill Lake whl...@gmail.com
To: Karen Johnson karen.johnson...@yahoo.ca
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Tuesday, June 4, 2013 11:41:11 AM
Subject: Re: [OSL | CCIE_Voice] FTP from CUE failed
Start with the simple stuff, can you ping the FTP server
Sounds like you are dropping the return packets but not sure why
What do you see in the FileZilla server?
Sent from my iPad
On Jun 4, 2013, at 5:16 PM, Karen Johnson karen.johnson...@yahoo.ca wrote:
yes i did not see when I do from CUE
but when i do from Filezilla client, I see
From:
RTMT after setting your trace level in serviceability
Sent from my iPad
On Jun 15, 2013, at 8:30 AM, Dharambir kumar varma dharambi...@gmail.com
wrote:
Hi
this is Dharambir here
i created a sip trunk on CUCM 7.0 with other cluster..
i am tryinng to make call over sip trunk..
how
/products_tech_note09186a00806d774f.shtml
It is a caching issue with the phone.
Jason Nielsen
On Sat, Jun 15, 2013 at 8:57 PM, Bill Lake whl...@gmail.com wrote:
So how do you create the Ringlist.xml? do you also ensure that you use the
exact format of the name (not ringlist.xml) and do you
Establish a working call and use your show voice commands?
Sent from my iPad
On Jun 16, 2013, at 6:30 AM, CISCO CCIE VOICE ccievoic...@gmail.com wrote:
Thanks for your reply its fine that we have select G729 codec under System
Parametes but my question was is there a way to test which
karen.johnson...@yahoo.ca wrote:
tks Robert and Bill.
so in exam i will leave to default and not worry on this : busy out if
(Inhibit re-starts at PRI initialization is checked)
From: Robert Thomas tho...@gmail.com
To: Karen Johnson karen.johnson...@yahoo.ca
Cc: ccie_voice@onlinestudylist.com
A-RTR-HQ-CCIE-V-IP-Expert-Lab-1#sh run | begin gatekeeper
gatekeeper
zone local GK ipexpert.com 10.10.100.1
zone local ViaGK ipexpert.com
zone remote backbone ipexpert.com 10.10.100.2 1719 outvia ViaGK
zone prefix backbone 011*
no shutdown
Pod1-TS-FRS-VPN-NAT-PSTN-CCIE-V-Lab-2#sh run | begin
Johnson karen.johnson...@yahoo.ca
wrote:
tks Robert and Bill.
so in exam i will leave to default and not worry on this : busy out if
(Inhibit re-starts at PRI initialization is checked)
From: Robert Thomas tho...@gmail.com
To: Karen Johnson karen.johnson...@yahoo.ca
Cc: ccie_voice
, 2013 12:10:28 PM
Subject: RE: [OSL | CCIE_Voice] cups best practice
hi Bill,
after verification VM, IM , Presence, all working fine, but still got 0 score.
only one thing, in the screenshot the user id was h...@ccievoice.com [not
h...@ccievoice.com, in CUCM it is HQ2].
do you have any idea
This is no way audio
Do a google search for it and find the answer
Sent from my iPad
On Jul 24, 2013, at 11:59 PM, Devakanth Gangavarapu devakanth2...@gmail.com
wrote:
If there is a routing issue, the CUPC would not get registered to CUCM :)
Dev
On Thu, Jul 25, 2013 at 3:50
Posting the config might help
Sent from my iPad
On Jul 28, 2013, at 4:56 AM, 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
You really can drop the subscriber as you will be missing very little
functionality
Also if you build your VMware with resource sharing it will run better, I
actually found that spreading the servers over multiple disks and NICs
Hope this helps
Sent from my iPad
On Jul 28, 2013, at 7:21 PM,
Could be hitting this
CSCsd14203
Symptom : BACD AA script crashes while trying to play Music on Hold (MoH)
Conditions : This happens with 12.4 (4) XC and 12.4 (4)T images.
Workaround : Configure Live MoH. Even if you don’t have a livefeed source, the
script will failover to the MoH file in
Agreed voice mail ports will make the call and they also provide the external
member mask so configure that also
Sent from my iPad
On Aug 25, 2013, at 3:54 PM, Edgar Feliz ejzi...@gmail.com wrote:
I just put the VM ports in my HQ CSS and did not have to make any other
changes.
Edgar
Couldn't you just create a virtual desktop on your server, give. It a
connection on both networks and RDP into Tito get access to both environments?
Sent from my iPad
On Jun 17, 2014, at 9:54 AM, Amdo Ngawa datapack...@gmail.com wrote:
Hi Folks:
My Esxi server has two ethernet ports; one
I started with the by section approach. It became a problem when I got
several sections into and needed to create Partitions and Calling Search
Spaces.
Start with the Basics and work up:
Switch/vlan configuration
DHCP servers
CM basic partition and CSS
CM phone configuration
I had problems with the 6608 in the front left pod, and a friend had
problems with the 6608 in the front right pod.
My problem was...I issued a set port #/# disable followed by set port
#/# enable on one of the 6608 ports and the entire 6500 locked up, zero
response from the console. Only thing I
http://www.vmware.com/download/esxi/
Try no sccp, then sccp. I had this problem saturday.
This message was sent from my Nokia E61i
-Original Message-
From: Jonathan Charles
Sent: Mon 09/01/2008 12:33 PM
To: OSL CCIE Voice Lab Exam
Subject: [OSL | CCIE_Voice] HQ thru GK to CUE
Transcoder appears to be not
I think the questions asked in this group are a good indication of what is
tested. Many on this list have previous lab attempts.
*This message was sent wirelessly from my Nokia E61i *
-Original Message-
From: Robert Schuknecht
Sent: Fri 10/10/2008 7:07 AM
To: OSL CCIE Voice
failed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergio
Polizer
Sent: Monday, October 13, 2008 2:29 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] test
test
1.) Yes they are considered external.
2.) so from HQ you can call a PSTN number local to BR1 or BR2 using toll
bypass
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, October 16, 2008 3:10 PM
To: ccie_voice@onlinestudylist.com
Subject: [OSL |
You can create a translation pattern for the called number and select Block
this Pattern
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hardesty, Scott
Sent: Tuesday, November 18, 2008 11:32 AM
To: Michael Shavrov; ccie_voice@onlinestudylist.com
I'm not sure, but are you allowed to do this with the b-acd script? I
thought b-acd required a ephone-hunt pilot number so the call could be
controlled, redirected, etc.
What do you see in the debugs? I would enable debugs, then refresh both
scripts to ensure they refresh without any errors.
Quick answer: Under the properties of the select resource step, change the
connect parameter to the appropriate value. Then you can drop in the play
prompt step and a connect step.
From: ccie_voice-boun...@onlinestudylist.com
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