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sansardanish
Member since
05-08-2015
Beginner
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
04-16-2019
06:02 AM
04-16-2019
06:02 AM
Hello Aadil, URL is outdated, do you have an update URL for this? Or steps?
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
02-01-2019
05:10 AM
02-01-2019
05:10 AM
Better to create a new thread than using a 10 year old thread.
And please upload related h323 logs to the thread
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
01-30-2019
02:46 AM
01-30-2019
02:46 AM
Is your registration user ID and password are accurate? Looks like there is some issue there.
Please have a look at the below
https://community.cisco.com/t5/collaboration-voice-and-video/handle-cube-registration-authentication-like-a-boss/ba-p/3660492
... View more
Created by
sansardanish
in Contact Center
in Contact Center
12-07-2018
03:32 AM
12-07-2018
03:32 AM
Hello Experts,
I am trying to set up voice bio metric authentication.
UCCX had to send call to a web URL, and URL will accept the call and compare the caller sound with the previously stored data. Then send back the call to UCCX with an ECC variable value which confirm the user authentication is successful or not. Can some please advice how can achieve this?
... View more
Labels:
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
10-24-2018
06:40 AM
10-24-2018
06:40 AM
Can you share debug sip messages
What fax protocol configured in ur ata 190??
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
07-24-2018
03:39 AM
07-24-2018
03:39 AM
Dear Prakash,
You don't require additional SIP trunk, one SIP trunk can handle as many calls depending upon the bandwidth of the connectivity.
Regarding the bandwidth consumption G.729 will consume 32 kbps and G711, will consume 87kbps per call.
https://www.voip-info.org/bandwidth-consumption/
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
07-18-2018
12:51 AM
07-18-2018
12:51 AM
Yes CME licenses are required.
Try using the telephony-service sertup and check maximum supported phones (max-ephone ?)
and then check show telephony service, which will give you the details of the CME version running on that if available
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
06-23-2018
10:34 PM
06-23-2018
10:34 PM
Just add it as below
Try as below
voice translation-rule 2 rule 1 /.*/ /35366850/
voice translation-rule 3 rule 1 /^9\ (........\)/ /\1/ ! voice translation-profile outbound-DID translate calling 2
translate called 3
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
05-05-2018
07:15 AM
05-05-2018
07:15 AM
What I understand is, there are two different CUCM clusters with one having E1 and other having SIP service provider connections. If you have WAN connectivity between clusters it users can call call each other through WAN irrespective of the PSTN connection type. Create a SIP trunk between cluster, or use inter cluster trunk (SIP will be preferred for me). If you are looking for something else please elaborate your question
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
03-28-2018
10:06 PM
03-28-2018
10:06 PM
As Dennis mentioned, this is mostly due to routing issue between the IP's involved. Please share complete messages for ccsip debug. is it a PRI or SIP ? What are the IP address's of CME, CUE ?
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
11-07-2017
12:29 AM
11-07-2017
12:29 AM
Are you able to resolve this?
... View more
08-13-2017
04:43 AM
Hello Experts,
I am using a third party SBC, in options method from and to are showing as same what could be the reason?
OPTIONS sip:10.242.32.50;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.161.126;rport;branch=z9hG4bK8g6cXp1KD2U3j
Max-Forwards: 70
From: <sip:NetBorder_Session_Controller@10.242.32.50;transport=udp>;tag=XygvUga3DecrQ
To: <sip:NetBorder_Session_Controller@10.242.32.50;transport=udp>
Call-ID: bbdfefc0-fab9-1235-3a9b-009027ee6464
CSeq: 1107 OPTIONS
Contact: <sip:NetBorder_Session_Controller@192.168.161.126:5060;transport=udp;gw=NGN-SIP-_trunk>
User-Agent: NetBorder Session Controller
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS
Allow-Events: talk, hold, presence, dialog, line-seize
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.161.126;branch=z9hG4bK8g6cXp1KD2U3j;rport=5060
Record-Route: <sip:10.242.32.50:9262;transport=udp;lr>
Call-ID: bbdfefc0-fab9-1235-3a9b-009027ee6464
From: <sip:NetBorder_Session_Controller@10.242.32.50;transport=udp>;tag=XygvUga3DecrQ
To: <sip:NetBorder_Session_Controller@10.242.32.50;transport=udp>;tag=sbc080602hxujo4
CSeq: 1107 OPTIONS
Content-Length: 0
... View more
Labels:
03-17-2017
10:19 AM
Hi,
Did you find an answer for this, We have the same issue in our environment.
Thank You
Danish
... View more
Created by
sansardanish
in IP Telephony and Phones
in IP Telephony and Phones
05-24-2016
03:27 AM
05-24-2016
03:27 AM
What message you are getting on phone?
... View more
Created by
sansardanish
in IP Telephony and Phones
04-16-2019 06:02 AM
04-16-2019 06:02 AM
Hello Aadil, URL is outdated, do you have an update URL for this? Or
steps?
Created by
sansardanish
in IP Telephony and Phones
02-01-2019 05:10 AM
02-01-2019 05:10 AM
Better to create a new thread than using a 10 year old thread. And
please upload related h323 logs t...
Created by
sansardanish
in IP Telephony and Phones
01-30-2019 02:46 AM
01-30-2019 02:46 AM
Is your registration user ID and password are accurate? Looks like there
is some issue there. Please...
Created by
sansardanish
in Contact Center
12-07-2018 03:32 AM
12-07-2018 03:32 AM
Hello Experts, I am trying to set up voice bio metric authentication.
UCCX had to send call to a web...
Created by
sansardanish
in IP Telephony and Phones
10-24-2018 06:40 AM
10-24-2018 06:40 AM
Can you share debug sip messages What fax protocol configured in ur ata
190??
Created by
sansardanish
in IP Telephony and Phones
07-24-2018 03:39 AM
07-24-2018 03:39 AM
Dear Prakash, You don't require additional SIP trunk, one SIP trunk can
handle as many calls dependi...
Created by
sansardanish
in IP Telephony and Phones
07-18-2018 12:51 AM
07-18-2018 12:51 AM
Yes CME licenses are required. Try using the telephony-service sertup
and check maximum supported ph...
Created by
sansardanish
in IP Telephony and Phones
06-23-2018 10:34 PM
06-23-2018 10:34 PM
Just add it as below Try as below voice translation-rule 2 rule 1 /.*/
/35366850/ voice translation-...
Created by
sansardanish
in IP Telephony and Phones
05-05-2018 07:15 AM
05-05-2018 07:15 AM
What I understand is, there are two different CUCM clusters with one
having E1 and other having SIP ...
Created by
sansardanish
in IP Telephony and Phones
03-28-2018 10:06 PM
03-28-2018 10:06 PM
As Dennis mentioned, this is mostly due to routing issue between the
IP's involved. Please share com...
Created by
sansardanish
in IP Telephony and Phones
11-07-2017 12:29 AM
11-07-2017 12:29 AM
Are you able to resolve this?
08-13-2017 04:43 AM
Hello Experts, I am using a third party SBC, in options method from and
to are showing as same what ...
03-17-2017 10:19 AM
Hi, Did you find an answer for this, We have the same issue in our
environment. Thank You Danish
Public Statistics
| Date Registered | 05-08-2015 03:52 AM |
| Date Last Visited | 05-15-2019 06:45 AM |
| Total Messages Posted | 20 |
| Total Helpful Votes Received | 5 |
Helpful Votes Given To
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