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- About Asad Raza
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Member since
05-31-2011
51
Posts
5
Helpful
0
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Created by
Asad Raza
in Network Security
in Network Security
12-27-2017
04:50 AM
12-27-2017
04:50 AM
Hi everyone,
I have a quick question regarding time-based access for self-registering and sponsor-approved guest accounts on Cisco ISE's guest portal. I understand that if a guest type has the "Allow access only on these days and times" time range defined, users cannot self-register (for self-registered hotspot portals) or log in even with active guest accounts (self-registered or sponsor-approved).
However, if a guest has an active account and they log in right before the "outside business hours" range is about to start, will the user remain logged in even after they have crossed over into the access-restricted time range, or will they be automatically disconnected? If not, is this something that can be configured? Thanks in advance!
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12-13-2014
08:59 PM
Hi Manannalage, You're right, that was the issue. We've notified the client that they need to upgrade the lwapp image. Thanks for your help! Regards, Asad
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Created by
Asad Raza
in Wireless and Mobility
in Wireless and Mobility
12-11-2014
02:26 AM
12-11-2014
02:26 AM
Hello everyone, I am trying to register a 1602i with a WLC 5508, but I am unable to do so. Here is the information I currently have: WLC: Model: 5508 Boot version: 12.4.25.1 IOS version: 12.4(25e)JA1$ Mini IOS version: 7.4.1.37 Access point: Model: AIR-CAP1602I-E-K9 IOS version: 15.2(4)JA1 LWAPP image: 7.5.1.73 The WLC is already controlling 1242 APs without issues. The 1602Is however, are unable to register with this WLC. There is no Option 60 or 43 defined on the DHCP server, however, both the WLC and the APs are on the same subnet. I am receiving the following messages on the console (where x denotes customer information that has been "Xed" out due to privacy reasons): Translating "CISCO-CAPWAP-CONTROLLER.XXXXXX.COM"...domain server (x.x.x.x) *Mar 1 00:02:55.070: %CAPWAP-3-ERRORLOG: Did not get log server settings from DHCP. *Mar 1 00:02:58.070: %CAPWAP-3-ERRORLOG: Could Not resolve CISCO-CAPWAP-CONTROLLER.XXXXXX.COM *Mar 1 00:03:08.070: %CAPWAP-3-ERRORLOG: Discovery response from MWAR 'XXX-XXX-XX'running version 7.2.110.0 is rejected. *Mar 1 00:03:08.070: %CAPWAP-3-ERRORLOG: Failed to decode discovery response. *Mar 1 00:03:08.070: %CAPWAP-3-ERRORLOG: CAPWAP SM handler: Failed to process message type 2 state 2. *Mar 1 00:03:08.070: %CAPWAP-3-ERRORLOG: Failed to handle capwap control message from controller *Mar 1 00:03:08.070: %CAPWAP-3-ERRORLOG: Failed to process unencrypted capwap packet from x.x.x.x *Mar 1 00:03:53.569: %CAPWAP-3-DHCP_RENEW: Could not discover WLC using DHCP IP. Renewing DHCP IP. *Mar 1 00:03:56.577: %CAPWAP-3-ERRORLOG: Invalid event 38 & state 2 combination. *Mar 1 00:03:56.645: %DHCP-6-ADDRESS_ASSIGN: Interface BVI1 assigned DHCP address x.x.x.x, mask x.x.x.x, hostname APxxxx.xxxx.xxxx Translating "CISCO-CAPWAP-CONTROLLER.OAMC.COM"...domain server (x.x.x.x) *Mar 1 00:04:03.569: %CAPWAP-3-ERRORLOG: Did not get log server settings from DHCP. I tried entering a static IP address, static controller IP address, static default gateway address, but I receive the following messages and the AP reboots: *Mar 1 00:14:49.463: %CAPWAP-3-STATIC_TO_DHCP_IP: Could not discover WLC using static IP. Forcing AP to use DHCP. I would appreciate any assistance with this problem! Regards, Asad Raza
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
08-13-2014
01:26 AM
08-13-2014
01:26 AM
Hello everyone, I have run into a bit of a situation. I was setting up an auto attendant service for a client, and ended up mistakenly recording the words "test test" in my voice over the default "Standard" opening greeting. Now, when users dial into voicemail, they hear the "test test"! I have quickly changed the Direct Routing Rule to foward to the "Greeting Administrator" which sounds better than my message, but I need to revert this back to the original opening greeting, which sounded something like: "Welcome to the Cisco Unity Connection messaging system..." Does anyone know how to restore this greeting? Is this file available somewhere on Unity? I would appreciate some urgent assistance! Thanks!
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08-03-2014
05:10 AM
Hi Suresh, Thanks for your reply. Yes, I do have that option checked. Actually, I just resolved the issue. It turns out, the VM ports did not have a CSS assigned. I've assigned a CSS to them containing partitions for all user extensions that should be reachable through Unity Connection. Thanks for your help anyway! Regards, Asad Raza
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
08-03-2014
04:19 AM
08-03-2014
04:19 AM
Hello everyone, I am trying to configure a basic Auto Attendant using the built-in greeting menu. The issue I am having is that when I get to the greeting menu, I am asked to "dial an extension at any time." When I dial the number, I hear "Sorry, <username> is not available. Record your message at the tone..." The way I want this to function is that when I dial the extension of the user, it should ring the phone first, and after the defined number of rings if the phone is not answered, the call goes to voicemail. I checked the Standard transfer rule and the only two options I have under this option are "Greeting" and "Extension." Neither of the two seem to be applicable to what I am trying to achieve. Any help would be greatly appreciated!
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
12-10-2012
03:02 AM
12-10-2012
03:02 AM
Hi Gajanan, They're running the following versoin: SCCP11.9-2-1S Thanks. Regards, Asad
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
12-10-2012
01:42 AM
12-10-2012
01:42 AM
Hello folks, We recently upgraded one of our clients from CCM 4.2(3) to CUCM 8.6(2a). All seems fine except for the fact that certain 7911 IP phones are expriencing a strange problem. For any calls (internally or externally originated) some of the 7911 phones ring normally, but lifting the receiver does not answer the phone call. The user has to press the speaker button to answer the call, and the same to hang up. Any ideas on what may be causing this? Thanks in advance. Regards, Asad Raza
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11-22-2012
09:46 AM
Hi Robert, It turns out that it was a codec mismatch problem. The incoming dial peer was statically configured to use a single codec. Once we created a voice class with all the commonly used codecs, everything started working just fine. The hold, transfer, and MoH problems all went away. In fact, it resolved another problem that that the client was experiencing as well, which was that whena remote phones were called from the PSTN, the voicemail did not engage. Once the codecs situtation at the dial peer was resolved, the voicemail started working too! Thanks for all your help! Regards, Asad
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2012
07:24 AM
11-21-2012
07:24 AM
Hello, Thank you for your reply. Yes I went through the upgrade path documentation but I didn't want to go through the DMA process. Instead, I have done a fresh install and although I have modified the csv files to adapt to version 8.x, the problem does not lie in the formatting, but the license units. After exportation, BAT does not import all phones due to a lack of licenses for the phones however, the client has purchased 600 UCLs, which should be more than sufficient to cover all 400 or so users. The license unit report show 1200 licenses, and 1258 being use, with an overage of 58. These licenses to my knowledge are cost free and are calculated based on the type of phone 3 for 7911, 4 for 7965, etc. Any ideas how I can get more of these licences without having to do a reinstall and run DMA? Thanks!
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
11-21-2012
02:32 AM
11-21-2012
02:32 AM
Hello all, I read somewhere that in order to use DMA, the XML file would have to be inserted during the installation process however, we have already installed the new version and are attempting to migrate the phones over using BAT, but are unsuccessful. Only part of the phones have transferred, whereas the remaining phones are being rejected from being imported by BAT. A Cisco licensing rep. has told me that I do not need to do a reinstall. Can anyone clarify whether I can proceed with the migration without using DMA? Thanks! Regards, Asad
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11-20-2012
10:29 PM
Hi Robert, Thank you for your response. I don't know if I should have mentioned this, but all communication between the HQ and the remote offices takes place over a VPN connection through their global HQ in UK. Most likely they are running some kind of firewall, but I fail to understand how it would interfere in the operation since the MTP at the remote site is on the local gateway, and the local gateway is able to communicate with the CUCM. Also, the MTP shows up as registered on the CUCM. Any other ideas? Regards, Asad
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
11-20-2012
05:47 AM
11-20-2012
05:47 AM
Hello everyone, One of our clients is experiencing an issue with being able to perform a simple hold and transfer function at their remote sites. They currently have a CUCM 8.6 at their HQ, with two remote sites using the WAN for signaling, but PSTN for all external calls, even for intersite dialing. Their remote site gateways are running an AA tcl script. Users at the HQ are experiencing no problems with any fucntions however, if a user from the HQ dials a remote site, the AA script kicks in as expected, and the extension is dialed to reach a user. Once the appropriate extension is dialed and the call is connected to the user, the call proceeds normally. However, if the user at the remote site places the HQ user on hold, MOH is not streamed. Also, if the remote site user attempts to transfer the HQ user to another internal extension at the remote site, no music, nor any ringback is heard. The call is transferred, but audio becomes one-way only where the HQ user can hear the remote site user, but the not the other way around. I would also like to point out that all gateways are registered as a SIP trunk to CUCM whereas all IP phones use SCCP. Could this be a problem? Also, I have configured the phones, MRGs/MRGLs/MoH/MTP at HQ and remote sites in exactly the same manner. However, I have registered a hardware MTP resource on the remote site gateway with the CUCM and assigned it to its device pool (Device pool for this remote site). Can someone shed some light on what the issue may be? Any help would be greatly apprecaited. Thank you! Update: I have attached a snapshot of the topology to better demonstrate the current network. Regards, Asad
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Created by
Asad Raza
in IP Telephony and Phones
in IP Telephony and Phones
03-23-2012
02:13 AM
03-23-2012
02:13 AM
Hello Ali, Can you provide the log from the diagnostic test? From the CLI, pull up the active log by run the following command: file list activelog tomcat/logs See if you can find a file titled something along the lines of: diagnostic-info.21849.tomcat.1332348357.txt to view the file, type the following command in the CLI: file view activelog tomcat/logs/ Where the parameter is to be replaced with the actual file name. Once you have the output, please attach it here. It may provide us with further details.
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01-24-2012
12:47 AM
Hi, I just realized that it was as simple as configuring the target DN in the "speed dial" field under the intercom settings. Thanks for your help!
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Created by
Asad Raza
in Network Security
12-27-2017
12-27-2017
Hi everyone, I have a quick question regarding time-based access for
self-registering and sponsor-ap...
12-13-2014
Hi Manannalage, You're right, that was the issue. We've notified the
client that they need to upgrad...
Created by
Asad Raza
in Wireless and Mobility
12-11-2014
12-11-2014
Hello everyone,I am trying to register a 1602i with a WLC 5508, but I am
unable to do so. Here is th...
12-11-2014
Hi everyone, I was able to resolve the problem with a quick fix and I
hope this helps someone else o...
10-28-2014
Thanks for your response, Anirudh. I did modify the "Default System
Transfer" table and just to be o...
10-28-2014
Hi Johari, Thanks for your response. The CME is added to CUCM as a H.323
gateway and a route pattern...
Created by
Asad Raza
in Unified Communications Infrastructure
10-28-2014
10-28-2014
Hi everyone, I set up an AA service for a client where a call comes into
a PRI, hits a CTI RP which ...
Created by
Asad Raza
in IP Telephony and Phones
08-13-2014
08-13-2014
Hello everyone, I have run into a bit of a situation. I was setting up
an auto attendant service for...
08-03-2014
Hi Suresh, Thanks for your reply. Yes, I do have that option checked.
Actually, I just resolved the ...
08-03-2014
Hello everyone,I am trying to configure a basic Auto Attendant using the
built-in greeting menu. The...
Created by
Asad Raza
in IP Telephony and Phones
12-10-2012
12-10-2012
Hi Gajanan,They're running the following versoin:
SCCP11.9-2-1SThanks.Regards,Asad
Created by
Asad Raza
in IP Telephony and Phones
12-10-2012
12-10-2012
Hello folks,We recently upgraded one of our clients from CCM 4.2(3) to
CUCM 8.6(2a). All seems fine ...
11-22-2012
Hi Robert,It turns out that it was a codec mismatch problem. The
incoming dial peer was statically c...
11-21-2012
Hello,Thank you for your reply. Yes I went through the upgrade path
documentation but I didn't want ...
Public Statistics
| Date Registered | 05-31-2011 11:59 PM |
| Date Last Visited | 09-18-2019 01:01 AM |
| Total Messages Posted | 51 |
| Total Helpful Votes Received | 5 |
Helpful Votes Given To
| User | Helpful Count |
|---|---|
| 5 | |
| 5 | |
| 5 | |
| 5 | |
| 5 |
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