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Today's Topics:
1. Re: Shared Lines and Intercom Line? (Nate VanMaren)
2. Re: Recommended SIP SBC (randal k)
3. Re: Recommended SIP SBC (Gavin Henry)
4. Re: Recommended SIP SBC (randal k)
5. Re: Recommended SIP SBC (Heim, Dennis)
6. 2400 ds0-group (Michael Tague)
7. Unassigned DNs does not include Intercom-DNs ? (Jason Aarons (AM))
8. Re: Unassigned DNs does not include Intercom-DNs ?
(Florian Kroessbacher)
9. Telco SIP trunk with Cisco Voice Gateway and CUCM 9.1
([email protected])
10. Re: Telco SIP trunk with Cisco Voice Gateway and CUCM 9.1
(Ted Nugent)
11. Re: SIP trunk failover question (Eric Pedersen)
----------------------------------------------------------------------
Message: 1
Date: Wed, 15 May 2013 15:34:07 +0000
From: Nate VanMaren <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>, Bill Talley
<[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Excel has had remote duplicates for a few years on the data tab.
From: cisco-voip [mailto:[email protected]] On Behalf Of Jason
Aarons (AM)
Sent: Wednesday, May 15, 2013 8:37 AM
To: Bill Talley
Cc: cisco-voip ([email protected])
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?
Yeah I got 5,000 phone in a excel spreadsheet. I'll have to see how I can
programmatically search for duplicates (this might be a case for using MS
Access to de-duplicate)
1 to many would be a nice feature (eg Intercom-DN being shared and use
multicast to send).
From: Bill Talley [mailto:[email protected]]
Sent: Wednesday, May 15, 2013 10:07 AM
To: Jason Aarons (AM)
Cc: cisco-voip ([email protected]<mailto:[email protected]>)
Subject: Re: [cisco-voip] Shared Lines and Intercom Line?
Sorry, clarification, not EXACTLY what you want to do... The intercom
extension on the other phones with the shared lines will have to be different
than the primary extension.
Sent from an Apple iOS device with very tiny touchscreen input keys. Please
excude my typtos.
On May 15, 2013, at 8:59 AM, "Jason Aarons (AM)"
<[email protected]<mailto:[email protected]>> wrote:
Any brilliant idea on how to handle Intercom feature on CallManager 9.1 when
you have shared lines?
I was going to make Line 1 4 digit number.
I was going to make Line 4 - Intercom DN with same 4 digit number.
If I have a Shared Line between two phones Intercom won't work (right?). Wish
CallManager would just multicast to the two phones (isnt' that how CME works?)
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Message: 2
Date: Wed, 15 May 2013 11:57:12 -0600
From: randal k <[email protected]>
To: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
<canelk7rmroeh4chuo+kyhbhpchp3b_ijznpb3obxyougeoo...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I too have been trying to get a software-only eval of Acme and have been
stonewalled by an unresponsive sales team. Sonus has been OK but a bit
slow. Sansay, on the other hand, has been very responsive and has
aggressive pricing, a good UI and a solid feature set, although they are
small.
We are also in the ~150-200 concurrent session land -- too small for huge,
too big to use open-everything -- and believe that the "virtual SBC"
concept is incredibly hot and we *really* want it ... but ... it's really
hard to get. Everyone wants to sell a box (sansay/sonus), just isn't
selling anything (acme), or has targeted virtual SBC at huge networks
instead of small (metaswitch).
Let me know how it goes for you!
On Tue, May 14, 2013 at 10:33 AM, Robert Kulagowski <[email protected]>wrote:
> [Sorry for breaking the thread, but original message is gone.]
>
> Other than Acme Packet, has anyone else deployed something like a
> Sonus? The merger with Oracle seems to be slowing things down over
> there, and I may not be able to get equipment from the Demo Depot
> until after they're done merging, and in the 3 weeks since our meeting
> the AP sales guy still hasn't gotten me a copy of the "software only"
> SBC to even start an eval even though I keep pestering him.
>
> Something that can do 200 simultaneous sessions is what we're looking
> for, G.711, no fancy transcoding required.
>
> I found a SBC comparison report, but it's $3400, so a bit expensive,
> but the executive summary put Sonus as their choice.
>
> If you have experience with multiple SBCs and can go into why one and
> not another then that's an added bonus.
>
> Thanks.
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
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Message: 3
Date: Wed, 15 May 2013 19:11:35 +0100
From: Gavin Henry <[email protected]>
To: randal k <[email protected]>
Cc: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
<CAPcb_GL=TWTJRkVy+iLRnNbHKv_0Hqz0cgKEMEbcO=kmcpq...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Hi Randal,
If you don't need transcoding I'd recommend a UM Labs EC-4200 SIP
Security Controller for Enterprises. We just got one in. Very nice and
also allows you to connect to a Lync system.
See http://www.um-labs.com/docs/EC-4200.pdf
It's not an SBC but much more.
Thanks.
On 15 May 2013 18:57, randal k <[email protected]> wrote:
> I too have been trying to get a software-only eval of Acme and have been
> stonewalled by an unresponsive sales team. Sonus has been OK but a bit slow.
> Sansay, on the other hand, has been very responsive and has aggressive
> pricing, a good UI and a solid feature set, although they are small.
>
> We are also in the ~150-200 concurrent session land -- too small for huge,
> too big to use open-everything -- and believe that the "virtual SBC" concept
> is incredibly hot and we *really* want it ... but ... it's really hard to
> get. Everyone wants to sell a box (sansay/sonus), just isn't selling
> anything (acme), or has targeted virtual SBC at huge networks instead of
> small (metaswitch).
>
> Let me know how it goes for you!
>
>
>
> On Tue, May 14, 2013 at 10:33 AM, Robert Kulagowski <[email protected]>
> wrote:
>>
>> [Sorry for breaking the thread, but original message is gone.]
>>
>> Other than Acme Packet, has anyone else deployed something like a
>> Sonus? The merger with Oracle seems to be slowing things down over
>> there, and I may not be able to get equipment from the Demo Depot
>> until after they're done merging, and in the 3 weeks since our meeting
>> the AP sales guy still hasn't gotten me a copy of the "software only"
>> SBC to even start an eval even though I keep pestering him.
>>
>> Something that can do 200 simultaneous sessions is what we're looking
>> for, G.711, no fancy transcoding required.
>>
>> I found a SBC comparison report, but it's $3400, so a bit expensive,
>> but the executive summary put Sonus as their choice.
>>
>> If you have experience with multiple SBCs and can go into why one and
>> not another then that's an added bonus.
>>
>> Thanks.
>> _______________________________________________
>> 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
>
--
Kind Regards,
Gavin Henry.
Managing Director.
T +44 (0) 1224 279484
M +44 (0) 7930 323266
F +44 (0) 1224 824887
E [email protected]
Open Source. Open Solutions(tm).
http://www.suretecsystems.com/
Suretec Systems is a limited company registered in Scotland. Registered
number: SC258005. Registered office: 24 Cormack Park, Rothienorman, Inverurie,
Aberdeenshire, AB51 8GL.
Subject to disclaimer at http://www.suretecgroup.com/disclaimer.html
Do you know we have our own VoIP provider called SureVoIP? See
http://www.surevoip.co.uk
Did you see our API? http://www.surevoip.co.uk/api
------------------------------
Message: 4
Date: Wed, 15 May 2013 12:29:44 -0600
From: randal k <[email protected]>
To: "Heim, Dennis" <[email protected]>
Cc: Cisco VOIP <[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
<CANeLk7RNoU49USHLYKcBJqJJYb2=xbwhaolg_jztbjspqao...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
I can only speak to the Sansay so far, but it has an excellent SOAP XML API
that is very full featured.
Randal
On Wed, May 15, 2013 at 12:28 PM, Heim, Dennis <[email protected]> wrote:
> DO any of the SBC?s out there have a good API?****
>
> ** **
>
> *Dennis Heim | **Sr. Unified Collaboration Team Lead & Solution Architect
> *****
>
> World Wide Technology | 314.212.1814 | [email protected]**
>
> ?Creating Impact, Ignition & Scalability?****
>
> ** **
>
> *From:* cisco-voip [mailto:[email protected]] *On Behalf
> Of *randal k
> *Sent:* Wednesday, May 15, 2013 1:57 PM
> *To:* Cisco VOIP
> *Subject:* Re: [cisco-voip] Recommended SIP SBC****
>
> ** **
>
> I too have been trying to get a software-only eval of Acme and have been
> stonewalled by an unresponsive sales team. Sonus has been OK but a bit
> slow. Sansay, on the other hand, has been very responsive and has
> aggressive pricing, a good UI and a solid feature set, although they are
> small.
>
> We are also in the ~150-200 concurrent session land -- too small for huge,
> too big to use open-everything -- and believe that the "virtual SBC"
> concept is incredibly hot and we *really* want it ... but ... it's really
> hard to get. Everyone wants to sell a box (sansay/sonus), just isn't
> selling anything (acme), or has targeted virtual SBC at huge networks
> instead of small (metaswitch).****
>
> Let me know how it goes for you!****
>
> ** **
>
> ** **
>
> On Tue, May 14, 2013 at 10:33 AM, Robert Kulagowski <[email protected]>
> wrote:****
>
> [Sorry for breaking the thread, but original message is gone.]
>
> Other than Acme Packet, has anyone else deployed something like a
> Sonus? The merger with Oracle seems to be slowing things down over
> there, and I may not be able to get equipment from the Demo Depot
> until after they're done merging, and in the 3 weeks since our meeting
> the AP sales guy still hasn't gotten me a copy of the "software only"
> SBC to even start an eval even though I keep pestering him.
>
> Something that can do 200 simultaneous sessions is what we're looking
> for, G.711, no fancy transcoding required.
>
> I found a SBC comparison report, but it's $3400, so a bit expensive,
> but the executive summary put Sonus as their choice.
>
> If you have experience with multiple SBCs and can go into why one and
> not another then that's an added bonus.
>
> Thanks.
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
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Message: 5
Date: Wed, 15 May 2013 13:28:27 -0500
From: "Heim, Dennis" <[email protected]>
To: randal k <[email protected]>, Cisco VOIP
<[email protected]>
Subject: Re: [cisco-voip] Recommended SIP SBC
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
DO any of the SBC's out there have a good API?
Dennis Heim | Sr. Unified Collaboration Team Lead & Solution Architect
World Wide Technology | 314.212.1814 |
[email protected]<mailto:[email protected]>
"Creating Impact, Ignition & Scalability"
From: cisco-voip [mailto:[email protected]] On Behalf Of
randal k
Sent: Wednesday, May 15, 2013 1:57 PM
To: Cisco VOIP
Subject: Re: [cisco-voip] Recommended SIP SBC
I too have been trying to get a software-only eval of Acme and have been
stonewalled by an unresponsive sales team. Sonus has been OK but a bit slow.
Sansay, on the other hand, has been very responsive and has aggressive pricing,
a good UI and a solid feature set, although they are small.
We are also in the ~150-200 concurrent session land -- too small for huge, too
big to use open-everything -- and believe that the "virtual SBC" concept is
incredibly hot and we *really* want it ... but ... it's really hard to get.
Everyone wants to sell a box (sansay/sonus), just isn't selling anything
(acme), or has targeted virtual SBC at huge networks instead of small
(metaswitch).
Let me know how it goes for you!
On Tue, May 14, 2013 at 10:33 AM, Robert Kulagowski
<[email protected]<mailto:[email protected]>> wrote:
[Sorry for breaking the thread, but original message is gone.]
Other than Acme Packet, has anyone else deployed something like a
Sonus? The merger with Oracle seems to be slowing things down over
there, and I may not be able to get equipment from the Demo Depot
until after they're done merging, and in the 3 weeks since our meeting
the AP sales guy still hasn't gotten me a copy of the "software only"
SBC to even start an eval even though I keep pestering him.
Something that can do 200 simultaneous sessions is what we're looking
for, G.711, no fancy transcoding required.
I found a SBC comparison report, but it's $3400, so a bit expensive,
but the executive summary put Sonus as their choice.
If you have experience with multiple SBCs and can go into why one and
not another then that's an added bonus.
Thanks.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 6
Date: Wed, 15 May 2013 16:38:17 -0400
From: "Michael Tague" <[email protected]>
To: "'cisco-voip'" <[email protected]>
Subject: [cisco-voip] 2400 ds0-group
Message-ID: <059901ce51ac$216e61f0$644b25d0$@net>
Content-Type: text/plain; charset="us-ascii"
Hi,
I am looking for help in configuring a ds0-group for a Cisco 2400. We want
the 2400 to generate a channelized T1 that will go to a PBX. The PBX is
currently connected to another carrier and appears to be configured to use
E&M Wink-Start. On the back-end the 2400 would receive calls via a PRI;
basically we want a PRI to T1 E&M Wink-Start converter!
We use the 2400 as a Swiss-Army knife of PRI/SIP/IP to PRI/POTS/IP, works
well but we've never tried to originate a channelized T1 with it (only
PRIs).
As a test bed, I am using a second (Test) 2400 with 8-FXS ports, one of
which is hooked to an analog phone. My Test 2400 is connected to the other
2400 (let's call it the CPE 2400) via a cross-over T1 cable.
My problem is double because I need to know how to configure a Test 2400 for
testing (so I can simulate the customer's PBX) essentially making the 2400 a
channel bank and I need to know how to get the other 2400 (let's call it the
CPE 2400) to signal ringing, provide dial-tone (collect digits), and ideally
deliver the caller-id number and name via DTMF.
Help!
Thanks,
Michael
[email protected]
502-815-7171
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Message: 7
Date: Wed, 15 May 2013 20:25:21 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] Unassigned DNs does not include Intercom-DNs ?
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d332a0...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
A search in Route Plan Report > Unassigned DNs does not find Unassigned
Intercom DNs ? Any workarounds ?
CallManager 9.1.1
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Message: 8
Date: Wed, 15 May 2013 21:16:55 -0700 (PDT)
From: "Florian Kroessbacher" <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] Unassigned DNs does not include Intercom-DNs
?
Message-ID: <1368677814038.8f67b416@Nodemailer>
Content-Type: text/plain; charset="utf-8"
hy,
what about run sql throught the numlan table
--
Florian Kroessbacher
gmail: [email protected]
On Thu, May 16, 2013 at 2:27 AM, Jason Aarons (AM)
<[email protected]> wrote:
> A search in Route Plan Report > Unassigned DNs does not find Unassigned
> Intercom DNs ? Any workarounds ?
> CallManager 9.1.1
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Message: 9
Date: Thu, 16 May 2013 16:06:25 +0200 (CEST)
From: "[email protected]" <[email protected]>
To: [email protected]
Subject: [cisco-voip] Telco SIP trunk with Cisco Voice Gateway and
CUCM 9.1
Message-ID: <[email protected]>
Content-Type: text/plain; charset="utf-8"
Hi guys I'll need to use a voice gateway Cisco to connect to SIP trunk on telco
provider.
The topology will be this one:
Telco -----SIP Trunk----> Cisco 2901 SRST -----> H.323 -------> CUCM 9.1
The question is, I'll need to purchase the SBC license? Or other license
features?
Thank you all.
Regards
Alessandro Bertacco
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Message: 10
Date: Thu, 16 May 2013 10:27:15 -0400
From: Ted Nugent <[email protected]>
To: "[email protected]" <[email protected]>
Cc: Cisco VoIPoE List <[email protected]>
Subject: Re: [cisco-voip] Telco SIP trunk with Cisco Voice Gateway and
CUCM 9.1
Message-ID:
<CAHs2VYvPb_LUMikCJw_OqOUSWNWO9VP46_6My3fZM_u=gf0...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Yes, you'll need a CUBE license, see link
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/order_guide_c07_462222.html
On Thu, May 16, 2013 at 10:06 AM, [email protected] <
[email protected]> wrote:
> Hi guys I'll need to use a voice gateway Cisco to connect to SIP trunk on
> telco provider.
>
> The topology will be this one:
>
> Telco -----SIP Trunk----> Cisco 2901 SRST -----> H.323 -------> CUCM 9.1
>
> The question is, I'll need to purchase the SBC license? Or other license
> features?
> Thank you all.
>
> Regards
>
> Alessandro Bertacco
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 11
Date: Thu, 16 May 2013 14:51:23 +0000
From: Eric Pedersen <[email protected]>
To: Erick Wellnitz <[email protected]>, Robert Kulagowski
<[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] SIP trunk failover question
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
If you're using TCP as the SIP transport, I believe CUCM will take the trunk
out of service if the TCP connection drops.
From: cisco-voip [mailto:[email protected]] On Behalf Of Erick
Wellnitz
Sent: 14 May 2013 11:00 AM
To: Robert Kulagowski
Cc: cisco-voip
Subject: Re: [cisco-voip] SIP trunk failover question
Looks like this isn't going to happen until our upgrade. It's waited this long
already...
On Tue, May 14, 2013 at 9:54 AM, Erick Wellnitz
<[email protected]<mailto:[email protected]>> wrote:
Great! That's exactly what I was looking for but didn't know what it was
called.
On Tue, May 14, 2013 at 9:46 AM, Robert Kulagowski
<[email protected]<mailto:[email protected]>> wrote:
On Tue, May 14, 2013 at 9:27 AM, Erick Wellnitz
<[email protected]<mailto:[email protected]>> wrote:
> I use top down distribution. Are there any other settings I need to make
> sure that the calls don't fail when the SIP trunk is unavailable for any
> reason?
Depending on the version of CM that you're using, you may want to
check "SIP PING".
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