Hi,
You can get digit manipulation to shown on the display of a IP Phone for calls
to a H.323 gw just as with an MGCP gw.
Add this to you H.323 config,
voice service voip
no supplementary-service h225-notify cid-update
Sincerely
Roger Källberg
CCIE #26199 (Voice)
Consultant
Cygate AB
hey guys, i truly have no clue what you are talking about :))) what VM has
to do with CFUR?
For and By fields are representing what you have configured as redirecting
Calling Party number mask (in this case redirecting ip phone) and what you
have configured as a destination for such calls (Unregis
Quite Strange,
Has anyone ran into this before, i integrated CUCM with Cisco unity connection
and every thing works including MWI , but when i try to ping my CUCM from the
servers in Cisco unity i get a timeout/ no response.
Though the AXL admin can successfully send a test.
Please i need s
I have seen this issue with both servers running in VMWare on the same hardware
not idea why but think it's probably VMWare related
On 18 Oct 2010, at 13:50, Pithog Oil wrote:
> Quite Strange,
>
> Has anyone ran into this before, i integrated CUCM with Cisco unity
> connection and every th
--- On Mon, 10/18/10, Graham Hopkins wrote:
Thanks i have been knocking my head for while now.
From: Graham Hopkins
Subject: Re: [OSL | CCIE_Voice] Cant ping CUCM from my UNITY CONNECTION SERVER
To: "Pithog Oil"
Cc: "ccie_voice@onlinestudylist.com"
Date: Monday, October 18, 2010, 12:56
Hi Experts
Am i suppose to calculate the T2 timer for DHCP whether its specified or not.
because in most cases i have seen only the T1 timer is given.
___
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www.ipex
Hi,
Can someone please explain me the difference between buffers and queue and
how they are implemented on 3750.
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
In the actual lab is it a good idea to Set my DHCP lease times to several days,
or use a local DHCP server to ensure that IP Phones do not reset themselves
when their lease expires during a WAN outage.
Can i be penalised for doing this.
___
if your dhcp lease is 24 hours, it does not mean that lease will expire
exactly 24 hours later and if dhcp server is not reachable then IP Address
will be released.
if DHCP lease is 24 hours, your phone will query the DHCP server after 12
hours to see if server is reachable, if it does the lease w
Hi Mark Holloway,
What bug are you talking about. Do you have a bug id?
On Mon, Oct 18, 2010 at 3:36 PM, sisiaji wrote:
> hey guys, i truly have no clue what you are talking about :))) what VM has
> to do with CFUR?
>
> For and By fields are representing what you have configured as redirecting
Thanks George for the hints.
The problem is solved. I enabled some debugs on the router and noticed that it
got the call and released it. I then went toward the CUCCX applications and
triggers, increased the session numbers , modified trigger numbers (it's not
relevant to this problem ,but
Hello
I couldn't register the 3 VGs of the 3 sites as MGCP on CUCM. I don't know
what's the problem, but this is the first time I face it. I did it several
times and worked successfully. I don't know what can cause this to happen.
here is my HQ-RTR config:
controller T1 0/0/0
framing esf
A great document to get started with is found here -
http://www.cisco.com/en/US/products/hw/switches/ps5023/tsd_products_support_
series_home.html
If that link doesn't work for some reason, you can access through the Doc CD
- Switches > LAN Switches - Access > Cisco Catalyst 3750 Series Switche
what does the show-ccmanager show ?
2010/10/18 Waleed Elhadidy
> Hello
>
> I couldn't register the 3 VGs of the 3 sites as MGCP on CUCM. I don't know
> what's the problem, but this is the first time I face it. I did it several
> times and worked successfully. I don't know what can cause this
Hi All,
I have the CUE integrated with SiteC CME and calls from HQ/B over
GK. Transcoder configued on SiteC for calls to CUE from WAN. When I call
3600 (CUE VM pilot) directly from HQ or SiteB phone I can get the CUE
prompt. However when I call 3002 and let it roll over to VM there is no
audio on
Hello Waleed,
What about the router domain name?
And the MGCP router name you added in the CUCM?
Is it similar?
What is their status in CUCM? Unknown? Rejected?
Tamer,
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Cristobal
Pri
Hi All,
I added the following two commands to the telephony service to enable
voiceview. I am able to login and see the greetings/messages. However cannot
play them back. Geta 404.
url services http://10.10.202.2/voiceview/common/login.do
url authentication http://10.10.202.1/CCMCIP/authenticate
If it's not VMWare, it may be windows firewall if you're running on 2008
server. Icmp reply gets knocked down by default
Sent from my phone, apologies for any typos.
On Oct 18, 2010, at 9:12 AM, Pithog Oil wrote:
>
>
> --- On Mon, 10/18/10, Graham Hopkins wrote:
> Thanks i have been knockin
Hi All,
I was working on a scenario where I need a number 1003 on a SIP CME
call-forward all to voicemail.
I created a voice register dn with number 1003 and call-forwaded to 1600 the
voicemail pilot.
When I dial this number I get fastbusy and debug shows no dial-peer with
1003 and there is no d
what does debug ccsip messages show you?
If you ring from CME to CUE does it work?
Can you provide a bit more info?
Cheers
On Tue, Oct 19, 2010 at 1:11 PM, David A wrote:
> Hi All,
>
> I was working on a scenario where I need a number 1003 on a SIP CME
> call-forward all to voicemail.
>
> I c
Hi Daniel,
I can reach Unity Connection (not a CUE) when I dial 1600 and from the
messages button. Also CFNA from 1002 (CME SIP ext) works fine.
Now I added a hunt group 1000
voice hunt-group 1 parallel
final 1003
list 1001,1002
timeout 12
pilot 1000
the final dest is 1003 with foll
voice
What is the purpose of doing it a SIP CME dn ?
For what you want, would be better just a regular SIP dial peer with a
translation rule that adds in the redirecting number:
voice translation-rule 1
rule 1 // /1003/
voice translation-rule 2
rule 1 /1003/ /1600/
voice translation-profile anythingu
Hi Randal,
I get your point. With this config how do I get an MWI for 1003 show up on
mailbox 1002?
Thanks,
DA
On Mon, Oct 18, 2010 at 9:50 PM, Randall Saborio wrote:
> What is the purpose of doing it a SIP CME dn ?
>
> For what you want, would be better just a regular SIP dial peer with a
> t
Hi all,
I'm trying to configure the CngPTP to transform (locallize) the calling
party number from E164 to 4 digit as per WB3, task2.4 stated.
I managed to transform the calling display to 4 digit, but some how the
missed and received call also displayed the 4 digit calling number
instead of E164.
Hi,
Please check that you have correct Calling Party CSS on the phone. Also make
sure the phone calling party css only sees that partition in which you have
the transformations.
HTH
Prashant
On Mon, Oct 18, 2010 at 10:46 PM, ShinGei Yong wrote:
> Hi all,
>
> I'm trying to configure the CngPTP t
You may check that under the port group in CUC, make sure you have the
subscriber set as the primary server and the publisher set as secondary.
Also, when you say ³Null partition & CSS², are you saying that the phones
and MWI are in the null partition?
HTH,
Amy
---
Amy Ryan CCIE #24677 (Voic
Hi Prashant,
Thanks for help
The correct CngCSS has been assigned to the DP, that why the calling party
E164 can be "localize" to 4 digit as mentioned earlier. The problem is, it
does "localize" the missed and received entry which is not desired. The
missed and received need to be in E164 form. I
Amy,
The subscriber is set as primary, while the publisher is set as secondary
server.
MWI is in Null partition, but phones are in the CSS that contains all
partitions. I'm able to call both MWI extensions for light on/off from all
phones.
Thanks for responding.
Francisco
Date: Mon, 18
Hi All,
I configured as below however after calling when I press 0 or 1 I go to the
3100 huntgroup. I never get the AA on 3500 even though I have param aa-hunt0
3500. I can reach 3100 and 3500 when dialing directly. What am I missing?
dial-peer voice 3000 voip
service aa
destination-pattern 300
Hi All,
More information submitted below:
CIPC (HQPH2, x5002, EPNM:2123945002, SIP)
IPBlue (HQPH5, x5005, EPNM: 2123945005, SCCP)
Globalization Configured:
PH5 call PH2
Main LCD Display: +12123945005, Missed & Received Entry: +12123945005
Vice versa for PH2 call PH5
DA,
In order to invoke dialing 0 from BACD you will need to use ³ param
aa-hunt10 3500 ³ vs ³ param aa-hunt0 3500 ³
HTH,
Amy
---
Amy Ryan CCIE #24677 (Voice)
Technical Instructor - IPexpert, Inc.
Mailto: ar...@ipexpert.com
Telephone: +1.810.326.1444
Live Assistance, Please visit: www.ipex
Hi Khaled,
I'm glad you found the problem.
I don't understand very well your problem with IPPA, the agent can't accept
the calls until he checks his status?
If I remember correctly, through IPPA, when agent logs on, he is NOT
automatically in "READY" state. So he has to change that, same as in C
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