Turn on suggestions
Auto-suggest helps you quickly narrow down your search results by suggesting possible matches as you type.
Showing results for
- Cisco Community
- :
- About Daniel Flieth
Stats
Member since
03-07-2011
121
Posts
0
Helpful
1
Solution
Created by
Daniel Flieth
in Cloud Calling
in Cloud Calling
01-23-2020
03:08 AM
01-23-2020
03:08 AM
Hello, i am interested in adding phones to the Webex Control Center from my own Application. I guess it is possible to do something like this. But in the Webex Development Guide i dont saw a Document where adding and configuring of phones for the webex cloud will be descibed. But i guess i dont really found it. Does somebody can send me a document or link for this? Best Regards Daniel
... View more
Labels:
Created by
Daniel Flieth
in Other Collaboration Subjects
in Other Collaboration Subjects
12-17-2019
03:48 AM
12-17-2019
03:48 AM
Hello, i would like to show the current calls which are running on a ISR2 or ISR3 GW on my web page. It would be nice to get the informations from "show call active voice" command. Is there any API to get this live informations from the GW? Best Regards Daniel
... View more
Labels:
11-17-2019
09:30 AM
Hello, we change our ISP from ISDN to SIP. With the new ISP we use SRTP. With our phones it is no problem to call via this connection. The phones have certificats and all is fine. But with our VG310 i have a question. I know i can connect the VG310 via TLS to the CUCM. But is there also a way to use a Transcoding ressource? So that we dont have to change the VG310s and only configure a secure Transcoder on the CUBE? Best Regards Daniel
... View more
Labels:
11-13-2019
03:17 AM
Hello, does anybody have an example for a basic http trigger? At the moment i have a script which can be executed via telephone number but i alsow want to execute this via http trigger. When i add a trigger "test" i would like to execute this script with URL http://:9080/test But nothing happens. I only get the message 404 not found. Does anybody have a example for this? I use CCX 12.0 Best Regards Daniel
... View more
Labels:
Created by
Daniel Flieth
in IP Telephony and Phones
in IP Telephony and Phones
01-28-2019
07:41 AM
01-28-2019
07:41 AM
Hello,
is there any way to enroll a certificate to a 8821 phone automatically? Or enroll it automatically via console connection?
Best Regards
Daniel
... View more
- Tags:
- 8821
- certificate
Labels:
09-20-2018
08:59 AM
Hallo,
we want to configure an wallboard (Dashboard) for our UCCX. We would like to see all current logged in Agents. When we create a live Report for this and add this Report to the Dashboard we get all informations we want.
But it would be nice if this Dashboard would Show the informations as soon as we open the Permlink.
At the Moment we have to press "run" at the Dashboard to execute the Report.
Does anybody know how to configure this?
Best Regards
Daniel
... View more
Labels:
Created by
Daniel Flieth
in Switching
in Switching
10-20-2017
06:40 AM
10-20-2017
06:40 AM
Sorry, wrong Community, but dont know how to move this.
... View more
10-20-2017
06:37 AM
Hi, our customer has CMS 2.2.5 installed and can call from lync to meetingroom. In CUCM a Spark RoomKit is configured. My question is, can the customer call from lync client a specific Spark Roomk kit?
The Meetingserver is only accessable via LAN, no external DMZ connection is configured.
Is this possible?
Best Regards
Daniel
... View more
Labels:
10-20-2017
06:05 AM
Thank you very much Julio, i will try this tomorrow at customer site.
... View more
10-20-2017
04:51 AM
Hello, our customer changed his old HP Switches to new cisco Switches. Now with his old wireless lan Controller and APs he has some issues with 802.1X authentication.
In the Wireless Controller we see now IPv6 Adresses. THis is the only difference to the old situation.
Now we would like to disable ipv6 on this Switches. Is this possible?
... View more
Labels:
Created by
Daniel Flieth
in IP Telephony and Phones
in IP Telephony and Phones
03-15-2017
05:52 AM
03-15-2017
05:52 AM
Thank you very much for this information, it worked immidiatly.
... View more
Created by
Daniel Flieth
in IP Telephony and Phones
in IP Telephony and Phones
03-15-2017
05:52 AM
03-15-2017
05:52 AM
Thank you very much for this information, it worked immidiatly.
... View more
Created by
Daniel Flieth
in IP Telephony and Phones
in IP Telephony and Phones
03-14-2017
07:52 AM
03-14-2017
07:52 AM
Hi, we have a problem with early media.
We changed the connection between cucm and gateway from H323 to SIP.
Now the customer tells us, that there is a PSTN number which played an announcement about the costs of this call. This worked with H323.
SInce we changed it to SIP, we dont get this announcemen.
To be honest, i really dont know much about early media.
Attached you see the isdn q931 debug.
As is understand, this:
Progress Ind i = 0x8488 - In-band info or appropriate now available
should indicate, that there is early media available.
Is this correct?
If there is early media, i dont know how to transfer this information via SIP to CUCM.
Does anybody know how to solve this?
Best Regards
Daniel
Mar 14 11:18:57 CET: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x24EF Sending Complete Bearer Capability i = 0x9090A3 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Progress Ind i = 0x8183 SEISR1(cfg-translation-rule)# - Origination address is non-ISDN Calling Party Number i = 0x0180, '289XXXX' Plan:ISDN, Type:Unknown Called Party Number i = 0x81, '199XXXX' Plan:ISDN, Type:Unknown Mar 14 11:18:57 CET: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0xA4EF Channel ID i = 0xA98383 Exclusive, Channel 3 SEISR1(cfg-translation-rule)# Mar 14 11:18:58 CET: ISDN Se0/0/0:15 Q931: RX <- PROGRESS pd = 8 callref = 0xA4EF Progress Ind i = 0x8482 - Destination address is non-ISDN Progress Ind i = 0x8488 - In-band info or appropriate now available SEISR1(cfg-translation-rule)#
... View more
Labels:
Created by
Daniel Flieth
in IP Telephony and Phones
in IP Telephony and Phones
11-01-2016
01:10 PM
11-01-2016
01:10 PM
Hello,
we want to change the configuration on our PSTN Gateway from H323 to SIP.
We use overlap sending, so the CUCM sends each digit one by one, not enbloc. When using SIP, this dont work with our configuration.
I thought with KPML this would be possible.
Does anybody know how to configure this? Or is this not possible with SIP?
Best Regards
Daniel
... View more
Labels:
01-23-2020
Hello,i am interested in adding phones to the Webex Control Center from
my own Application. I guess ...
12-17-2019
Hello, i would like to show the current calls which are running on a
ISR2 or ISR3 GW on my web page....
11-17-2019
Hello,we change our ISP from ISDN to SIP. With the new ISP we use SRTP.
With our phones it is no pro...
11-13-2019
Hello, does anybody have an example for a basic http trigger? At the
moment i have a script which ca...
08-23-2019
Hello, our customer has some external Video Devices which should be
registered on expressway. Becaus...
07-24-2019
Hello, as far as i know azure AD is still not supported for SSO. But my
customer asks if there are a...
Created by
Daniel Flieth
in Collaboration Applications
06-05-2019
06-05-2019
Hi, is this issue fixed? Because with Jabber 12.5.1 we still dont have
notification for Chat Rooms.W...
Created by
Daniel Flieth
in Collaboration Applications
02-04-2019
02-04-2019
Hello, i have a question related to the Jabber File Transfer. I know
File Transfer is not Supported ...
01-28-2019
Hello, is there any way to enroll a certificate to a 8821 phone
automatically? Or enroll it automati...
09-20-2018
Hallo, we want to configure an wallboard (Dashboard) for our UCCX. We
would like to see all current ...
Created by
Daniel Flieth
in TelePresence and Video Infrastructure
05-07-2018
05-07-2018
Hi we would like to connect from an Skype Client in the cloud to an
Meeting Room in CMS and want to ...
Created by
Daniel Flieth
in Unified Communications Infrastructure
03-26-2018
03-26-2018
Hi, is there any way to create policys to prevent some devices from
loggin in via MRA? When MRA is e...
Created by
Daniel Flieth
in Collaboration Applications
03-07-2018
03-07-2018
Hi, i just want to ask if its possible to save the personal Chat history
with any contact to an data...
Created by
Daniel Flieth
in Collaboration Applications
11-06-2017
11-06-2017
Does anybody have an idea?
Public Statistics
| Date Registered | 03-07-2011 11:05 PM |
| Date Last Visited | 03-20-2020 12:07 AM |
| Total Messages Posted | 121 |
Helpful Votes Given To
| User | Helpful Count |
|---|---|
| 8 | |
| 5 | |
| 5 | |
| 5 | |
| 5 |
.jpg "Hall of Fame Cisco Employee"){kind=icon}
.jpg "VIP Collaborator"){kind=icon}
