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- About roger.kallberg
06-21-2019
roger.kallberg
Member since
02-16-2004
Rising star
I have experience of work- and project management and leadership. Have worked with presale, investigations/analysis, design/installation/implementations of unified communication infrastructure, documentation, security issues and support.
I'm very customer oriented and have great abilities to find constructive solutions even within tight time frames. My work is target focused and I draw good use of my background to see important economical connections. Verbal with great abilities to express myself in both in speech and writing, and very particular and ambitious. I have excellent social and interpersonal skills with long experience in dealing with people.
Specialties
Unified communication, IP telephony, IP Contact Center, Voice mail, Routing (OSPF, EIGRP, RIP), Switching (VLAN, VTP, STP, Trunking), QoS, Multicast, PPP, ISDN, FR, X.25, DDR, VPN, DMZ, IPSEC, IKE, AAA.
I have experience of work- and project management and leadership. Have worked with presale, investigations/analysis, design/installation/implementations of unified communication infrastructure, documentation, security issues and support.
I'm very customer oriented and have great abilities to find constructive solutions even within tight time frames. My work is
411
Posts
385
Helpful
65
Solutions
Recent Badges
Awards
2
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
03-05-2019
05:53 AM
03-05-2019
05:53 AM
Trunks groups can be configured under controllers for PRIs also. It's not just for CAS configurations. The referenced Cisco IOS Voice Command Reference is quite old as it's for 12.3 I believe.
Where you choose to configure is a matter of preference. Personally I prefer to do it on the controllers.
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
11-12-2015
11:18 AM
11-12-2015
11:18 AM
I've also had this issue, you can try to add this "no tone dialtone remote-onhook" under your dial-peer to see if that fixes this. It did for me on various models of VGs with SCCP controlled FXS ports.
An example dial-peer voice 214 pots no tone dialtone remote-onhook service stcapp port 2/14
Sincerely
Roger
... View more
11-09-2015
03:03 PM
Hi,
Can you please provide the out-put from
deb voip dial
deb voip trans
when you call to the number you listed. Also the out-put from this show command would be helpful.
sh isdn status
Sincerely
Roger
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-29-2015
01:59 AM
10-29-2015
01:59 AM
As said, there are many ways to do almost all things. :)
The main objection I have to stop the CM process is that it will affect all sites at once. This might not be what is wanted, but then again it might. "It depends" is a favourite expression that can be fitted into most discussions.
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-28-2015
11:49 PM
10-28-2015
11:49 PM
It's a matter of opinion and personal preference what is the recommended way to put a site into SRST.
If you where to use my suggestion or Ayodeji's depends on other factors in your network, like route redistribution and if the goal is to only test with a subset of phones at the site.
As always there are more than one way to skin a cat, it's at your discretion to select what suits you best from the options you have presented,
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-28-2015
01:45 PM
10-28-2015
01:45 PM
Same procedure for SIP and SCCP SRST. The phones will loose connection to CM and then fail over into SRST.
... View more
10-28-2015
12:22 PM
Hi,
my prefered method is to use routes to null, primarily because it's much easier and less prone to mistakes. Given your example this would be what I would use.
ip route 10.10.1.24 255.255.255.255 null0
This way you only affect traffic to CM, it will not block any other traffic from the site.
Sincerely
Roger
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-28-2015
11:22 AM
10-28-2015
11:22 AM
The lines don't have to be in different partitions, they only need to have different CSS. SDs do not have any PT or CSS, they use the CSS of the device/line.
... View more
10-28-2015
08:39 AM
Hi John,
There are always more then one way to skin a cat :) Glad that you got it to work.
Please rate all useful post.
Sincerely
Roger
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-28-2015
05:59 AM
10-28-2015
05:59 AM
The translation you have will be triggered on the inbound direction, IE for calls from fax-a. From your description it sounds like you want to translate all calls to fax-a to send them on to fax-b, to do that you need to put the translation at an earlier place in the call flow.
But instead of mucking around with translations the absolute easiest way to achieve this would in my book be to change the port on your dial peer to the port with the operational fax machine.
dial-peer voice 402 pots destination-pattern 4296$ port 0/0/ 3
Sincerely
Roger
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-27-2015
01:06 PM
10-27-2015
01:06 PM
Try with setting the calling number to the same format as the Telco send incoming called number to you. In the original debug it shows as " Called Number= 8038055301 (TON=Subscriber, NPI=ISDN)" IE Telco send it in type subscriber and plan ISDN.
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-27-2015
11:23 AM
10-27-2015
11:23 AM
For the incoming part you are missing the translation profile, add this to your config. voice translation-profile PSTN translate called 1 For the outgoing it doesn't affect the calling number as your configuration looks currently. You need to translate from your 3-digit numbers to the number that your Telco expect. Probably you need to translate from 301 to 803855301. Create a new rule set and profile and attach that in the outbound direction on your voice port. Sincerely Roger
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-27-2015
10:55 AM
10-27-2015
10:55 AM
Hi, The inbound called number from PSTN is " Called Number= 8038055301" but your DNs are 3-digits, for example 301. That's why the incoming call fails. You must either translate the incoming called number to your 3-digits numbers or change the DNs so that they match the full number from your Telco. This is also the reason, but in reverse to incoming reason, why the number shown for outbound calls isn't shown as the individual DNs. Sincerely Roger
... View more
Created by
roger.kallberg
in IP Telephony and Phones
in IP Telephony and Phones
10-23-2015
11:51 PM
10-23-2015
11:51 PM
Hi, Are you sure that the pin out on the phones match the FXS ports? For more information about this see this url, http://www.cisco.com/c/en/us/support/docs/voice/analog-signaling-e-m-did-fxs-fxo/7938-2fxs-vic.html#info
... View more
10-23-2015
11:50 PM
If you do decide to change the case in AD so it's consistent please bear in mind that this will remove the current configuration for those agents in UCCX, like team belongings and skills. :-/ Not that hard to add back, but sort of a pita if you're not aware of it beforehand.
... View more
Created by
roger.kallberg
in IP Telephony and Phones
03-05-2019 05:53 AM
03-05-2019 05:53 AM
Trunks groups can be configured under controllers for PRIs also. It's
not just for CAS configuration...
Created by
roger.kallberg
in Collaboration Applications
09-25-2018 10:24 AM
09-25-2018 10:24 AM
Have you tried to start CIPC as Administrator?
Created by
roger.kallberg
in IP Telephony and Phones
11-12-2015 11:18 AM
11-12-2015 11:18 AM
I've also had this issue, you can try to add this "no tone dialtone
remote-onhook" under your dial-p...
11-09-2015 03:03 PM
Hi, Can you please provide the out-put from deb voip dial deb voip trans
when you call to the number...
10-30-2015 01:24 PM
Hi, A option would be to add a partition that holds a translation
pattern that matched your entire D...
Created by
roger.kallberg
in IP Telephony and Phones
10-29-2015 01:59 AM
10-29-2015 01:59 AM
As said, there are many ways to do almost all things. :) The main
objection I have to stop the CM pr...
Created by
roger.kallberg
in IP Telephony and Phones
10-28-2015 11:49 PM
10-28-2015 11:49 PM
It's a matter of opinion and personal preference what is the recommended
way to put a site into SRST...
10-28-2015 01:45 PM
Same procedure for SIP and SCCP SRST. The phones will loose connection
to CM and then fail over into...
10-28-2015 12:22 PM
Hi, my prefered method is to use routes to null, primarily because it's
much easier and less prone t...
Created by
roger.kallberg
in IP Telephony and Phones
10-28-2015 11:22 AM
10-28-2015 11:22 AM
The lines don't have to be in different partitions, they only need to
have different CSS. SDs do not...
10-28-2015 08:39 AM
Hi John, There are always more then one way to skin a cat :) Glad that
you got it to work. Please ra...
Created by
roger.kallberg
in IP Telephony and Phones
10-28-2015 05:59 AM
10-28-2015 05:59 AM
The translation you have will be triggered on the inbound direction, IE
for calls from fax-a. From y...
Created by
roger.kallberg
in IP Telephony and Phones
10-27-2015 01:06 PM
10-27-2015 01:06 PM
Try with setting the calling number to the same format as the Telco send
incoming called number to y...
Created by
roger.kallberg
in IP Telephony and Phones
10-27-2015 11:23 AM
10-27-2015 11:23 AM
For the incoming part you are missing the translation profile, add this
to your config.voice transla...
Created by
roger.kallberg
in IP Telephony and Phones
10-27-2015 10:55 AM
10-27-2015 10:55 AM
Hi,The inbound called number from PSTN is "Called Number=8038055301" but
your DNs are 3-digits, for ...
Public Statistics
| Date Registered | 02-16-2004 12:38 AM |
| Date Last Visited | 06-21-2019 01:02 AM |
| Total Messages Posted | 411 |
| Total Helpful Votes Received | 385 |
Helpful Votes Given To
Helpful Votes From
| User | Helpful Count |
|---|---|
| 5 | |
| 10 | |
| 5 | |
| 5 | |
| 30 |
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Mobile App Contributor October 2012
