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Today's Topics:
1. IOS GW licensing for 3rd-party PBX (Norton, Mike)
2. UCCX 9 SRND question (Paul)
3. Re: IOS GW licensing for 3rd-party PBX (Ed Leatherman)
4. Re: IOS GW licensing for 3rd-party PBX (Ted Nugent)
5. Re: FXS over VPN (Paul)
6. Re: IOS GW licensing for 3rd-party PBX (Norton, Mike)
7. Re: Can a CUBE be used to connect CUCM and Asterisk to a SIP
Trunk (Matthew Saskin)
8. Re: Can CTS-CTRL-DV8 be used as a Phone for end user?
(Matthew Saskin)
----------------------------------------------------------------------
Message: 1
Date: Fri, 7 Dec 2012 19:30:23 +0000
From: "Norton, Mike" <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] IOS GW licensing for 3rd-party PBX
Message-ID:
<2c4107b204f8e74eb0a63ef59b0f97024a9...@pwsdexchange06.pwsb33.ab.ca>
Content-Type: text/plain; charset="us-ascii"
Can someone confirm what IOS licensing I need to use a 2900 as a simple
SIP-to-POTS gateway for a non-Cisco IP-PBX? Until now all my gateways have been
for CUCM/MGCP, so I've just been buying the SRST bundles.
I'm thinking I don't need an SRST or CME license, right? As long as I have UC
license I should be good?
I know my account team could answer this but it might be best for now if they
didn't know we were piloting a non-Cisco system. ;-)
--
Mike Norton
I.T. Specialist
Peace Wapiti School Division No. 76
Helpdesk: 780-831-3080
Direct: 780-831-3076
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Message: 2
Date: Fri, 7 Dec 2012 13:00:47 -0800 (PST)
From: Paul <[email protected]>
To: Cisco Vo IP Group <[email protected]>
Subject: [cisco-voip] UCCX 9 SRND question
Message-ID:
<[email protected]>
Content-Type: text/plain; charset=iso-8859-1
UCCX 9 SRND says "A Unified CCX system can provide up to 300 logical IVR ports
(also called CTI Ports)." while?Maximum number of active inbound agents
supported = 400.?
Is this correct or does the documentation need to be updated?
------------------------------
Message: 3
Date: Fri, 7 Dec 2012 16:01:49 -0500
From: Ed Leatherman <[email protected]>
To: "Norton, Mike" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] IOS GW licensing for 3rd-party PBX
Message-ID:
<CAFC4dspJpyD3fDmMkA2bmct458yN2eaietLwuJ=12dx97-x...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
I think the UC license would be all you need.
On Fri, Dec 7, 2012 at 2:30 PM, Norton, Mike <[email protected]>wrote:
> Can someone confirm what IOS licensing I need to use a 2900 as a simple
> SIP-to-POTS gateway for a non-Cisco IP-PBX? Until now all my gateways have
> been for CUCM/MGCP, so I?ve just been buying the SRST bundles.****
>
> ** **
>
> I?m thinking I don?t need an SRST or CME license, right? As long as I have
> UC license I should be good?****
>
> ** **
>
> I know my account team could answer this but it might be best for now if
> they didn?t know we were piloting a non-Cisco system. ;-)****
>
> ** **
>
> -- ****
>
> Mike Norton****
>
> I.T. Specialist****
>
> Peace Wapiti School Division No. 76****
>
> Helpdesk: 780-831-3080****
>
> Direct: 780-831-3076****
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
--
Ed Leatherman
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Message: 4
Date: Fri, 7 Dec 2012 16:03:50 -0500
From: Ted Nugent <[email protected]>
To: "Norton, Mike" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] IOS GW licensing for 3rd-party PBX
Message-ID:
<cahs2vys6ujac5n3+y6juqj00+cz6g38utjk8jrnumqg+7ug...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Right, for a straight up SIP/H323/MGCP gateway you don't need SRST or CME
for that.... UC would be sufficient.
On Fri, Dec 7, 2012 at 2:30 PM, Norton, Mike <[email protected]>wrote:
> Can someone confirm what IOS licensing I need to use a 2900 as a simple
> SIP-to-POTS gateway for a non-Cisco IP-PBX? Until now all my gateways have
> been for CUCM/MGCP, so I?ve just been buying the SRST bundles.****
>
> ** **
>
> I?m thinking I don?t need an SRST or CME license, right? As long as I have
> UC license I should be good?****
>
> ** **
>
> I know my account team could answer this but it might be best for now if
> they didn?t know we were piloting a non-Cisco system. ;-)****
>
> ** **
>
> -- ****
>
> Mike Norton****
>
> I.T. Specialist****
>
> Peace Wapiti School Division No. 76****
>
> Helpdesk: 780-831-3080****
>
> Direct: 780-831-3076****
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 5
Date: Fri, 7 Dec 2012 13:04:22 -0800 (PST)
From: Paul <[email protected]>
To: Angel Moon <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] FXS over VPN
Message-ID:
<[email protected]>
Content-Type: text/plain; charset=iso-8859-1
Create a voip dial-peer with a session target pointing to the remote 2801 on
the central CCME box and then have a pots dial-peer on the 2801 addressed with
desired DID. I'm assuming you already have a dial-peer to catch all inbound
calls on the PRI. 'debug voice dialpeer inout' for verification/troubleshooting.
________________________________
From: Angel Moon <[email protected]>
To: [email protected]
Sent: Friday, December 7, 2012 6:20 AM
Subject: [cisco-voip] FXS over VPN
Hello All,
I have a scenario with a Central CME and a remote site connected via VPN with a
2801 at the remote site.? PRI at central site.? I would like to get a DID
routed over the VPN to an FXS port on the remote router.? Remote router is not
running telephony-services, just routing.? Is this possible and if so can
someone give me a high level overview??? Thanks in advance!
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip??
------------------------------
Message: 6
Date: Fri, 7 Dec 2012 23:00:39 +0000
From: "Norton, Mike" <[email protected]>
To: Ted Nugent <[email protected]>, Ed Leatherman
<[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] IOS GW licensing for 3rd-party PBX
Message-ID:
<2c4107b204f8e74eb0a63ef59b0f97024aa...@pwsdexchange06.pwsb33.ab.ca>
Content-Type: text/plain; charset="us-ascii"
Okay good, thanks for confirming.
-mn
From: Ted Nugent [mailto:[email protected]]
Sent: December-07-12 2:04 PM
To: Norton, Mike
Cc: [email protected]
Subject: Re: [cisco-voip] IOS GW licensing for 3rd-party PBX
Right, for a straight up SIP/H323/MGCP gateway you don't need SRST or CME for
that.... UC would be sufficient.
On Fri, Dec 7, 2012 at 2:30 PM, Norton, Mike
<[email protected]<mailto:[email protected]>> wrote:
Can someone confirm what IOS licensing I need to use a 2900 as a simple
SIP-to-POTS gateway for a non-Cisco IP-PBX? Until now all my gateways have been
for CUCM/MGCP, so I've just been buying the SRST bundles.
I'm thinking I don't need an SRST or CME license, right? As long as I have UC
license I should be good?
I know my account team could answer this but it might be best for now if they
didn't know we were piloting a non-Cisco system. ;-)
--
Mike Norton
I.T. Specialist
Peace Wapiti School Division No. 76
Helpdesk: 780-831-3080<tel:780-831-3080>
Direct: 780-831-3076<tel:780-831-3076>
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 7
Date: Fri, 7 Dec 2012 21:54:05 -0600
From: Matthew Saskin <[email protected]>
To: Nick Matthews <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Can a CUBE be used to connect CUCM and
Asterisk to a SIP Trunk
Message-ID:
<CAMSV-msP3j973xzssgnO_uHNPStYNvC2myUV2sDa6CQ=rs1...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Direct CUBE to CUCM is easy. The following is a snippet from my asterisk
lab sip.conf - very simple, no authentication happening, but sets up
trunking to both nodes in the lab cluster and dumps inbound calls to a
common context for routing. On the CUCM side, it's nothing but a pretty
standard SIP trunk config with nothing changed from the defaults.
[cucm9pub]
type=friend
context=from-callmanager
host=192.168.11.50
disallow=all
allow=ulaw
nat=no
qualify=yes
[cucm9sub]
type=friend
context=from-callmanager
host=192.168.11.51
disallow=all
allow=ulaw
nat=no
qualify=yes
Matthew Saskin
[email protected]
203-253-9571
On Sun, Dec 2, 2012 at 11:39 AM, Nick Matthews <[email protected]> wrote:
> Realistically you could probably trunk them together without CUBE. CUCM
> has a lot more capability in altering the inbound/outbound SIP settings
> that CUBE was used for previously. CUBE would make it potentially easier
> and still makes the media exchange simpler, but my guess is with Asterisk
> for 50 phones this design aspect isn't significant.
>
> I haven't seen any guides for either CUCM or CUBE connectivity to
> Asterisk, but they're probably out there on the interwebs somewhere.
>
> -nick
>
>
>
> On Sun, Dec 2, 2012 at 10:33 AM, Matthew Saskin <[email protected]> wrote:
>
>> This would not be a problem at all. The short answer is you'd end up
>> having dial peers on the CUBE that route (via destination pattern, etc.)
>> specific calls/DID's to the asterisk IP address and specific calls/DID's to
>> the be6k address.
>>
>> -matthew
>>
>> Matthew Saskin
>> [email protected]
>> 203-253-9571
>>
>>
>>
>> On Sun, Dec 2, 2012 at 12:40 AM, Dieter Jansen
>> <[email protected]>wrote:
>>
>>> Hi All,
>>>
>>> We are looking to replace an existing Asterisk PBX with about 40 SIP
>>> handsets with a CUCM UC solution (BE6K) with about 50 new CP-8945s.
>>>
>>> The current installation uses a SIP trunk and we plan to keep using this
>>> to connect to the PSTN. We have a 100 DID block for our incoming calls.
>>>
>>> As we have comparatively little experience with the Cisco solution we're
>>> looking at a soft introduction and will probably have both sets of handsets
>>> in place in parallel for a while.
>>>
>>> I've got all the edge switches and VLANing working - independent data,
>>> voice and legacy voice VLANs/subnets and all required routing between.
>>> We're at the point where test phones can place video calls to each other
>>> so we now need to get routing and PSTN access happening
>>>
>>> Originally I was thinking I'd set up a CUCM-behind-Asterisk
>>> or Asterisk behind-CUCM situation whilst testing but on reading the Cisco
>>> documents and books I've noted the suggested use of a CUBE SBC. As we have
>>> a spare router that can probably do CUBE I was wondering if I should set
>>> that up to register with our VSP.
>>>
>>> In the first instance I'd get the Asterisk box to pass calls to the CUBE
>>> and have the CUBE forward all incoming calls to the Asterisk.
>>>
>>> What I'm hoping is that I could then also have the CUCM pass calls to
>>> the same CUBE and route a subset of the inward DIDs to the CUCM whilst
>>> testing - allowing me to get all the setup sorted whilst easily flipping
>>> incoming calls to specific DID between Asterisk and CUCM.
>>>
>>> I've not seen an example of the CUBE being used in this way - is it
>>> possible?
>>>
>>> Regards, JustTemprament.
>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> [email protected]
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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Message: 8
Date: Fri, 7 Dec 2012 21:56:42 -0600
From: Matthew Saskin <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] Can CTS-CTRL-DV8 be used as a Phone for end
user?
Message-ID:
<camsv-mveri0kszg2rfg5p--zwresu30i9bnm_qn3jouuv5d...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Negative. The Codec/CTS unit is what actually provides registration, the
CTS-CTRL is just a control device for the codec/CTS unit.
Matthew Saskin
[email protected]
203-253-9571
On Mon, Dec 3, 2012 at 1:41 PM, Jason Aarons (AM) <
[email protected]> wrote:
> CTS-CTRL-DV8
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End of cisco-voip Digest, Vol 110, Issue 8
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