have you got it registers with the CCM before on any of your VM spanshots ?
i faced this issue before and it only fixed when i have reverted the
ccm snapshot to older one
Thanks
Ash
On Sun, Jul 17, 2011 at 11:35 AM, steven moran smoran...@gmail.com wrote:
No matter what I do I cannot get
remember doing this on VM.. but yes i think i have.
im sorry, i dint understand what you eant by the ccm snapshots
could you clarify..
On Sun, Jul 17, 2011 at 2:46 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
have you got it registers with the CCM before on any of your VM spanshots
?
i
By default the priority for each CCM will be server and you can prioritize
them using command Gw-priority .
Vik have explain/configure this in each GK config example he have in WB2
Thanks
Ash
On Sun, Jul 17, 2011 at 6:39 PM, Peter peter.imman...@rediffmail.comwrote:
Hi,
In the lab, suppose
if the Questions ask to prioritize YES, i believe the Q will be clear enough
if they ask for this requirement in specific ,
if i am in the exam now and this is not specified clearly i will ask the
proctor and he must give me more accurate info about this :)
but General speaking they must tell you
Hello All ,
can we all try to make useful info from this Question because we all
might face this at the lab and there is no Alias in there to save us
:)
- can you please confirm this domain name matter for all of us , i
need to know if you have configured wrong domain name or it just stuck
in
Hello ALL ,
Mobile voice access does not look at the calling party name configured
on the DN when placing the outbound call. Therefore you will not see
calling name being passed to H323 gateway and so the PSTN connection .
this is a working as design issue , and there is no workaround for it
at
Many congrate Bro , and thanks for the good info/news :)
Best regards
Ash
On Tue, Jul 19, 2011 at 6:50 PM, Cristobal Priego
cristobalpri...@gmail.com wrote:
i forgot to mention
Tom the proctor in San Jose is retired, he took early retirement
so there are only 2 proctors in San Jose now, Tong
Congratulatuion Emin , all the best ,
Ash
On Wed, Jul 20, 2011 at 6:22 PM, Emin Guliyev eguli...@fidelus.com wrote:
Guys,
I passed the lab yesterday on second attempt. I appreciate
all the help I have gotten from OSL.
Regards,
Emin
Hello All ,
This error indicates that while loading the script, the VXML file
IVRMainpage.vxml failed to be downloaded from 10.10.210.10:8080 (HTTP
GET).
maybe a firewall in the network is blocking the request from your GW
interface to 10.10.210.10:8080 or the GET HTTP souce ip is the reason
Rashid ,
1- CCM 4.2 get out of Cisco TAC support :) plan your upgrade please in
cases you need TAC to fix any issue :)
2- if the phone is unknown = unregistered how can you make calls from
it !? what the user hear when the call got disconnected ?
if i were you i will troubleshoot this issue in
Hello Joe ,
i think the transcorder work as MTP when you allocate it and that's
why you get it working , (of course, no logs=assumptions)
but have you tried to invoke any supplementary service in the case
when you had XCODER allocated ? see if it will work
-can you try to enable MTP ( for the
if its site CME means H323 in between and so the + will not pass out ,
you have to add it again manual using TR
i believe the issue you referring to is related to the FW version in
use and the type of the phone is use ,
Ash
On Wed, Jul 27, 2011 at 8:20 PM, CCIE for Me cciefo...@hotmail.com
Hello William , Joe
the default on the is SS not FS and if you enable FS for the outgoing
calls on the CCM you will always need MTP otherwise the option will be
grad out ,
MTP will fix this issue , can you please try to not enable FS and
allocate MTP and test and then enable FS on both the CCM
correction :
the default on the GW is FS , and the CCM can fall back to SS automatically :)
sorry guys, thanks for refresh my memory
Ash
On Wed, Jul 27, 2011 at 9:12 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hello William , Joe
the default on the is SS not FS and if you enable FS
the person managing the list at
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. Re: calling through GK and CUBE - calling phone keeps ringing
(Ashraf Ayyash)
2
.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashraf Ayyash
Sent: Thursday, July 28, 2011 9:37 AM
To: Kshitij Singhi
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Urgent: CUE
hello Joe ,
in the CCM service parameters search forSIP Station KeepAlive
Interval , its by default 120 sec , the minimum is 60 change it
reset your phone while they are registered to the CCM and then create
the fall back scenario and check the behavior .
Best regards
Ash
On Sat, Jul 30,
. (it works with
boothelper)
Best Regards,
Moustafa Medhat | CCNP-V,RS,CCS.
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Moustafa Medhat
Sent: Friday, July 29, 2011 2:49 PM
To: Ashraf Ayyash
Cc: ccie_voice
hello ,
those are the debugs i always use for those kind of problems and they
are always suffice :
show ccn status ccm-manager
show ccn trigger jtapi
show ccn subsystem jtapi
show ccn status ccm-manager
trace webInterface initwizard all
trace ccn jtapiAutoUpdate all
trace ccn SubsystemJtapi
Hello ,
the default codec on the GW and the CCM is G729r8
Best regards
Ash
On Mon, Aug 1, 2011 at 9:53 AM, eng_firasoq...@yahoo.com
eng_firasoq...@yahoo.com wrote:
Hi,
My question is related to codec selection under transcoder configuration so
if we want g729 call between HQ BR1 G711
configuration
Regards
--- On *Mon, 8/1/11, Ashraf Ayyash ash.ayy...@gmail.com* wrote:
From: Ashraf Ayyash ash.ayy...@gmail.com
Subject: Re: [OSL | CCIE_Voice] G729 Codec
To: eng_firasoq...@yahoo.com eng_firasoq...@yahoo.com
Cc: ccie_voice@onlinestudylist.com
Date: Monday, August 1, 2011
participants are cisco phones, also across wan (g279), no problem.
if using dial-peers on iOS, they use default g729r8
here is the point where it could go wrong, i think.
i include the g729r8 in the dsp profile.
just my view on it.
regards Ron
Op 1 aug. 2011, om 20:06 heeft Ashraf Ayyash het
Hello Brain ,
can you check what DP assigned to your MTP and what codec is being
used in there ?
Thanks
Ash
On Sun, Aug 7, 2011 at 5:50 AM, Brian Rudy brianr...@gmail.com wrote:
Hello -
The call flow I am working on looks like this
hq-phone(sccp)-gk-cube-cme-phone(sccp). I am working on Vol
--
From: Ashraf Ayyash ash.ayy...@gmail.com
Sent: Sunday, August 07, 2011 3:01 AM
To: Brian Rudy brianr...@gmail.com
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Bandwidth Interzone w/ MTP Issue
Hello Brain ,
can you check
.
Thanks
Brian
On Sun, Aug 7, 2011 at 3:01 AM, Ashraf Ayyash ash.ayy...@gmail.com
wrote:
Hello Brain ,
can you check what DP assigned to your MTP and what codec is being
used in there ?
Thanks
Ash
On Sun, Aug 7, 2011 at 5:50 AM, Brian Rudy brianr...@gmail.com wrote:
Hello
one last thing ,
can you please remove all the Media assinged to the BR trunk and make
MRGL contain the IOS MTP only ?
thanks
Ash
On Sun, Aug 7, 2011 at 8:54 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hello Brian ,
Thank you for the detailed explanation , in fact like i said the CCM
MCUs in local zone USA : do not use proxy
Best regards
Ash
On Sun, Aug 7, 2011 at 10:23 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
one last thing ,
can you please remove all the Media assinged to the BR trunk and make
MRGL contain the IOS MTP only ?
thanks
Ash
On Sun, Aug 7, 2011 at 8:54
Hi ,
can you collect deb ccsip message and voip dialpeer from your CME ?
show run from CUE/CME
Thanks
Ash
On Wed, Aug 10, 2011 at 7:46 AM, mgscip gpsvoiceexpe...@yahoo.com wrote:
H All,
I'm testing with CUE with CME SRST Mode facing the issue in MWI.
configuration as below
hello Stuart ,
the RSVP assume the worst case of the sampling rate so it will go with
10 MS which will result the RSVP to need 40 Kbps for 1 G729 call and
96 Kpbs for G711 call ,
now the bandwidth you have assigned to the RSVP should give you 1 G729
call and this is also failed so i would
Wael ,
this option should do the trick for you , please provide the exact
step you done to change the selection order
and also you can always do deb ccsip message and check if the
diversion header order got changed after you change the selection
order but this is the only option to play in the
Hello ,
1- this is not related to the CCIE-V lab exam which is the concern of
this alias
2- here is the solution of the issue you are facing :
http://www.cisco.com/en/US/customer/products/sw/voicesw/ps556/products_tech_note09186a00801786cb.shtml#solution2
Problem-Error Reading File
Hello Amit , All,
Thank you for the good link , and i would like to add the rest of the
config as well clarify what is the main functionality for the TAPS
after you will complete the installation of the TAPS plugin , it will
automatically create application with trigger calling the TAPS script
Hello All ,
To answer Adil's Questions , personally i don't think this will be a
a question but its good to know it , TAPS have nothing to do with
custom scripting and as well it will take time for what ( for
registering phones and i dont think that there will be visible way to
know how it has
Hello Ray ,
in which phase exactly you got this error ? can you check if you have
TAPS file in the C: directory ?
Also try to remove it from Add / remove programs and then restart the
Engine and re-install it again .
let me know how it will go .
Ash
On Thu, Aug 18, 2011 at 8:10 PM, Ray
Hello Wael ,
from the RmCmResources page you can change the Automatic
Available from disable to enable and this should met your requirement
,
i recommend you to brows the CCX pages as well and try to find what
options we have and of course the admin guide of the CCX is so handy
to understand
Hello Ray , All ,
also here is the troubleshooting DOC for the feature :
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a008084321e.shtml
Ash
On Thu, Aug 18, 2011 at 10:08 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hello Ray ,
in which phase exactly you got
/7_0_1/t15taps.html
Ash
On Sun, Aug 21, 2011 at 12:14 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hello Ray , All ,
also here is the troubleshooting DOC for the feature :
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_tech_note09186a008084321e.shtml
Ash
On Thu, Aug 18
-
Oct 16 2010 07:53:33 AM SEP081FF362981B 1000 2589 2589 PASSED
Oct 16 2010 08:03:58 AM SEP081FF362981B 1000 2589 2589 PASSED
Watch out !
Ash
On Sun, Aug 21, 2011 at 3:20 AM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
Hello All,
i
Hello jorge ,
the only difference between these 2 ways in that when you used the
feature block , you will see the CONF softkey but when its grad out
and when you click on it , you will see the key is not active here ,
however using the soft key in the ephone template will not show you
the CONF
Hello Dew ,
the CUBE is IP-IP protocols level and the DSP is in lower layer in the IOS ,
the CUBE will not allocate any DSP for ANY protocol translations ,
the CUBE will allocate DSP if it will find DTMF/CODEC mismatch on the
calls which is terminated by him and if so , the CUBE will start
Hello DeShon ,
this is correct but the point i was explaining is the fact that the
IP-IP functionality will not need any media device to be allocated ,
you guys are talking about the media part of call which will be after
the signaling negotiation part and the protocol translation ,
the CUBE
than Re: Contents of CCIE_Voice digest...
Today's Topics:
1. When DSP is used in CUBE ? (Dew Swen)
2. Bootcamp Seat for SALE (Duncan Hamilton-Walker)
3. Re: When DSP is used in CUBE ? (Ashraf Ayyash)
4. Bootcamp Seat for SALE (Duncan Hamilton-Walker)
5. Bootcamp Seat for SALE
This mean you didn't enabled the MVA from the Service parameters of
the CCM , or your phone is not MVA enabled , please check the config
on the CCM side as this is a config error in the CCM not on the GW
Ash
On Fri, Aug 26, 2011 at 6:33 PM, Ray jonha...@yahoo.com wrote:
- Forwarded Message
So i hope i have replied your proper question answer, yes i have given my
personal address if anyone want to contact ask anything they can ask it,
Because i dont hide any knowledge which you might be hehehehe
Regards
From: Ashraf Ayyash ash.ayy...@gmail.com
To: Labdone Labdone labd
the details hehehehehehe
Now, Request and then i will think and reply to u
I shared my happyiness whether you want to take it and not it is your choice
dont give your bull shit on open forum otherwise people know how to reply it
:)
Regards
From: Ashraf Ayyash ash.ayy...@gmail.com
To: Labdone
Hello Erwan ,
Paging Feature will be available on the latest 8.6 CCM Version ,
Ash
On Mon, Aug 29, 2011 at 8:38 AM, Erwan Erwan e_er...@yahoo.com wrote:
hi all,
does call manager 7 have paging feature we can use ? for basic paging
tks
___
, 2011 at 3:20 AM, Ashraf Ayyash ash.ayy...@gmail.comwrote:
Hello Erwan ,
Paging Feature will be available on the latest 8.6 CCM Version ,
Ash
On Mon, Aug 29, 2011 at 8:38 AM, Erwan Erwan e_er...@yahoo.com wrote:
hi all,
does call manager 7 have paging feature we can use ? for basic
I see it
*From:* Ashraf Ayyash ash.ayy...@gmail.com
*Sent:* Monday, August 29, 2011 1:56 PM
*To:* Mark Reed marklr...@gmail.com
*Cc:* ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] paging with CallManager 7
the release still coming over in the few next weeks,the release
features , one worth to mention is
Redirecting Number Xformation :)
Best regards
Ash
On Tue, Aug 30, 2011 at 4:39 AM, Roger Carpio roger.car...@gmail.comwrote:
Don't you worry boys... I know if we behave Santa Claus will program it for
us... LOL
On Mon, Aug 29, 2011 at 12:56 PM, Ashraf Ayyash
this error mean that you have install/run CUPC version not compatible
with your CUPS Version ,
CUPC 7.1 is only compatible with CUPS version 7.0.5
http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.html#wp40258
downgrade your CUPC or upgrade your CUPS
Ash
On
very logic , you been asking for non-existing B-channel and so your
PSTN router decline the setup with disconnect cause make sense
you can also fix this issue by revers the bchannel order from the CCM
gateway config page so your ccm wil start asking for channel 1 for the
first call and then go up
have you configured the QOS section in that lab ?
On Sun, Sep 4, 2011 at 1:46 AM, John McGaughey (jomcgaug)
jomcg...@cisco.com wrote:
Hello,
I’m in workbook 1, lab 9A. I’m testing the Intercom feature between the
assistant and manager. It doesn’t matter who initiates the intercom, I get
? take it off from the routers ?
Ash
On Sun, Sep 4, 2011 at 8:40 PM, John McGaughey (jomcgaug)
jomcg...@cisco.com wrote:
Yes I have.
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Sunday, September 04, 2011 12:06 PM
To: John McGaughey (jomcgaug)
Cc
-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Sunday, September 04, 2011 12:55 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
okay , with RSVP the RTP traffic will be terminated by the MTP's so
bind the sip traffic to the correct ip , the voice vlan
voice service voip
sip
bind all sou
Ash
On Mon, Sep 5, 2011 at 1:45 PM, Rynard Coetzee
rynard.coet...@bytes.co.za wrote:
Hi All
I have a problem with my Unity Connection and Call Manager Express
integration ,when I dial the voicemail
sip
bind control source-interface Vlan400
bind media source-interface Vlan400
registrar server
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: 05 September 2011 01:41 PM
To: Rynard Coetzee
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL
Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: 05 September 2011 02:22 PM
To: Rynard Coetzee
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration
please get full ccsip mess for one of the calls please
On Mon, Sep 5
was bound to previously ,I had changed the binding to the voice vlan but
forgot to change it on UC. Working 100% now.
Thanks for the help.
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: 05 September 2011 03:16 PM
To: Rynard Coetzee
Cc: ccie_voice
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: Monday, September 05, 2011 2:11 PM
To: John McGaughey (jomcgaug)
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Strange Intercom/RSVP issue
great so now we isolte the problem to be in specific from
and bind the sccp in the HQ router under the sccp ccm group 1
Ash
On Mon, Sep 5, 2011 at 10:18 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
thats looks good , lets see if the ios upgrade can give any new results
On Mon, Sep 5, 2011 at 10:15 PM, John McGaughey (jomcgaug)
jomcg...@cisco.com
to be intercom between 2 phones in
the same site ( i hope Vik can check this behavior as well )
2- as you know how to configure the intercom and now we know why its
not working we can just ignore this Question ,
Ash
On Mon, Sep 5, 2011 at 9:19 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
and bind
Dubai lab is very slow (not always , its up to your chance ) and very busy
, they got 1 seat to serve the middle east so if you want to get seat now ,
you will need to wait for about 2-3 months in advanced .
Ash
On Mon, Sep 5, 2011 at 9:26 PM, eng_firasoq...@yahoo.com
eng_firasoq...@yahoo.com
Hello John , All ,
i have followed up with the development team about this issue and they
confirmed that the defect note about the RSVP support with intercom is
not accurate, the RSVP with codec pass through should work with
intercom call and this might be new defect or something but i have
Many many congrats Bro , celebrate the Victory to the MAX
Ash
On Fri, Sep 9, 2011 at 6:57 PM, Ahmed Ellboudy
ahmed_ellbo...@rayacorp.comwrote:
HI ALL ,
I Pass MY exam after 2nd attempt , finaly I did it my CCIE #30079 ,Thanks
for IPX document it was a good documents for your preprations,
check the following :
1- when you dial the trigger have you seen the called number on the
phone screen changed to the CTI port number instead of the CTI route
point number you have already dialed ? if so this mean the call have
been accepted by the CCX engin but something bad happened after that
you will need to check what ANI/DNIS/RDINS.etc.. in this calls , from
ther RTMT , there is something called port monitor , which will give
you real time info about the call and what exactly happened to it and
what rule/call handler it reached , and then you can proceed further
based on what you
as unsolicited but it doesnt work.
Thanks team
--- On *Tue, 6/9/11, Ashraf Ayyash ash.ayy...@gmail.com* wrote:
From: Ashraf Ayyash ash.ayy...@gmail.com
Subject: Re: [OSL | CCIE_Voice] Problem with UC and CUCME integration
To: Rynard Coetzee rynard.coet...@bytes.co.za
Cc: ccie_voice
All ,
the get reporting static will give get value for you and that's all ,
and in the scenario you are talking about you need to make the CCX
script say this number so you will need to have generate prompt step
to read the value which you got from the get reporting statics .
so the decrement
of your fellow list members. Not everyone has a clue all
the time, about everything, so be gentle in your communications.
On Sat, Sep 10, 2011 at 9:17 PM, Ashraf Ayyash ash.ayy...@gmail.comwrote:
Hello Ken ,
your search and explanation is pretty good and made sense and i think
Hi all ,
if we took off the master command we will not be able to sync our internal
network entities to the HQ router , please feel free to correct me if i am
wrong ,
the output you gave Ray for the first email and the config say that you have
added the master command before you synced with the
are you running 7.0.2 ccm ? CSCsy60115
Ash
On Sun, Sep 18, 2011 at 12:48 AM, Sanoj Thomas san...@yahoo.com wrote:
Hi All,
CUCM does not recognize the Remote destination based on the caller ID and
prompt for a pin number, but instead the system prompts to enter the remote
destination
. Yes i am running CUCM 7.0.2.1-18.
Cheers
Sanoj
From: Ashraf Ayyash ash.ayy...@gmail.com
To: Sanoj Thomas san...@yahoo.com
Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com
Sent: Sunday, September 18, 2011 3:42 PM
Subject: Re: [OSL
you are able to sync with out the master command to the HQ router ?
can you get show ntp status assoc please ?
On Sun, Sep 18, 2011 at 6:21 AM, Mini Me cciev.min...@gmail.com wrote:
Hi Ashraf,
You will still be able to sync time without the master command.
HTH
From: Ashraf Ayyash ash.ayy
was studying and it works.Make sure the HQ router syncs
with external NTP . I passed and no longer have a lab to send you output.
Sorry about that.
HTH
From: Ashraf Ayyash ash.ayy...@gmail.com
Date: Sun, 18 Sep 2011 06:24:06 -0700
To: Prashant Patel cciev.min...@gmail.com
Cc: Ray jonha
of the ntp source.
HTH
From: Ashraf Ayyash ash.ayy...@gmail.com
Date: Sun, 18 Sep 2011 06:31:51 -0700
To: Prashant Patel cciev.min...@gmail.com
Cc: Ray jonha...@yahoo.com, ccie voice ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] ntp
okay thanks for the clarification but can
is 0.00 msec
loopfilter state is 'CTRL' (Normal Controlled Loop), drift is -0.01989
s/s
system poll interval is 64, last update was 1 sec ago.
hq#
--
*From:* Mini Me cciev.min...@gmail.com
*To:* Ashraf Ayyash ash.ayy...@gmail.com
*Cc:* Ray jonha...@yahoo.com
instability in timekeeping if the machines do
not agree on the time.
===
Michael
On Sun, Sep 18, 2011 at 4:28 AM, Ashraf Ayyash ash.ayy...@gmail.comwrote:
Hi all ,
if we took off the master command we will not be able to sync our
internal network entities to the HQ router , please
when you'll need ntp master.
Hope this helps!
Michael
On Sun, Sep 18, 2011 at 10:14 AM, Ashraf Ayyash ash.ayy...@gmail.comwrote:
i have read this note but it didn't stated that the router can provide
clock without master command , in here we have one ntp source in our network
which is the HQ
can you get what cause the call got when it disconnected ? debug ccapi
inout at the GK router , also for such issue CCM SDI/SDL traces is
needed
can you get them ?
that scenario worked for me before , i dont have lab to test it now
but i would give it try once again and try to produce the issue
fixed , check this as well
Ash
On Sun, Sep 18, 2011 at 4:12 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
can you get what cause the call got when it disconnected ? debug ccapi
inout at the GK router , also for such issue CCM SDI/SDL traces is
needed
can you get them ?
that scenario worked
Hey Rynard ,
as you are using Cbarge , there is no need for Built in Bridge , you
will need have the privacy off and the barge method as Cbarge from the
phones pages , reset the phone and check the behavior ( make sure that
you will have conf brigde available for the phones inside your MRGL
Ash
is registered to UCM
Thanks
-Original Message-
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of DeShon Crayton
Sent: Sunday, September 18, 2011 8:11 PM
To: 'Ashraf Ayyash'
Cc: 'OSL Voice'
Subject: Re: [OSL | CCIE_Voice] GK Calls
off the built-in bridge but was still getting
the error ,I then realised I had Locations CAC set from previous part of the
lab and that this was causing the cbarge to fail. Working now.
Thanks
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: 19 September
Hello All ,
i have review the NTP principle again with my Colleagues and with Amit
who did lab for this ( Thanks ALOT) and we have done some show
commands on the CCM and the IOS in regard of this issue
what we found is that the the lowest stratum is the Best and stratum
-0 is the lowest value
that might be a call routing matte , maybe the call from that phones
didnt reached the VM pilot at all (css , pt , Tanslation etc,,)
ofcourse we cannot be accurate in the answer because its your
dial-plan but you can do digit analyizer for that call and see what
you will get ,
if all is fine ,
them .
-Original Message-
From: Ashraf Ayyash [mailto:ash.ayy...@gmail.com]
Sent: 20 September 2011 01:24 PM
To: Rynard Coetzee
Cc: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] Problem with CUCM and CUC SIP integration
that might be a call routing matte , maybe
remove the / from here
your command tftp-server flash:/Desktops/320x212x12/List.xml
the correct command :
tftp-server flash:Desktops/320x212x12/List.xml
Ash
On Thu, Sep 22, 2011 at 8:40 AM, Ray jonha...@yahoo.com wrote:
has anyone done background image and it works.. when i go to user
set the mask on the cucm side , and it will take effect ,
Ash
On Mon, Sep 26, 2011 at 9:18 AM, zamuel del Toro sdelto...@hotmail.com wrote:
the uccx port network mask can't be assigned using + sign (ej.
+1212394) from uccx admin page, should be assigned manually from cucm
admin page
the first option and you can vary it by also adding : 9.[1-9]..
Ash
On Mon, Sep 26, 2011 at 9:22 AM, zamuel del Toro sdelto...@hotmail.com wrote:
if the requirement is for national 9 plus 8 any digits, it overlap for
900T and can't make international calls
is correct using
when you but the caller on hold from the phone , that will use what is
in your phone confg page of mrgl and moh user/network resources but
when you did the same from the ARC now you are using the setting on
the CTI port of the ARC , can you please go ahead and stream UNICAST
MOH through the CTI
the mtp will have by default G711u once you add it and as the mtp
accept on codec it was complaining , you need to do no codec and then
add whatever codec you want , and Yes it will accept g711a
Thanks
Ash
On Wed, Sep 28, 2011 at 8:10 AM, Gerence Guan cisco.g...@gmail.com wrote:
Hi Everyone,
Hey All ,
the TCP (by default) will happen after the connect and it will be
using the H245 portion of the H323 family so the command you guys
referring to will not show you what is going on and even h245 asn1
will also will not show you anything because what happened is that the
CCM is waiting
Hello All ,
in the explination you gave about the problem you must know what type of moh
you got on the HQ and you can confirm that by hear the moh and you will know
if its MMOH or unicast and this is very important to figure out because this
will turn the troubleshooting to diff direction ,
do you have the check box of reconnect to the higher ccm checked in the
server config page on unity ?
can you make sure that you have the port registered with the sub when it
will back ?
Ash
On Mon, Sep 26, 2011 at 11:39 PM, Ken Wyan kew...@gmail.com wrote:
Hi,
I have a strange issue of
81 mean unallocated unassigned number , you dial the wrong number
check this DOC :
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Thanks
Ash
On Thu, Sep 29, 2011 at 8:37 AM, Ray jonha...@yahoo.com wrote:
what option or command do i have to take out
i dont fully understand what you are looking for ,
on the ccm traces you will find Cause = the disconnect cause and
its Q931 in there few line above the release complete message ,
and to know what is the cause you need to use the DOC i have sent to
you , which will explain and show you what does
TransitionMASK::0040
Q931CauseIe IEData= 08 02 80 AF check the doc i have pointed and you
will find that AF is resources unavailable ...
i hope this is clear
Ash
On Thu, Sep 29, 2011 at 2:58 PM, Ashraf Ayyash ash.ayy...@gmail.com wrote:
i dont fully understand what you are looking for ,
on the ccm
because you added the command mls qos trust device cisco-phone the
switch will know via CDP that you are connecting Cisco Phone other
wise the default behavior (*after enable the mls qos on the switch)
will be applied of not trust ,
i would say that 99% you will be trusting the COS on the switch
the correct answer for your Question is to busy out the non-defind B
channels from the Service parameters ,
Note that CCM doesn't support Fractional T1/E1 over mgcp , this
service parameter is kind of workaround and i will use it myself if
will setup Fractional T1/E1 over mgcp ,
Ash
In IPX
Guys ,
the h323 bind command is not a GK command , its for the H323 GW so you
can have GK on interface and have the normal config of the GK
and the H323 bind under the interface which will work as H323 GW with the ccm
H323 bind can be usefull if you have Q said bind all media and
signalling to
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