About roger.kallberg

roger.kallberg · ‎03-05-2019

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.

roger.kallberg · ‎11-12-2015

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

roger.kallberg · ‎11-09-2015

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

roger.kallberg · ‎10-29-2015

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.

roger.kallberg · ‎10-28-2015

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,

roger.kallberg · ‎10-28-2015

Same procedure for SIP and SCCP SRST. The phones will loose connection to CM and then fail over into SRST.

roger.kallberg · ‎10-28-2015

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

roger.kallberg · ‎10-28-2015

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.

roger.kallberg · ‎10-28-2015

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

roger.kallberg · ‎10-28-2015

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

roger.kallberg · ‎10-27-2015

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.

roger.kallberg · ‎10-27-2015

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

roger.kallberg · ‎10-27-2015

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

roger.kallberg · ‎10-23-2015

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

roger.kallberg · ‎10-23-2015

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.

roger.kallberg · 03-05-2019

Trunks groups can be configured under controllers for PRIs also. It's not just for CAS configuration...

roger.kallberg · 09-25-2018

Have you tried to start CIPC as Administrator?

roger.kallberg · 11-12-2015

I've also had this issue, you can try to add this "no tone dialtone remote-onhook" under your dial-p...

roger.kallberg · 11-09-2015

Hi, Can you please provide the out-put from deb voip dial deb voip trans when you call to the number...

roger.kallberg · 10-30-2015

Hi, A option would be to add a partition that holds a translation pattern that matched your entire D...

roger.kallberg · 10-29-2015

As said, there are many ways to do almost all things. :) The main objection I have to stop the CM pr...

roger.kallberg · 10-28-2015

It's a matter of opinion and personal preference what is the recommended way to put a site into SRST...

roger.kallberg · 10-28-2015

Same procedure for SIP and SCCP SRST. The phones will loose connection to CM and then fail over into...

roger.kallberg · 10-28-2015

Hi, my prefered method is to use routes to null, primarily because it's much easier and less prone t...

roger.kallberg · 10-28-2015

The lines don't have to be in different partitions, they only need to have different CSS. SDs do not...

roger.kallberg · 10-28-2015

Hi John, There are always more then one way to skin a cat :) Glad that you got it to work. Please ra...

roger.kallberg · 10-28-2015

The translation you have will be triggered on the inbound direction, IE for calls from fax-a. From y...

roger.kallberg · 10-27-2015

Try with setting the calling number to the same format as the Telco send incoming called number to y...

roger.kallberg · 10-27-2015

For the incoming part you are missing the translation profile, add this to your config.voice transla...

roger.kallberg · 10-27-2015

Hi,The inbound called number from PSTN is "Called Number=8038055301" but your DNs are 3-digits, for ...

Date Registered	‎02-16-2004 12:38 AM
Date Last Visited	‎09-09-2019 01:34 AM
Total Messages Posted	414
Total Helpful Votes Received	390

Re: PRI Trunk group configuration

I've also had this issue, you

Hi,

As said, there are many ways

It a matter of opinion what

Same procedure for SIP and

Hi,

The lines don't has to be in

Hi John,

The translation you have will

Try with setting the calling

For the incoming part you are

Hi,The inbound called number

Hi,Are you sure that the pin

If you do decide to change