Hi,
Apart from checking the network infrastructure configuration. what other
troubleshooting has been done?
Have you verified the phone actually gets correct voice vlan? does it
receive IP Address in the correct pool? does it get the TFTP server?
Check the settings on the phone.
If the phones
Hi,
Check if your application user that you are using for AXL has indeed
correct access configured.
And that the username/password matches.
Regards,
On Mon, Apr 2, 2012 at 2:46 AM, Baktha Muralidharan muralic...@gmail.comwrote:
Hi folks
I am having trouble integrating UCCX with call
configured with 'set dscp default' and the router's auto qos creates a
seperate class 'remark' just to do this?
Someone?
As usual, thx for the feedback!
Op 31 maart 2012 22:16 schreef George Goglidze gogli...@gmail.com het
volgende:
Hi Juan,
This is a good question, and the answer lies
You have to trust DSCP on interface connected to the router. Routers do not set
cos bits in dot1q header!!!
Same goes for interface connected to CUCM.
Sent from my iPhone
On 29 Mar 2012, at 01:14, Chris devsin2...@gmail.com wrote:
Hi Steve,
My approach would be same as yours. However few
Hi,
I'll just write regarding pass-through codec on MTP.
You must have pass-through codec when the voice payload is not supported.
For example, in case of SRTP traffic, the MTP resource without pass-through
codec would not support encryption, but if you use pass-through codec it
does not care if
well, it all depends where exactly in the network you got that wireshark
traces :-
switches by default will default all the ToS bits, unless configured
otherwise.
Regards,
On Tue, Mar 27, 2012 at 8:54 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote:
** **
Hello Guys,
** **
**
** **
** **
** **
** **
** **
*De:* George Goglidze [mailto:gogli...@gmail.com]
*Enviado el:* martes, 27 de marzo de 2012 01:14 p.m.
*Para:* Santiago Figueroa
*CC:* ccie_voice-boun...@onlinestudylist.com;
ccie_voice@onlinestudylist.com
*Asunto:* Re: [OSL | CCIE_Voice] CIPC QOS
** **
well, it all depends where exactly
Marco, I reckon you could do voice in less than 2 months!
Btw, this has nothing to do with voice, but with flash filesystem access on IOS
in general...
Ken, I've never seen need in #3.
And #1 sometimes does not work, not sure on which platforms...
#2 has always worked for me, regardless of
If you have noticed on CUPS Administration page, you have to configure two
things for voicemail.
1) Voicemail server--This uses Unity Connections Visual Voicemail
Web Service
2) Mailstore --- This uses IMAP to connect to Unity Connection.
Regards,
On Thu, Mar 22, 2012 at 8:49 AM, Juan
IOS is not using the longest match mechanism like CUCM.
It uses first match mechanism to route calls.
So as soon as it has matched any dial-peer it will send the call.
At least that is true with KPLM and SCCP digit by digit collection.
if en-bloq is used then it would correctly match the
on the RP on the CUCM side
*From:* George Goglidze [mailto:gogli...@gmail.com]
*Sent:* 16 March 2012 11:10 AM
*To:* Rynard Coetzee
*Cc:* vikas wankhede; ccie_voice@onlinestudylist.com
*Subject:* Re: [OSL | CCIE_Voice] Problem with dial-peers in SRST
IOS is not using
You have configured port serial 0/1/0 on the IOS router, but on CUCM you have
SU0 which I believe would be 0/0/0
Cheers,
Sent from my iPad
On 19 Feb 2012, at 17:05, Emanuel Damasceno aedamasc...@gmail.com wrote:
Hello Experts,
Here I come with another problem. Why is it showing as
I don't know where you got the table that you are referencing below... But you
can never set threshold 3, it's 100 percent always:
mls qos queue-set output X threshold X T1 T2 R M
T1 and T2 are self explanatory.
Then it's R for Reserved and M for Maximum.
So to answer your question, yes you
When agents first log in they will be not ready.
Automatic available - enable - refers to after call treatment, after wrap-up
time automatic availability.
And for your final problem check CSS of the CTI Ports, make sure they have
access to agents DN's partition.
Basically after an agent
have you tried google translate?
just kiddin... :-)
On Fri, Feb 17, 2012 at 11:59 AM, Ken Wyan kew...@gmail.com wrote:
Hi,
Using Q.931 Translator , we could translate CUCM trace files to equivalent
IOS ISDN messages.
I couldn't find it in CUCM ver 7. Is it available with CUCM 7 or
Hi Boris,
You are wrong, the TTL on multicast music on hold is NOT 1...
and depending on what you need, you can have pim enabled and deliver the
mmoh from flash to other remote networks, and by remote I mean across any
L3 device.
As well, you do not have a way to specify the TTL on the mmoh from
or SRST even) which are on
subnet separated by L3 from the CUCME Routers interface.
On Wed, Feb 15, 2012 at 11:56 AM, George Goglidze gogli...@gmail.comwrote:
Hi Boris,
You are wrong, the TTL on multicast music on hold is NOT 1...
and depending on what you need, you can have pim enabled
and show ip igmp groups, but the actual stream of Music
was not heard, based on CUCME Admin guide.
On Wed, Feb 15, 2012 at 3:08 PM, George Goglidze gogli...@gmail.comwrote:
and voila, here's the proof:
sorry but had to remove IP addresses for security reasons.
I'm using pim sparse-dense mode
There is a step to generate a number... M not in front of UCCX right now so
can't tell you exact name of the step, but it is very easy to find it.
In this step you tell it what you want it to gemerate, in your case number, and
then provide it with the value, and then output prompt.
Sent from
Hi Bruno,
The command paramspace callsetup after-hours-exempt true is only when
there is a call with that dial-peer as incoming leg, and then going out of
another dial-peer.
basically when you have call coming from somewhere else like CUCM, then
choosing as incoming leg dialpeer one with that
Datucha,
The whole point of Multicast is that it does not matter how many holds
there are in the site, there is still only one stream per codec per moh wav
file.
so if you have two moh music files, and g729 and g711a and g711ulaw then
you will have 3 mmoh streams!
and you can have as many
As my friend Kevin Spicer nicely corrected me, 2 files per 3 codecs are 6
streams!!!
Was thinking to multiply by 2, but then forgot to do it while writing the reply
:-
Sent from my iPad
On 11 Feb 2012, at 19:33, George Goglidze gogli...@gmail.com wrote:
Datucha,
The whole point
Frame-Relay is known for it's oversubscription capabilities...
So feel free to oversubscribe... It will be fine to configure one PVC 1536 and
another 128 or whatever is required...
Sent from my iPad
On 7 Feb 2012, at 19:32, datucha123 datucha123 datucha...@gmail.com wrote:
But that won't go
Hi,
Where are you calling from? From CME phones or PSTN? And is this real rack or
GNS3 simulation?
And attach the following debugs:
debug voip application
debug voice dialpeer inout
Sent from my iPad
On 5 Feb 2012, at 00:43, William Affeldt william.affe...@yahoo.com wrote:
I configured
Hi Anthony,
Case 1 if correct. That is the best way to configure it.
In case 2, you should not need to configure that forwarding rule at all.
As a matter of fact, I believe the call should not have RDNIS at all, it's
a direct call so should only have DNIS/ANI.
If I remember correctly at least.
Bad news.
I hope you have backups from Disaster Recovery System...
Otherwise I hope it's not production environment, just a lab, because
you'll have to rebuild.
Cheers,
2012/1/19 khaled Saholy khaled_sah...@hotmail.com
Hi,
What can we do in case of the Publisher failed to boot and the
Juan,
We are discussing AAR not fallback mode... that's completely different.
On Thu, Jan 19, 2012 at 9:07 AM, Juan Lopez
lopez.hernandez.j...@gmail.comwrote:
would 'call-forward busy 570.' work under fallback config?
This needs also that the 570X range is part of the HQ's DID range. If this
ISDN Se0/0/0:23 Q931: RX - RELEASE pd = 8 callref = 0x80F7
ISDN Se0/0/0:23 Q931: TX - RELEASE_COMP pd = 8 callref = 0x00F7
On Thu, Jan 19, 2012 at 4:26 PM, George Goglidze gogli...@gmail.comwrote:
Hi Anthony,
Case 1 if correct. That is the best way to configure it.
In case 2, you
yes it's possible, as long as it's correctly done:
http://www.cisco.com/en/US/products/ps6837/products_tech_note09186a00808a2b0d.shtml
Unable to View Directory Information Problem
With CUPC 8.x, you are unable to view directory information, you are unable
to see the status/Presence information
Hi,
The card is supported:
http://www.cisco.com/en/US/prod/collateral/routers/ps259/product_data_sheet0900aecd8057f2e0.pdf
Paste your voice gateway configuration, there must be a problem somewhere.
Cheers,
On Sun, Jan 15, 2012 at 8:45 PM, mercy forall mercy_for_...@hotmail.comwrote:
Hi,
I
Hi Mercy,
Did you configure this via tftp or manually?
I'm asking because you've got this: ccm-manager config server x.x.x.x
Can you do:
*debug ccm-manager config-download all
*then do *
no ccm-manager config
ccm-manager config*
And see if you are being able to download the config. paste the
correctisimo :-)
On Mon, Jan 16, 2012 at 3:25 PM, datucha123 datucha123 datucha...@gmail.com
wrote:
When configuring the simple BLF Speed Dial, we need to configure the
Subscribe CSS for watching Device. So that it could the the Watched DN.
But, the Owner User ID, Line Association with
Hi Datucha,
Actually for CUPS you need only line association. you do not need to
specify owner user id.
Misread the question a little bit initially.
the owner user id on the phone is for SNR.
Cheers,
On Mon, Jan 16, 2012 at 4:01 PM, George Goglidze gogli...@gmail.com wrote:
correctisimo
MVA always needs a voice gateway!!!
On CUCM 7.0 it supports h323 only
On CUCM 7.1.3 (not tested in lab) it supports h323 and SIP.
To answer initial question, no you do not need dial-peers to reach dialed
numbers through MVA. Only MVA dial-peer.
Cheers,
Sent from my iPad
On 7 Jan 2012, at
...
Also this kind of configuration does not support IVR prompts.
But still the MVA is supported without H323 Gateway at all.
On Sat, Jan 7, 2012 at 4:05 PM, George Goglidze gogli...@gmail.com wrote:
MVA always needs a voice gateway!!!
On CUCM 7.0 it supports h323 only
On CUCM 7.1.3
Datucha,
In the last case you described. If you have UC as your voicemail system,
and phones go to SRST mode. I would strongly recommend against using CUE as
backup because all the voice messages that the users would receive while in
SRST mode would never reach the user once they're back to CUCM.
Hi,
You have to create two different partitions with two different TP-s. There
is no other way around it.
PT_PSTN_sa
PT_PSTN_sb
I would suggest you never try to use same pattern, even RP for two
different sites in a lab, unless it's stricktly necessary by task
requirement.
Don't try to save
whether those digits were sent by SIP or SCCP (in case of CUCME as well),
or even H323, it will route them immediately and it does not have the
interdigit timeout for enblock digits.
Is it correct?
On Tue, Dec 27, 2011 at 4:05 AM, George Goglidze gogli...@gmail.comwrote:
Hi Datucha,
I'll answer
Hi Datucha,
I'll answer to the last part, about enbloc and why when re-dialed from SCCP
Phone there is no interdigit timeout, but there is when redialed from SIP
Phone.
SCCP has defined message type StationEnblocCall, and when the call is
redialed it sends it with this parameter.
CUCM recognizes
if you only manipulate DNIS with Called Party Transformation Pattern, ANI
is NOT affected - therefore any prior transformations done by RL/GW or RP
will be intact.
Same goes other way around. basically ANI and DNIS are treated as two
seperate entities.
Cheers,
On Fri, Dec 23, 2011 at 4:11 PM,
Hi Datucha,
You already know all this, but let me describe it all agian anyway:
1) ICT (Non-gatekeeper controlled) - is for communication with another CM.
Not the voice gateway - regardless it's CME or not.
2) ICT (Gatekeeper controlled) - for compatibility with CM prior to version
3.2
3) h225
is
also remarked/ marked based on the outoing dial-peer settings.
As for inbound direction, the dial-peer QoS settings does not take effect.
On Fri, Dec 16, 2011 at 10:07 PM, George Goglidze gogli...@gmail.comwrote:
wrong, the media stream might not even hit the router at all! :-)
it would only
Hi Datucha,
when you have ip qos dscp under dial-peer configuration, that will only
change the DSCP marking for RTP stream originated from the router.
For example, incoming PSTN call - PSTN --- GW dial-peer 10 dscp cs1 -
CUCM (IP Phone)
In this case the stream originated from the voice gateway
MGCP gateways do not have preferred codecs, they are dummy endpoints, they
only announce their capabilities to CUCM and then it's up to CUCM's
preferences and region configurations to choose which codec will be used.
And show mgcp will show you what capabilities the gateway has(including
codecs).
Hi Alex,
From the output you are providing, there is not realy much to go on.
The first thought I had was that the counters on HQ router were not on 0
when you started the call, or there were more than one calls active at the
same time and not to the same site.
The only things I can deduce for
Hi Steven,
As you're saying yourself, you have configured port 0/0 on a CUCM gateway
configuration: vwic-2mft-E1 0/0
and on the real router, you have port 0/1, hence it can't register.
Correct this, and you should be fine.
Cheers,
On Sun, Jul 17, 2011 at 9:35 AM, steven moran
Hi Adil,
Without debug h245 asn1 it's difficult to be sure, but I bet on this other rack
rental they do support g729 codec on incoming dial-peer on PSTN-WAN router and
thats why the call succeeds.
Sent from my iPad
On 7 Jul 2011, at 02:08, Adil Shaikh adil.sha...@gmail.com wrote:
hi
Hi all,
As the subject suggests, it's official, I'm dual CCIE #19926 RS and Voice
starting yesterday.
Finally, it's a relief, no more studying late, spending weekends on a
computer, making calls, my neighbours think I escaped a psychiatric, after
hearing voices at night TEST VOICEMAIL FOR HQ
this helps someone,
Regards,
On Wed, Jun 29, 2011 at 9:29 PM, Jeff Garvas j...@cia.net wrote:
On Wed, Jun 29, 2011 at 9:44 AM, George Goglidze gogli...@gmail.comwrote:
By the way, I made it technically on my first attempt. Well, I payed once
only, although I went to exam 3 times.
1st time
who's responsible for that anomally? :
I guess CUCM Administrator :-) you...
On Thu, Jun 23, 2011 at 2:05 PM, Pithog Oil pithog...@yahoo.com wrote:
Hi Bill
Thanks to all those who have answered my first questions, what do i do so
as to change a SIP phone to SCCP. Also in a situation
would I be breaking NDA if I said they had horrible coffee in the lab? :-)
On Thu, Jun 23, 2011 at 7:32 PM, ccieid1ot ccieid...@gmail.com wrote:
I can't even spell NDA without shriveling.
On Thu, Jun 23, 2011 at 1:28 PM, Bill Lake whl...@gmail.com wrote:
Yes it is but I am not breaking NDA
WTF man, I can't believe this is happening to you.
I wouldn't be able to sleep until I got the results.
On Wed, Jun 22, 2011 at 10:46 PM, Pablo Meneses pmenese...@gmail.comwrote:
Can you believe that I haven't received the answer yet!
This has been the longest wait ever.
I even opened a
I personally think you've made a right choice with IPExpert.
And if you want a graded lab, you can always do one graded lab with Assessor
labs.
I personally did one, and to be honest it wasn't the best. The lab was
easier than IPExpert's, and easier than real lab too, so I feel like I spent
db replication problems?
On Sun, Jun 19, 2011 at 4:49 PM, Santiago Figueroa sfigue...@mnet.com.mxwrote:
Hello I have a doubt what is reason the ip phone 7941 do not registered
to cucm the status reject, I am sure the MAC and phone type in the cucm is
add correctly.
** **
Thanks.
ohhh, don't get me started on this one mate... I could say much... too
much.
On Sun, Jun 19, 2011 at 7:22 PM, Cristobal Priego cristobalpri...@gmail.com
wrote:
https://supportforums.cisco.com/message/3380695#3380695
what do you think ?
___
For
Hi Alex,
This has been discussed in numerous ocasions, here's one link to archives:
http://onlinestudylist.com/archives/ccie_voice/2010-April/015608.html
Regards,
On Fri, Jun 17, 2011 at 10:07 AM, Alex Goh ncsalex@gmail.com wrote:
Hi Guys,
Anyone encounter this issue before, after
Show us the CUE module sip subsystem configuration...
Sent from my iPad
On 15 Jun 2011, at 06:30, shelesh shel...@hotmail.com wrote:
Hi,
I have CME 7 and this gateway work as H323, and I have configure CUE, CUE is
working fine, but I am not getting MWI on phone., please find my
Hi Shelash,
The easiest for you would be to prefix 7 to the called number.
On Call Manager, go to MGCP Gateway configuration, then on Endpoint
configuration page, in section: Call Routing Information - Inbound Calls**
There's an option to Prefix DN: and put 7 there.
You will have to reset the
LOL
Fine, then I guess your router is SRST as well?
You can try this:
voice translation-rule 1
rule 1 /^...$/ /7/
voice translation-profile PSTN-IN
translate called 1
voice-port 0/0/0:23!this has to be your D channel, if you're in
europe it will be 15.
translation-profile incoming
Hi Sam,
The known good config is what Cisco suggests in their SRND, and they
suggest to have two different numbers:
Here's a snippet from SRND:
*Note *When deploying Mobile Voice Access in hairpinning mode, Cisco
recommends configuring the Mobile Voice Access DID at the PSTN gateway and
the
Hi Adil,
Do the following:
On CUCM, on RP 1999 - translate to 1998, so that when h323 setup is sent,
it's already 1998.
Also make sure the call between MGCP GW Region and h323 GW Region supports
g711.
And on h323 gateway you can have just one dial-peer:
!
dial-peer voice 1998 voip
not best practice but if someone is
familiar with IPX material, it is a known good. and anyway, I recommended
that for his testing - not final config.
Cheers
Sam.
On Wed, Jun 15, 2011 at 10:41 AM, George Goglidze gogli...@gmail.comwrote:
Hi Sam,
The known good config is what Cisco suggests
LOL, I didn't think it was that complicated that it needed 4-5 years to
implement.
Maybe with the next blueprint :)
On Sun, Jun 12, 2011 at 3:58 AM, Michael Luo hout...@gmail.com wrote:
Just ran across this post by accident:
https://cisco-support.hosted.jivesoftware.com/message/477914#477914
Hi Donny,
I really don't think they will be asking to limit the port usage for
Voicemail on CUE.
It's license limited anyway, but if they do, just do max session 5.
Regards,
On Sun, Jun 12, 2011 at 3:09 AM, donny f f.faraday...@gmail.com wrote:
hi all,
wondering , where to configure if
I'll reply under your choices...
Choice 1- need to use only 4 digit ANI on CUCM route pattern to match with
translation-rule 10 on GW but you won't get mark because you did not
configure 10 digit ANI on route pattern !
Why do you think you will loose points? As long as provider sees 10digit ANI
presence will not work properly if you don't modify default SIP Trunk
Security Profile
You must have enabled Accept Presence Subscription, otherwise Presence
will never display On Hook/Off Hook status.
It will display only available as long as you're logged in.
Regards,
On Fri, Jun 10, 2011
if switchport voice vlan is used. make sure you have cdpv2 advertisements
enabled.
On Thu, Jun 9, 2011 at 3:52 PM, Sam Park upperlevelpark...@gmail.comwrote:
Couple things here
1. Don't use DHCP, hard code the phone for something like 10.10.100.25.
Hard code the TFTP server address too.
2.
MPX needs to resolve it's own hostname to be able to function
correctly. I'm not entirely sure about this though.
Anyway, thanks all for reading this,
Regards,
George,
On Thu, Jun 9, 2011 at 5:39 PM, George Goglidze gogli...@gmail.com wrote:
Hi all,
Cisco has announced MeetingPlace Express
Congradulations Miron.
Sent from my iPad
On 8 Jun 2011, at 06:06, Miron Kobelski findko...@gmail.com wrote:
I received my number yesterday :)
This list has been very helpful during my preparations, so thank you all and
I wish you good luck with your exams!
best regards
kobel
Hi,
Authentication URL should not be CUE, it should be CUCME and the URL is
different:
url authentication http://cme-ip-address/CCMCIP/authenticate.asp cisco cisco
Regards,
On Tue, Jun 7, 2011 at 7:25 AM, Rahul Kapor rahul.kapo...@gmail.com wrote:
Hi all ,
Trying to configure
VoiceView
,
On Mon, Jun 6, 2011 at 2:34 PM, George Goglidze gogli...@gmail.comwrote:
No you can't because it's trying to use CTI porta to play the message, not
SIP INVITE as it would do if it was integrated with CUCME...
Regards,
Sent from my iPad
On 6 Jun 2011, at 20:18, Victor Malyuga victor_maly
the phoens
when service button is pressed , message no service is configured is
displayed on phone.
JTAPI interation btw CUCM and CME works fine. Phone is associated with
JTAPI user.
Thx,
Rahul
On Tue, Jun 7, 2011 at 1:59 PM, George Goglidze gogli...@gmail.comwrote:
Hi
I mean in call manager.
On Tue, Jun 7, 2011 at 10:00 AM, Rahul Kapor rahul.kapo...@gmail.comwrote:
which service ?
Voiceview enabled in cue.
On Tue, Jun 7, 2011 at 2:15 PM, George Goglidze gogli...@gmail.comwrote:
Is the service Enabled?
On Tue, Jun 7, 2011 at 9:37 AM, Rahul Kapor
its throwing error ...
Authentication error Report this error to your system administrator...
any further config ?
thx,
Rahul
On Tue, Jun 7, 2011 at 2:58 PM, Rahul Kapor rahul.kapo...@gmail.comwrote:
ll check and ll update u ...
On Tue, Jun 7, 2011 at 2:54 PM, George Goglidze gogli
Hi Adam,
No need to remember all ports by memory at all...
You have all tools available in IOS to lookup ports:
h323 signalling:
VOICEGW#show ip port-map h323
Default mapping: h323 tcp port 1720
system defined
h323 RAS:
VOICEGW#show ip port-map h225ras
Default mapping: h225ras
figure it out?
Cheers,
On Mon, Jun 6, 2011 at 3:11 AM, George Goglidze gogli...@gmail.comwrote:
Hi Adam,
No need to remember all ports by memory at all...
You have all tools available in IOS to lookup ports:
h323 signalling:
VOICEGW#show ip port-map h323
Default mapping: h323
theory here, but to me it seems like
Cisco doc is just wrong on this.
However, if Cisco doesn't know, then proctor shouldn't either, so the
11000-11999 I'm sure will be graded as good anyway :D
On Mon, Jun 6, 2011 at 8:50 AM, George Goglidze gogli...@gmail.comwrote:
:)
you got me
NO... Unfortunately.
Sent from my iPad
On 6 Jun 2011, at 18:22, Santiago Figueroa sfigue...@mnet.com.mx wrote:
Hello I need to know if the feature Meet-me can initiate for first call from
PSTN Gateway instead of cisco ip phone, when Gw have a DID to number meet-me.
I don't think you can do add by extension without having LDAP imtegration done.
Sent from my iPad
On 6 Jun 2011, at 19:01, Alex Goh ncsalex@gmail.com wrote:
Hi Guys,
Anyone encounter this issue before? when I try to adding contact in IPPM
using the AddByExt options,
and it says no
No you can't because it's trying to use CTI porta to play the message, not SIP
INVITE as it would do if it was integrated with CUCME...
Regards,
Sent from my iPad
On 6 Jun 2011, at 20:18, Victor Malyuga victor_maly...@yahoo.com wrote:
Is it possible to preserve VoiceView functionality in
Hi Michael,
Let me put how I understand it, if it's wrong I'd like someone to correct me
please.
Imagine whole buffer for egress queue of one interface, this whole buffer is
divided by four queues.
But maybe at some point some queues might need more buffers than others, so
they don't reserve
Hi,
I've had the same problem, and then I just bought some real phones... I
don't know if it was my PC, or Windows that I was running, or the actual
softphone does not have that functionality.
Regards,
On Thu, Jun 2, 2011 at 9:32 PM, Ravindra Lakpriya lakpr...@gmail.comwrote:
Guys ,
i'm
Hi,
It will always send RSIP... graceful and RSIP restart when it comes back.
Graceful to the old one, to indicate that it's graceful disconnect and that
he should not disconnect calls.
and when it registers with the primary gain, it sends RSIP restart to
indicate the reason why it is restarting
Hi Michael,
What is the gateway protocol? MGCP or h323?
Can you provide configs in both cases?
If it's h323 what do you have on outgoing dial-peer? hardcored g711? you can
try:
voice-class codec 1
codec preference 1 g711ulaw
codeb preference 2 g729r
Cheers,
On Fri, Jun 3, 2011 at 5:21 AM,
Debug MGCP packets is all you need... Great tool to see all messages between
call agent and the gw.
Sent from my iPad
On 1 Jun 2011, at 21:52, adam compton com...@gmail.com wrote:
What are the best MGCP debugs to run?
___
For more information
public website at www.ipexpert.com http://www.ipexpert.com/
From: Ki Wi kiwi.vo...@gmail.com
Date: Sun, 29 May 2011 21:38:31 +0800
To: George Goglidze gogli...@gmail.com
Cc: OSL Group ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CUC Phone View
I have associated every
HQ phone 1 must be associated to the application user that is configured for UC
phone view.
Check the others too,
Sent from my iPad
On 29 May 2011, at 11:34, Ki Wi kiwi.vo...@gmail.com wrote:
anyone tried it?
I followed the guide closely from here
What codec are you using with your ITSP provider? If it's g729 then in-band
dtmf is already lost even before it comes into your network.
Can you paste your providers invite message here? With SDP present? You can
find it in sdi traces,
Sent from my iPad
On 29 May 2011, at 12:43, Franjo
ser ser 0/0 ses clear
Sent from my iPad
On 29 May 2011, at 22:07, Randall Crumm rrcr...@yahoo.com wrote:
HI,
I'm having a weird issue. (see snipits below)
I configured mt CME for CUE(not telephony-service yet) and I cannot ser ser
0/0 ses to it
I can ping 10.10.202.2 from hq rtr
The
Did you actually try this and it didn't work?
ser ser 0/0 ses clear
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On 29 May 2011, at 23:50, Randall Crumm rrcr...@yahoo.com wrote:
It looks like CUE is on line 194, but I can't clear it.
BR2-RTR#ser ser 0/0 status
Service Module is Cisco Service-Engine0/0
Service
Check db replication on CUCM, or lob in to the subscriber directly on the web
interface and check it there.
As well check show ccm-manager host
And see who exactly it's registered to.
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On 28 May 2011, at 23:00, Cristobal Priego cristobalpri...@gmail.com wrote:
hello all
It's Ringlist.xml not RingList.xml
Regards,
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On 26 May 2011, at 15:33, Ki Wi kiwi.vo...@gmail.com wrote:
I uploaded some crap image file into the tftp and after rebooting the
service, i'm able to retrieve that file.
On Fri, May 27, 2011 at 6:29 AM, Ki Wi
Hi donny,
Do not specify the load file at all, it's completely unnecessary. If you did
change the software from SCCP to sip on ccm that's it just register it.
It's not strictly necessary to use authentication register if all the phones
are local to you on same vlan, but if they are not local
If you have to change it, my bet is they will provide the FTP server...
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On 24 May 2011, at 19:26, Cristobal Priego cristobalpri...@gmail.com wrote:
hello all
i was just curious if you need to change the cue license on the lab, if you
don't have an ftp server to do so
like a tanscoder is required if we want to do G.729 between MGCP
and H323 GWs.
MVA IVR only supports G.711.
Michael
On Fri, May 20, 2011 at 10:34 AM, George Goglidze gogli...@gmail.comwrote:
Hi Michael,
Have you tried to put g729 on the dial-peer?
Because if the h323 Gateway and MGCP
Hi Brian,
I'm with you on this one, I think you should consider worst case scenario,
and if you have TEHO calls there could potentially be 3 g729 calls... so 40
MIPS...
On Mon, May 23, 2011 at 2:08 PM, Brian Mulgrew btmulg...@gmail.com wrote:
Hi - Going though some of the DSGs when calculating
Hi Michael,
What is the destination phone? I guess it's CUCME? but is it SCCP Phone or
SIP?
You've showed the dial-peers on CUBE, but what about dialpeers on the
destination CUCME?
What dial-peer is it matching? and what codec does that dial-peer have?
Do the following on the CUCME:
debug
Hi Miron,
From what I understand in your case you only have one trunk, with CUCM Group
that had two call managers.
In this case the second call manager is only used in case if the first fails,
so it will not work as CUCME can't get to the Publisher.
If you want to cover the failover in the
Rogers,
Congratulations mate, well done!
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On 20 May 2011, at 07:38, Rogers Ochieng rogersochi...@gmail.com wrote:
Freshly minted! I took my exam yesterday in Bangalore and the good news is
here!!!
Thanks to my study partners Michael, Fatai and Rahul!
CCIE#28970 -
Hi all,
I do appologize for this OT, but I searched the whole google and big part of
cisco.com and couldn't find the answer to my question.
One of our customers have the following requirement:
when a caller calls, if he is in a queue too long (configured threshold), he
will be offered an option
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