Mailing List Archive: Cisco: VOIP

CUCM SIP To PSTN

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george.hendrix at l-3com

Jun 14, 2012, 5:44 AM

Post #1 of 12 (1372 views)
Permalink

CUCM SIP To PSTN

Hi everyone,

I know with h.323 gateways in CUCM, you have to configure dial peers on each gateway and for each CUCM. Is it possible to create a SIP trunk directly from CUCM to the Telco? Without having to do any dial peers on the gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?

Thanks,

Bill Hendrix

jorge.rodriguez at netxar

Jun 14, 2012, 6:53 AM

Post #2 of 12 (1342 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

You can, but unless, you want to open your entire network to the provider, you should not as the Signalling will be SIP from the CUCM to the Media GW on the CUCM but the RTP will be from your phones directly to the Telco Media GW.

Jorge Rodriguez Aguila, CCVP-Voice
Senior Voice/Data Consultant
Netxar Technologies
jorge.rodriguez [at] netxar<mailto:jorge.rodriguez [at] netxar>
787-688-8530

From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of george.hendrix [at] l-3com
Sent: Thursday, June 14, 2012 8:44 AM
To: cisco-voip [at] puck
Subject: [cisco-voip] CUCM SIP To PSTN

Hi everyone,

I know with h.323 gateways in CUCM, you have to configure dial peers on each gateway and for each CUCM. Is it possible to create a SIP trunk directly from CUCM to the Telco? Without having to do any dial peers on the gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?

Thanks,

Bill Hendrix

adel.abushaev at gmail

Jun 14, 2012, 10:17 AM

Post #3 of 12 (1335 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

You can set up a SIP trunk to your SP, assuming that you are a SIP
client of them.Otherwise, if you want to go over T1, then you need to
terminate SIP on the GW to translate between SIP and ISDN PRI or
whatever other signalling you are using between you and telco.

A.

On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
> Hi everyone,
>
>
>
> I know with h.323 gateways in CUCM, you have to configure dial peers on
> each gateway and for each CUCM. Is it possible to create a SIP trunk
> directly from CUCM to the Telco? Without having to do any dial peers on the
> gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on
> the SIP GW for both CUCM and the PSTN?
>
>
>
> Thanks,
>
>
>
> Bill Hendrix
>
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

matthnick at gmail

Jun 14, 2012, 7:16 PM

Post #4 of 12 (1333 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

The scenario of:
CUCM--SIP--GW--SIP--Provider

The gateway magically turns into a CUBE, and there's a whole bunch of
marketing and technical information on what happens when you do that.

If you compare these two solutions:
-CUCM direct SIP trunk to provider
-Use MTP to keep media to a single IP
-Or use NAT to keep internal addressing safe
or
-Use CUBE

There's a whole bunch of scalability and troubleshooting problems that
can arise from the first. Having a demarcation point at the GW (CUBE)
is extremely helpful. As well, it prevents you from needing to NAT SIP
which historically is a pretty terrible idea. It also has some SIP
security and flexibility options, and is a good centralization point
for trunks.

I haven't worked with anyone doing the direct trunk from CUCM to
provider. Many providers are going to make their own rules which will
include an SBC (industry term for CUBE).

Short story - just use an SBC. There's about 3-4 compelling reasons.

-nick

On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
> You can set up a SIP trunk to your SP, assuming that you are a SIP
> client of them.Otherwise, if you want to go over T1, then you need to
> terminate SIP on the GW to translate between SIP and ISDN PRI or
> whatever other signalling you are using between you and telco.
>
> A.
>
> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>> Hi everyone,
>>
>>
>>
>> I know with h.323 gateways in CUCM, you have to configure dial peers on
>> each gateway and for each CUCM. Is it possible to create a SIP trunk
>> directly from CUCM to the Telco? Without having to do any dial peers on the
>> gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on
>> the SIP GW for both CUCM and the PSTN?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Bill Hendrix
>>
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

lelio at uoguelph

Jun 14, 2012, 9:39 PM

Post #5 of 12 (1329 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

The question I've had is, what is the best practice for this? Should you use a separate router for the CUBE functionality? Or use an existing gateway? I'm sure the answer is 'it depends' ;), but if there are some scenarios where it makes sense not to, that would be good to know.

Sent from my iPhone...

"There's no place like 127.0.0.1"

On Jun 14, 2012, at 7:17 PM, Nick Matthews <matthnick [at] gmail> wrote:

> The scenario of:
> CUCM--SIP--GW--SIP--Provider
>
> The gateway magically turns into a CUBE, and there's a whole bunch of
> marketing and technical information on what happens when you do that.
>
> If you compare these two solutions:
> -CUCM direct SIP trunk to provider
> -Use MTP to keep media to a single IP
> -Or use NAT to keep internal addressing safe
> or
> -Use CUBE
>
> There's a whole bunch of scalability and troubleshooting problems that
> can arise from the first. Having a demarcation point at the GW (CUBE)
> is extremely helpful. As well, it prevents you from needing to NAT SIP
> which historically is a pretty terrible idea. It also has some SIP
> security and flexibility options, and is a good centralization point
> for trunks.
>
> I haven't worked with anyone doing the direct trunk from CUCM to
> provider. Many providers are going to make their own rules which will
> include an SBC (industry term for CUBE).
>
> Short story - just use an SBC. There's about 3-4 compelling reasons.
>
> -nick
>
> On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
>> You can set up a SIP trunk to your SP, assuming that you are a SIP
>> client of them.Otherwise, if you want to go over T1, then you need to
>> terminate SIP on the GW to translate between SIP and ISDN PRI or
>> whatever other signalling you are using between you and telco.
>>
>> A.
>>
>> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>>> Hi everyone,
>>>
>>>
>>>
>>> I know with h.323 gateways in CUCM, you have to configure dial peers on
>>> each gateway and for each CUCM. Is it possible to create a SIP trunk
>>> directly from CUCM to the Telco? Without having to do any dial peers on the
>>> gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on
>>> the SIP GW for both CUCM and the PSTN?
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Bill Hendrix
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

asobihoudai at yahoo

Jun 14, 2012, 10:01 PM

Post #6 of 12 (1340 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

I think primary factors for this would be the BHCA call volume (i.e., the maximum number of simultaneous SIP sessions expected), the maximum number of simultaneous SIP sessions required for the gateway to support (if it even can cover the customer requirements), and of course cost since those CUBE licenses are PRICEY!

E.g., if you're going to max out a 3945E with 2500 simultaneous SIP sessions, it's probably not a wise idea to use it for much else...

----- Original Message -----
From: Lelio Fulgenzi <lelio [at] uoguelph>
To: Nick Matthews <matthnick [at] gmail>
Cc: "cisco-voip [at] puck" <cisco-voip [at] puck>
Sent: Friday, June 15, 2012 12:39 AM
Subject: Re: [cisco-voip] CUCM SIP To PSTN

The question I've had is, what is the best practice for this? Should you use a separate router for the CUBE functionality? Or use an existing gateway? I'm sure the answer is 'it depends' ;), but if there are some scenarios where it makes sense not to, that would be good to know.

Sent from my iPhone...

"There's no place like 127.0.0.1"

On Jun 14, 2012, at 7:17 PM, Nick Matthews <matthnick [at] gmail> wrote:

> The scenario of:
> CUCM--SIP--GW--SIP--Provider
>
> The gateway magically turns into a CUBE, and there's a whole bunch of
> marketing and technical information on what happens when you do that.
>
> If you compare these two solutions:
> -CUCM direct SIP trunk to provider
> -Use MTP to keep media to a single IP
> -Or use NAT to keep internal addressing safe
> or
> -Use CUBE
>
> There's a whole bunch of scalability and troubleshooting problems that
> can arise from the first. Having a demarcation point at the GW (CUBE)
> is extremely helpful. As well, it prevents you from needing to NAT SIP
> which historically is a pretty terrible idea. It also has some SIP
> security and flexibility options, and is a good centralization point
> for trunks.
>
> I haven't worked with anyone doing the direct trunk from CUCM to
> provider. Many providers are going to make their own rules which will
> include an SBC (industry term for CUBE).
>
> Short story - just use an SBC. There's about 3-4 compelling reasons.
>
> -nick
>
> On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
>> You can set up a SIP trunk to your SP, assuming that you are a SIP
>> client of them.Otherwise, if you want to go over T1, then you need to
>> terminate SIP on the GW to translate between SIP and ISDN PRI or
>> whatever other signalling you are using between you and telco.
>>
>> A.
>>
>> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>>> Hi everyone,
>>>
>>>
>>>
>>> I know with h.323 gateways in CUCM, you have to configure dial peers on
>>> each gateway and for each CUCM. Is it possible to create a SIP trunk
>>> directly from CUCM to the Telco? Without having to do any dial peers on the
>>> gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on
>>> the SIP GW for both CUCM and the PSTN?
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Bill Hendrix
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

matthnick at gmail

Jun 15, 2012, 5:29 AM

Post #7 of 12 (1334 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

Technically you could have a router terminating DMVPN connections, a
full BGP routing table, running UCS express on a SRE with VMware,
doing IOS firewall, and doing CUBE at the same time.

For most folks it's about task isolation, or sometimes groups are
silo'd and want their own equipment. As long as the CPU/memory are
fine it's not a problem. Sometimes calculating the estimated CPU is an
art. But it is very common to turn an existing gateway into CUBE, and
even keep some TDM on it.

-nick

On Fri, Jun 15, 2012 at 1:01 AM, Paul <asobihoudai [at] yahoo> wrote:
> I think primary factors for this would be the BHCA call volume (i.e., the maximum number of simultaneous SIP sessions expected), the maximum number of simultaneous SIP sessions required for the gateway to support (if it even can cover the customer requirements), and of course cost since those CUBE licenses are PRICEY!
>
> E.g., if you're going to max out a 3945E with 2500 simultaneous SIP sessions, it's probably not a wise idea to use it for much else...
>
>
> ----- Original Message -----
> From: Lelio Fulgenzi <lelio [at] uoguelph>
> To: Nick Matthews <matthnick [at] gmail>
> Cc: "cisco-voip [at] puck" <cisco-voip [at] puck>
> Sent: Friday, June 15, 2012 12:39 AM
> Subject: Re: [cisco-voip] CUCM SIP To PSTN
>
> The question I've had is, what is the best practice for this? Should you use a separate router for the CUBE functionality? Or use an existing gateway? I'm sure the answer is 'it depends' ;), but if there are some scenarios where it makes sense not to, that would be good to know.
>
> Sent from my iPhone...
>
> "There's no place like 127.0.0.1"
>
> On Jun 14, 2012, at 7:17 PM, Nick Matthews <matthnick [at] gmail> wrote:
>
>> The scenario of:
>> CUCM--SIP--GW--SIP--Provider
>>
>> The gateway magically turns into a CUBE, and there's a whole bunch of
>> marketing and technical information on what happens when you do that.
>>
>> If you compare these two solutions:
>> -CUCM direct SIP trunk to provider
>> -Use MTP to keep media to a single IP
>> -Or use NAT to keep internal addressing safe
>> or
>> -Use CUBE
>>
>> There's a whole bunch of scalability and troubleshooting problems that
>> can arise from the first. Having a demarcation point at the GW (CUBE)
>> is extremely helpful. As well, it prevents you from needing to NAT SIP
>> which historically is a pretty terrible idea. It also has some SIP
>> security and flexibility options, and is a good centralization point
>> for trunks.
>>
>> I haven't worked with anyone doing the direct trunk from CUCM to
>> provider. Many providers are going to make their own rules which will
>> include an SBC (industry term for CUBE).
>>
>> Short story - just use an SBC. There's about 3-4 compelling reasons.
>>
>> -nick
>>
>> On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
>>> You can set up a SIP trunk to your SP, assuming that you are a SIP
>>> client of them.Otherwise, if you want to go over T1, then you need to
>>> terminate SIP on the GW to translate between SIP and ISDN PRI or
>>> whatever other signalling you are using between you and telco.
>>>
>>> A.
>>>
>>> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>>>> Hi everyone,
>>>>
>>>>
>>>>
>>>> I know with h.323 gateways in CUCM, you have to configure dial peers on
>>>> each gateway and for each CUCM. Is it possible to create a SIP trunk
>>>> directly from CUCM to the Telco? Without having to do any dial peers on the
>>>> gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on
>>>> the SIP GW for both CUCM and the PSTN?
>>>>
>>>>
>>>>
>>>> Thanks,
>>>>
>>>>
>>>>
>>>> Bill Hendrix
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip [at] puck
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

george.hendrix at l-3com

Jun 15, 2012, 7:46 AM

Post #8 of 12 (1325 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

Basically, I currently have 2 gateways (2x 3925) and each have 4 PRIs (total of 8). They are located on site with CUCM, so they are configured as MGCP gateways. However, we are standing up a lot of other sites. So I am thinking about switching these circuits to 2 10mb SIP circuits. I would have increased call capacity and I could also work with the Telco to re-route calls destined to remote sites if their SIP circuit goes down to these circuits at the main office. That's just something not offered with old PRI circuits, or at least no provider I've worked with has offered that with old PRI. Then if I go with SIP, what's the best configuration in CUCM? I haven't worked much with SIP circuits, but the ones I have worked with, the gateway was just an h.323 GW in CUCM. Could it be added as a different type that would register with CUCM or is h.323 the best way to go?

Appreciate any inputs...

Regards,

Bill Hendrix

-----Original Message-----
From: matthn [at] gmail [mailto:matthn [at] gmail] On Behalf Of Nick Matthews
Sent: Thursday, June 14, 2012 10:17 PM
To: Adel Abushaev
Cc: Hendrix, George (Bill) @ LSG - STRATIS; cisco-voip [at] puck
Subject: Re: [cisco-voip] CUCM SIP To PSTN

The scenario of:
CUCM--SIP--GW--SIP--Provider

The gateway magically turns into a CUBE, and there's a whole bunch of
marketing and technical information on what happens when you do that.

If you compare these two solutions:
-CUCM direct SIP trunk to provider
-Use MTP to keep media to a single IP
-Or use NAT to keep internal addressing safe
or
-Use CUBE

There's a whole bunch of scalability and troubleshooting problems that
can arise from the first. Having a demarcation point at the GW (CUBE)
is extremely helpful. As well, it prevents you from needing to NAT SIP
which historically is a pretty terrible idea. It also has some SIP
security and flexibility options, and is a good centralization point
for trunks.

I haven't worked with anyone doing the direct trunk from CUCM to
provider. Many providers are going to make their own rules which will
include an SBC (industry term for CUBE).

Short story - just use an SBC. There's about 3-4 compelling reasons.

-nick

On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
> You can set up a SIP trunk to your SP, assuming that you are a SIP
> client of them.Otherwise, if you want to go over T1, then you need to
> terminate SIP on the GW to translate between SIP and ISDN PRI or
> whatever other signalling you are using between you and telco.
>
> A.
>
> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>> Hi everyone,
>>
>>
>>
>> I know with h.323 gateways in CUCM, you have to configure dial peers on
>> each gateway and for each CUCM. Is it possible to create a SIP trunk
>> directly from CUCM to the Telco? Without having to do any dial peers on the
>> gateway. Or would it CUCM SIP <> SIP GW <> SIP PSTN, with SIP dial peers on
>> the SIP GW for both CUCM and the PSTN?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Bill Hendrix
>>
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

Dennis.Heim at wwt

Jun 15, 2012, 8:02 AM

Post #9 of 12 (1334 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

So I would go SIP end to end. Plus using cube, keeps most of the complexities of the SIP configuration out of callmanager. I would go with a SIP gateway. What version of callmanager are you using? If you are prior to 8.6, you will need to allocate MTPs if your provides is doing early offer.

Dennis Heim
Sr. UC Engineer
World Wide Technology
Office: 314.212.1814
Email: dennis.heim [at] wwt
www.wwt.com

-----Original Message-----
From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of george.hendrix [at] l-3com
Sent: Friday, June 15, 2012 9:46 AM
To: Nick Matthews; Adel Abushaev
Cc: cisco-voip [at] puck
Subject: Re: [cisco-voip] CUCM SIP To PSTN

Basically, I currently have 2 gateways (2x 3925) and each have 4 PRIs (total of 8). They are located on site with CUCM, so they are configured as MGCP gateways. However, we are standing up a lot of other sites. So I am thinking about switching these circuits to 2 10mb SIP circuits. I would have increased call capacity and I could also work with the Telco to re-route calls destined to remote sites if their SIP circuit goes down to these circuits at the main office. That's just something not offered with old PRI circuits, or at least no provider I've worked with has offered that with old PRI. Then if I go with SIP, what's the best configuration in CUCM? I haven't worked much with SIP circuits, but the ones I have worked with, the gateway was just an h.323 GW in CUCM. Could it be added as a different type that would register with CUCM or is h.323 the best way to go?

Appreciate any inputs...

Regards,

Bill Hendrix

-----Original Message-----
From: matthn [at] gmail [mailto:matthn [at] gmail] On Behalf Of Nick Matthews
Sent: Thursday, June 14, 2012 10:17 PM
To: Adel Abushaev
Cc: Hendrix, George (Bill) @ LSG - STRATIS; cisco-voip [at] puck
Subject: Re: [cisco-voip] CUCM SIP To PSTN

The scenario of:
CUCM--SIP--GW--SIP--Provider

The gateway magically turns into a CUBE, and there's a whole bunch of marketing and technical information on what happens when you do that.

If you compare these two solutions:
-CUCM direct SIP trunk to provider
-Use MTP to keep media to a single IP
-Or use NAT to keep internal addressing safe or -Use CUBE

There's a whole bunch of scalability and troubleshooting problems that can arise from the first. Having a demarcation point at the GW (CUBE) is extremely helpful. As well, it prevents you from needing to NAT SIP which historically is a pretty terrible idea. It also has some SIP security and flexibility options, and is a good centralization point for trunks.

I haven't worked with anyone doing the direct trunk from CUCM to provider. Many providers are going to make their own rules which will include an SBC (industry term for CUBE).

Short story - just use an SBC. There's about 3-4 compelling reasons.

-nick

On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
> You can set up a SIP trunk to your SP, assuming that you are a SIP
> client of them.Otherwise, if you want to go over T1, then you need to
> terminate SIP on the GW to translate between SIP and ISDN PRI or
> whatever other signalling you are using between you and telco.
>
> A.
>
> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>> Hi everyone,
>>
>>
>>
>> I know with h.323 gateways in CUCM, you have to configure dial
>> peers on each gateway and for each CUCM. Is it possible to create a
>> SIP trunk directly from CUCM to the Telco? Without having to do any
>> dial peers on the gateway. Or would it CUCM SIP <> SIP GW <> SIP
>> PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Bill Hendrix
>>
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

george.hendrix at l-3com

Jun 15, 2012, 8:20 AM

Post #10 of 12 (1334 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

Currently running 8.5. Getting ready to upgrade to 8.6.

Bill Hendrix

-----Original Message-----
From: Heim, Dennis [mailto:Dennis.Heim [at] wwt]
Sent: Friday, June 15, 2012 11:02 AM
To: Hendrix, George (Bill) @ LSG - STRATIS; Nick Matthews; Adel Abushaev
Cc: cisco-voip [at] puck
Subject: RE: [cisco-voip] CUCM SIP To PSTN

So I would go SIP end to end. Plus using cube, keeps most of the complexities of the SIP configuration out of callmanager. I would go with a SIP gateway. What version of callmanager are you using? If you are prior to 8.6, you will need to allocate MTPs if your provides is doing early offer.

Dennis Heim
Sr. UC Engineer
World Wide Technology
Office: 314.212.1814
Email: dennis.heim [at] wwt
www.wwt.com

-----Original Message-----
From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of george.hendrix [at] l-3com
Sent: Friday, June 15, 2012 9:46 AM
To: Nick Matthews; Adel Abushaev
Cc: cisco-voip [at] puck
Subject: Re: [cisco-voip] CUCM SIP To PSTN

Basically, I currently have 2 gateways (2x 3925) and each have 4 PRIs (total of 8). They are located on site with CUCM, so they are configured as MGCP gateways. However, we are standing up a lot of other sites. So I am thinking about switching these circuits to 2 10mb SIP circuits. I would have increased call capacity and I could also work with the Telco to re-route calls destined to remote sites if their SIP circuit goes down to these circuits at the main office. That's just something not offered with old PRI circuits, or at least no provider I've worked with has offered that with old PRI. Then if I go with SIP, what's the best configuration in CUCM? I haven't worked much with SIP circuits, but the ones I have worked with, the gateway was just an h.323 GW in CUCM. Could it be added as a different type that would register with CUCM or is h.323 the best way to go?

Appreciate any inputs...

Regards,

Bill Hendrix

-----Original Message-----
From: matthn [at] gmail [mailto:matthn [at] gmail] On Behalf Of Nick Matthews
Sent: Thursday, June 14, 2012 10:17 PM
To: Adel Abushaev
Cc: Hendrix, George (Bill) @ LSG - STRATIS; cisco-voip [at] puck
Subject: Re: [cisco-voip] CUCM SIP To PSTN

The scenario of:
CUCM--SIP--GW--SIP--Provider

The gateway magically turns into a CUBE, and there's a whole bunch of marketing and technical information on what happens when you do that.

If you compare these two solutions:
-CUCM direct SIP trunk to provider
-Use MTP to keep media to a single IP
-Or use NAT to keep internal addressing safe or -Use CUBE

There's a whole bunch of scalability and troubleshooting problems that can arise from the first. Having a demarcation point at the GW (CUBE) is extremely helpful. As well, it prevents you from needing to NAT SIP which historically is a pretty terrible idea. It also has some SIP security and flexibility options, and is a good centralization point for trunks.

I haven't worked with anyone doing the direct trunk from CUCM to provider. Many providers are going to make their own rules which will include an SBC (industry term for CUBE).

Short story - just use an SBC. There's about 3-4 compelling reasons.

-nick

On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
> You can set up a SIP trunk to your SP, assuming that you are a SIP
> client of them.Otherwise, if you want to go over T1, then you need to
> terminate SIP on the GW to translate between SIP and ISDN PRI or
> whatever other signalling you are using between you and telco.
>
> A.
>
> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>> Hi everyone,
>>
>>
>>
>> I know with h.323 gateways in CUCM, you have to configure dial
>> peers on each gateway and for each CUCM. Is it possible to create a
>> SIP trunk directly from CUCM to the Telco? Without having to do any
>> dial peers on the gateway. Or would it CUCM SIP <> SIP GW <> SIP
>> PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Bill Hendrix
>>
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

matthnick at gmail

Jun 15, 2012, 1:07 PM

Post #11 of 12 (1337 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

Just a note on the CUCM prior to 8.6 MTP thing - this is pretty
commonly stated but isn't what I would recommend.

CUBE does delayed offer to early offer conversion (DO-EO) which will
convert delayed offer into early offer without needing an MTP. That's
how people did it before CUCM 8.6 and while it's a nice feature, it
doesn't necessarily buy you much with SIP trunking since DO-EO
(early-offer forced) doesn't cost you anything or have any real
downsides. If I was on 8.6 or later I would utilize that feature over
DO-EO conversion since it's cleaner, but not by much.

-nick

On Fri, Jun 15, 2012 at 11:20 AM, <george.hendrix [at] l-3com> wrote:
> Currently running 8.5. Getting ready to upgrade to 8.6.
>
> Bill Hendrix
>
>
> -----Original Message-----
> From: Heim, Dennis [mailto:Dennis.Heim [at] wwt]
> Sent: Friday, June 15, 2012 11:02 AM
> To: Hendrix, George (Bill) @ LSG - STRATIS; Nick Matthews; Adel Abushaev
> Cc: cisco-voip [at] puck
> Subject: RE: [cisco-voip] CUCM SIP To PSTN
>
> So I would go SIP end to end. Plus using cube, keeps most of the complexities of the SIP configuration out of callmanager. I would go with a SIP gateway. What version of callmanager are you using? If you are prior to 8.6, you will need to allocate MTPs if your provides is doing early offer.
>
> Dennis Heim
> Sr. UC Engineer
> World Wide Technology
> Office: 314.212.1814
> Email: dennis.heim [at] wwt
> www.wwt.com
>
> -----Original Message-----
> From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of george.hendrix [at] l-3com
> Sent: Friday, June 15, 2012 9:46 AM
> To: Nick Matthews; Adel Abushaev
> Cc: cisco-voip [at] puck
> Subject: Re: [cisco-voip] CUCM SIP To PSTN
>
>
>
> Basically, I currently have 2 gateways (2x 3925) and each have 4 PRIs (total of 8). They are located on site with CUCM, so they are configured as MGCP gateways. However, we are standing up a lot of other sites. So I am thinking about switching these circuits to 2 10mb SIP circuits. I would have increased call capacity and I could also work with the Telco to re-route calls destined to remote sites if their SIP circuit goes down to these circuits at the main office. That's just something not offered with old PRI circuits, or at least no provider I've worked with has offered that with old PRI. Then if I go with SIP, what's the best configuration in CUCM? I haven't worked much with SIP circuits, but the ones I have worked with, the gateway was just an h.323 GW in CUCM. Could it be added as a different type that would register with CUCM or is h.323 the best way to go?
>
> Appreciate any inputs...
>
> Regards,
>
> Bill Hendrix
>
>
> -----Original Message-----
> From: matthn [at] gmail [mailto:matthn [at] gmail] On Behalf Of Nick Matthews
> Sent: Thursday, June 14, 2012 10:17 PM
> To: Adel Abushaev
> Cc: Hendrix, George (Bill) @ LSG - STRATIS; cisco-voip [at] puck
> Subject: Re: [cisco-voip] CUCM SIP To PSTN
>
> The scenario of:
> CUCM--SIP--GW--SIP--Provider
>
> The gateway magically turns into a CUBE, and there's a whole bunch of marketing and technical information on what happens when you do that.
>
> If you compare these two solutions:
> -CUCM direct SIP trunk to provider
> -Use MTP to keep media to a single IP
> -Or use NAT to keep internal addressing safe or -Use CUBE
>
> There's a whole bunch of scalability and troubleshooting problems that can arise from the first. Having a demarcation point at the GW (CUBE) is extremely helpful. As well, it prevents you from needing to NAT SIP which historically is a pretty terrible idea. It also has some SIP security and flexibility options, and is a good centralization point for trunks.
>
> I haven't worked with anyone doing the direct trunk from CUCM to provider. Many providers are going to make their own rules which will include an SBC (industry term for CUBE).
>
> Short story - just use an SBC. There's about 3-4 compelling reasons.
>
> -nick
>
> On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail> wrote:
>> You can set up a SIP trunk to your SP, assuming that you are a SIP
>> client of them.Otherwise, if you want to go over T1, then you need to
>> terminate SIP on the GW to translate between SIP and ISDN PRI or
>> whatever other signalling you are using between you and telco.
>>
>> A.
>>
>> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com> wrote:
>>> Hi everyone,
>>>
>>>
>>>
>>> I know with h.323 gateways in CUCM, you have to configure dial
>>> peers on each gateway and for each CUCM. Is it possible to create a
>>> SIP trunk directly from CUCM to the Telco? Without having to do any
>>> dial peers on the gateway. Or would it CUCM SIP <> SIP GW <> SIP
>>> PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Bill Hendrix
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

bob.zanett at dimensiondata

Jun 20, 2012, 6:14 AM

Post #12 of 12 (1211 views)
Permalink

Re: CUCM SIP To PSTN [In reply to]

Cube has the ability to convert from delayed to early offer. Thus, you can set CUCM up for delayed offer (outbound) -thus no MTP is required. Cube then converts to early-offer before sending to provider. With Cube, since media is passing through (assuming this since you are hiding your network), the IP/port is somewhat already locked down.

Kind Regards -

Bob
Dimension Data Americas

www.dimensiondata.com

From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of george.hendrix [at] l-3com
Sent: Friday, June 15, 2012 10:20 AM
To: Heim, Dennis; Nick Matthews; Adel Abushaev
Cc: cisco-voip [at] puck
Subject: Re: [cisco-voip] CUCM SIP To PSTN

Currently running 8.5. Getting ready to upgrade to 8.6.

Bill Hendrix

-----Original Message-----
From: Heim, Dennis [mailto:Dennis.Heim [at] wwt]<mailto:[mailto:Dennis.Heim [at] wwt]>
Sent: Friday, June 15, 2012 11:02 AM
To: Hendrix, George (Bill) @ LSG - STRATIS; Nick Matthews; Adel Abushaev
Cc: cisco-voip [at] puck<mailto:cisco-voip [at] puck>
Subject: RE: [cisco-voip] CUCM SIP To PSTN

So I would go SIP end to end. Plus using cube, keeps most of the complexities of the SIP configuration out of callmanager. I would go with a SIP gateway. What version of callmanager are you using? If you are prior to 8.6, you will need to allocate MTPs if your provides is doing early offer.

Dennis Heim
Sr. UC Engineer
World Wide Technology
Office: 314.212.1814
Email: dennis.heim [at] wwt<mailto:dennis.heim [at] wwt>
www.wwt.com<http://www.wwt.com>

-----Original Message-----
From: cisco-voip-bounces [at] puck<mailto:cisco-voip-bounces [at] puck> [mailto:cisco-voip-bounces [at] puck]<mailto:[mailto:cisco-voip-bounces [at] puck]> On Behalf Of george.hendrix [at] l-3com<mailto:george.hendrix [at] l-3com>
Sent: Friday, June 15, 2012 9:46 AM
To: Nick Matthews; Adel Abushaev
Cc: cisco-voip [at] puck<mailto:cisco-voip [at] puck>
Subject: Re: [cisco-voip] CUCM SIP To PSTN

Basically, I currently have 2 gateways (2x 3925) and each have 4 PRIs (total of 8). They are located on site with CUCM, so they are configured as MGCP gateways. However, we are standing up a lot of other sites. So I am thinking about switching these circuits to 2 10mb SIP circuits. I would have increased call capacity and I could also work with the Telco to re-route calls destined to remote sites if their SIP circuit goes down to these circuits at the main office. That's just something not offered with old PRI circuits, or at least no provider I've worked with has offered that with old PRI. Then if I go with SIP, what's the best configuration in CUCM? I haven't worked much with SIP circuits, but the ones I have worked with, the gateway was just an h.323 GW in CUCM. Could it be added as a different type that would register with CUCM or is h.323 the best way to go?

Appreciate any inputs...

Regards,

Bill Hendrix

-----Original Message-----
From: matthn [at] gmail<mailto:matthn [at] gmail> [mailto:matthn [at] gmail]<mailto:[mailto:matthn [at] gmail]> On Behalf Of Nick Matthews
Sent: Thursday, June 14, 2012 10:17 PM
To: Adel Abushaev
Cc: Hendrix, George (Bill) @ LSG - STRATIS; cisco-voip [at] puck<mailto:cisco-voip [at] puck>
Subject: Re: [cisco-voip] CUCM SIP To PSTN

The scenario of:
CUCM--SIP--GW--SIP--Provider

The gateway magically turns into a CUBE, and there's a whole bunch of marketing and technical information on what happens when you do that.

If you compare these two solutions:
-CUCM direct SIP trunk to provider
-Use MTP to keep media to a single IP
-Or use NAT to keep internal addressing safe or -Use CUBE

There's a whole bunch of scalability and troubleshooting problems that can arise from the first. Having a demarcation point at the GW (CUBE) is extremely helpful. As well, it prevents you from needing to NAT SIP which historically is a pretty terrible idea. It also has some SIP security and flexibility options, and is a good centralization point for trunks.

I haven't worked with anyone doing the direct trunk from CUCM to provider. Many providers are going to make their own rules which will include an SBC (industry term for CUBE).

Short story - just use an SBC. There's about 3-4 compelling reasons.

-nick

On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev [at] gmail<mailto:adel.abushaev [at] gmail>> wrote:
> You can set up a SIP trunk to your SP, assuming that you are a SIP
> client of them.Otherwise, if you want to go over T1, then you need to
> terminate SIP on the GW to translate between SIP and ISDN PRI or
> whatever other signalling you are using between you and telco.
>
> A.
>
> On Thu, Jun 14, 2012 at 5:44 AM, <george.hendrix [at] l-3com<mailto:george.hendrix [at] l-3com>> wrote:
>> Hi everyone,
>>
>>
>>
>> I know with h.323 gateways in CUCM, you have to configure dial
>> peers on each gateway and for each CUCM. Is it possible to create a
>> SIP trunk directly from CUCM to the Telco? Without having to do any
>> dial peers on the gateway. Or would it CUCM SIP <> SIP GW <> SIP
>> PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Bill Hendrix
>>
>>
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck<mailto:cisco-voip [at] puck>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck<mailto:cisco-voip [at] puck>
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck<mailto:cisco-voip [at] puck>
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck<mailto:cisco-voip [at] puck>
https://puck.nether.net/mailman/listinfo/cisco-voip

itevomcid

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