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SIP as a gateway Protocol

 

 

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voicenoob at gmail

Nov 3, 2009, 3:09 PM

Post #1 of 17 (269 views)
Permalink
SIP as a gateway Protocol

Has anyone started using SIP on the PSTN gateway? I want to use it instead
of H.323 or MGCP and start migrating it to SIP on the gateway. Any
experience with this? Can I get Calling Name and Number from the PSTN side?


dan.voip at danofive

Nov 3, 2009, 3:17 PM

Post #2 of 17 (258 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Is there limitations to DTMF and requiring MTP to pass it through?

That's what I would be looking at I know we had similar issues with MPX as
a SIP trunk.

On Wed, Nov 4, 2009 at 9:09 AM, Voice Noob <voicenoob[at]gmail.com> wrote:

> Has anyone started using SIP on the PSTN gateway? I want to use it instead
> of H.323 or MGCP and start migrating it to SIP on the gateway. Any
> experience with this? Can I get Calling Name and Number from the PSTN side?
> _______________________________________________
> cisco-voip mailing list
> cisco-voip[at]puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


voicenoob at gmail

Nov 3, 2009, 3:31 PM

Post #3 of 17 (258 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

I don't know anyone chime in?

On Tue, Nov 3, 2009 at 5:17 PM, Daniel <dan.voip[at]danofive.id.au> wrote:

> Is there limitations to DTMF and requiring MTP to pass it through?
>
> That's what I would be looking at I know we had similar issues with MPX as
> a SIP trunk.
>
> On Wed, Nov 4, 2009 at 9:09 AM, Voice Noob <voicenoob[at]gmail.com> wrote:
>
>> Has anyone started using SIP on the PSTN gateway? I want to use it instead
>> of H.323 or MGCP and start migrating it to SIP on the gateway. Any
>> experience with this? Can I get Calling Name and Number from the PSTN side?
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>


SCASPER at mtb

Nov 3, 2009, 5:25 PM

Post #4 of 17 (250 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

I assume you are talking traditional analog and digital PSTN gateways, why are you considering migrating to SIP to control these as opposed to H323? .

Steve

>>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
Has anyone started using SIP on the PSTN gateway? I want to use it instead of H.323 or MGCP and start migrating it to SIP on the gateway. Any experience with this? Can I get Calling Name and Number from the PSTN side?
************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
************************************


thsglobal at gmail

Nov 3, 2009, 5:45 PM

Post #5 of 17 (250 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

I like the idea.

More and more SIP trunks will be turning up. Why bother having to go from
H323 to SIP. Simpler just to run SIP.

I also like SIP and how you can set it up to monitor the destination of your
dial-peers. Shut them down if a CCM is down.

Cheers,

Tim

On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:

> I assume you are talking traditional analog and digital PSTN
> gateways, why are you considering migrating to SIP to control these as
> opposed to H323? .
>
> Steve
>
> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>
> Has anyone started using SIP on the PSTN gateway? I want to use it instead
> of H.323 or MGCP and start migrating it to SIP on the gateway. Any
> experience with this? Can I get Calling Name and Number from the PSTN side?
>
> ************************************
> This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
> There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
> ************************************
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip[at]puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


--

Cheers,

Tim


Sent from Sydney, Nsw, Australia


thsglobal at gmail

Nov 3, 2009, 5:46 PM

Post #6 of 17 (250 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Also, SIP is slightly easier to troubleshoot than H323, much more so than
MGCP. (And I also dont like MGCP anyway :)

Cheers,

Tim.

On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:

> I like the idea.
>
> More and more SIP trunks will be turning up. Why bother having to go from
> H323 to SIP. Simpler just to run SIP.
>
> I also like SIP and how you can set it up to monitor the destination of
> your dial-peers. Shut them down if a CCM is down.
>
> Cheers,
>
> Tim
>
> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:
>
>> I assume you are talking traditional analog and digital PSTN
>> gateways, why are you considering migrating to SIP to control these as
>> opposed to H323? .
>>
>> Steve
>>
>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>>
>> Has anyone started using SIP on the PSTN gateway? I want to use it instead
>> of H.323 or MGCP and start migrating it to SIP on the gateway. Any
>> experience with this? Can I get Calling Name and Number from the PSTN side?
>>
>> ************************************
>> This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
>> There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
>> ************************************
>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
> --
>
> Cheers,
>
> Tim
>
>
> Sent from Sydney, Nsw, Australia




--

Cheers,

Tim


Sent from Sydney, Nsw, Australia


lelio at uoguelph

Nov 3, 2009, 6:18 PM

Post #7 of 17 (250 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

>From our initial conversations with our PSTN providers, SIP was a few years away with feature parity with H323/MGCP/PRI trunks.

FAX support was definately out of the question, and there were crazy requirements about not being able to do voice only on the ethernet trunk. We had to buy a data package that was no more than 50% voice traffic. For us, we get our internet through our regional network at dirt cheap prices because we basically run a co-op. For others it might make sense to move to the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup internet link is cheaper than the PSTN provider could price I believe.

The other thing was route diversity and multiple demarcs. I think those were quite expensive where as now, we get it at no extra cost.

I've long been a proponent of if it ain't broke, don't fix it. Even when we went to tender and ended up switching our PRIs to another local carrier, it was a LOT of work. I understood it saved us quite a bit of money, so it was worth it in the end for a three year contract. That being said, don't expect that SIP will be cheaper than PRIs and/or without it's own problems.

Caveat Emptor as my friend Caesar said.

----- Original Message -----
From: Tim Smith
To: STEVEN CASPER
Cc: CiscosupportUpuck
Sent: Tuesday, November 03, 2009 8:46 PM
Subject: Re: [cisco-voip] SIP as a gateway Protocol


Also, SIP is slightly easier to troubleshoot than H323, much more so than MGCP. (And I also dont like MGCP anyway :)

Cheers,

Tim.


On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:

I like the idea.

More and more SIP trunks will be turning up. Why bother having to go from H323 to SIP. Simpler just to run SIP.

I also like SIP and how you can set it up to monitor the destination of your dial-peers. Shut them down if a CCM is down.

Cheers,

Tim


On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:

I assume you are talking traditional analog and digital PSTN gateways, why are you considering migrating to SIP to control these as opposed to H323? .

Steve

>>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>

Has anyone started using SIP on the PSTN gateway? I want to use it instead of H.323 or MGCP and start migrating it to SIP on the gateway. Any experience with this? Can I get Calling Name and Number from the PSTN side?
************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
************************************


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





--

Cheers,

Tim


Sent from Sydney, Nsw, Australia



--

Cheers,

Tim


Sent from Sydney, Nsw, Australia


------------------------------------------------------------------------------


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


thsglobal at gmail

Nov 3, 2009, 6:32 PM

Post #8 of 17 (248 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

We dont have too many SIP providers here in Oz at the moment anyway.
We were talking about just using SIP between CCM and the Gateway. Vs MGCP
and H323.

Fax / modem could definitely be a good point though.

Cheers,

Tim.

On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca> wrote:

> From our initial conversations with our PSTN providers, SIP was a few
> years away with feature parity with H323/MGCP/PRI trunks.
>
> FAX support was definately out of the question, and there were crazy
> requirements about not being able to do voice only on the ethernet trunk. We
> had to buy a data package that was no more than 50% voice traffic. For us,
> we get our internet through our regional network at dirt cheap prices
> because we basically run a co-op. For others it might make sense to move to
> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup
> internet link is cheaper than the PSTN provider could price I believe.
>
> The other thing was route diversity and multiple demarcs. I think those
> were quite expensive where as now, we get it at no extra cost.
>
> I've long been a proponent of if it ain't broke, don't fix it. Even when we
> went to tender and ended up switching our PRIs to another local carrier, it
> was a LOT of work. I understood it saved us quite a bit of money, so it was
> worth it in the end for a three year contract. That being said, don't expect
> that SIP will be cheaper than PRIs and/or without it's own problems.
>
> Caveat Emptor as my friend Caesar said.
>
>
> ----- Original Message -----
> *From:* Tim Smith <thsglobal[at]gmail.com>
> *To:* STEVEN CASPER <SCASPER[at]mtb.com>
> *Cc:* CiscosupportUpuck <cisco-voip[at]puck.nether.net>
> *Sent:* Tuesday, November 03, 2009 8:46 PM
> *Subject:* Re: [cisco-voip] SIP as a gateway Protocol
>
> Also, SIP is slightly easier to troubleshoot than H323, much more so than
> MGCP. (And I also dont like MGCP anyway :)
>
> Cheers,
>
> Tim.
>
> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
>
>> I like the idea.
>>
>> More and more SIP trunks will be turning up. Why bother having to go from
>> H323 to SIP. Simpler just to run SIP.
>>
>> I also like SIP and how you can set it up to monitor the destination of
>> your dial-peers. Shut them down if a CCM is down.
>>
>> Cheers,
>>
>> Tim
>>
>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:
>>
>>> I assume you are talking traditional analog and digital PSTN
>>> gateways, why are you considering migrating to SIP to control these as
>>> opposed to H323? .
>>>
>>> Steve
>>>
>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>>>
>>> Has anyone started using SIP on the PSTN gateway? I want to use it
>>> instead of H.323 or MGCP and start migrating it to SIP on the gateway. Any
>>> experience with this? Can I get Calling Name and Number from the PSTN side?
>>>
>>> ************************************
>>> This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
>>> There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
>>> ************************************
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip[at]puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>>
>> --
>>
>> Cheers,
>>
>> Tim
>>
>>
>> Sent from Sydney, Nsw, Australia
>
>
>
>
> --
>
> Cheers,
>
> Tim
>
>
> Sent from Sydney, Nsw, Australia
>
> ------------------------------
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip[at]puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


--

Cheers,

Tim


Sent from Sydney, Nsw, Australia


matthnick at gmail

Nov 3, 2009, 7:46 PM

Post #9 of 17 (248 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

You can get an over-the-top SIP provider, but if you get voice quality
problems you'll have some trouble getting your ISP and SIP provider to
play nicely. Once it leaves your gateway you can't prove who may be
causing the problem if there is jitter or packet loss. Your ISP
probably won't have any idea how to deal with it, because for
traditional data these types of packet problems do not have much
consequence.

If you're cool with that, there are hundreds of providers of varying quality.

The suggestion is still to go with the data line from the SIP
provider. You may be able to save some money on equipment
consolidation or pricing depending on your volume / area as well.
It's not the best scenario for every case, but there are certainly
cases where it makes since and these cases are growing.


-nick

On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
> We dont have too many SIP providers here in Oz at the moment anyway.
> We were talking about just using SIP between CCM and the Gateway. Vs MGCP
> and H323.
>
> Fax / modem could definitely be a good point though.
>
> Cheers,
>
> Tim.
>
> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca> wrote:
>>
>> From our initial conversations with our PSTN providers, SIP was a few
>> years away with feature parity with H323/MGCP/PRI trunks.
>>
>> FAX support was definately out of the question, and there were crazy
>> requirements about not being able to do voice only on the ethernet trunk. We
>> had to buy a data package that was no more than 50% voice traffic. For us,
>> we get our internet through our regional network at dirt cheap prices
>> because we basically run a co-op. For others it might make sense to move to
>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup
>> internet link is cheaper than the PSTN provider could price I believe.
>>
>> The other thing was route diversity and multiple demarcs. I think those
>> were quite expensive where as now, we get it at no extra cost.
>>
>> I've long been a proponent of if it ain't broke, don't fix it. Even when
>> we went to tender and ended up switching our PRIs to another local carrier,
>> it was a LOT of work. I understood it saved us quite a bit of money, so it
>> was worth it in the end for a three year contract. That being said, don't
>> expect that SIP will be cheaper than PRIs and/or without it's own problems.
>>
>> Caveat Emptor as my friend Caesar said.
>>
>>
>> ----- Original Message -----
>> From: Tim Smith
>> To: STEVEN CASPER
>> Cc: CiscosupportUpuck
>> Sent: Tuesday, November 03, 2009 8:46 PM
>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>> Also, SIP is slightly easier to troubleshoot than H323, much more so than
>> MGCP. (And I also dont like MGCP anyway :)
>>
>> Cheers,
>>
>> Tim.
>>
>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
>>>
>>> I like the idea.
>>>
>>> More and more SIP trunks will be turning up. Why bother having to go from
>>> H323 to SIP. Simpler just to run SIP.
>>>
>>> I also like SIP and how you can set it up to monitor the destination of
>>> your dial-peers. Shut them down if a CCM is down.
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:
>>>>
>>>> I assume you are talking traditional analog and digital PSTN
>>>> gateways, why are you considering migrating to SIP to control these as
>>>> opposed to H323? .
>>>>
>>>> Steve
>>>>
>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
>>>> instead of H.323 or MGCP and start migrating it to SIP on the gateway. Any
>>>> experience with this? Can I get Calling Name and Number from the PSTN side?
>>>>
>>>> ************************************
>>>> This email may contain privileged and/or confidential information that
>>>> is intended solely for the use of the addressee. If you are not the
>>>> intended recipient or entity, you are strictly prohibited from disclosing,
>>>> copying, distributing or using any of the information contained in the
>>>> transmission. If you received this communication in error, please contact
>>>> the sender immediately and destroy the material in its entirety, whether
>>>> electronic or hard copy. This communication may contain nonpublic personal
>>>> information about consumers subject to the restrictions of the
>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or
>>>> indirectly reuse or disclose such information for any purpose other than to
>>>> provide the services for which you are receiving the information.
>>>> There are risks associated with the use of electronic transmission. The
>>>> sender of this information does not control the method of transmittal or
>>>> service providers and assumes no duty or obligation for the security,
>>>> receipt, or third party interception of this transmission.
>>>> ************************************
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip[at]puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>>
>>> Sent from Sydney, Nsw, Australia
>>
>>
>> --
>>
>> Cheers,
>>
>> Tim
>>
>>
>> Sent from Sydney, Nsw, Australia
>>
>> ________________________________
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> --
>
> Cheers,
>
> Tim
>
>
> Sent from Sydney, Nsw, Australia
> _______________________________________________
> cisco-voip mailing list
> cisco-voip[at]puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip[at]puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


thsglobal at gmail

Nov 3, 2009, 8:07 PM

Post #10 of 17 (248 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Hi Nick,

What about using SIP just as protocol to replace H323 / MGCP between CCM and
your Voice Gateway?

Cheers,

Tim

On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com> wrote:

> You can get an over-the-top SIP provider, but if you get voice quality
> problems you'll have some trouble getting your ISP and SIP provider to
> play nicely. Once it leaves your gateway you can't prove who may be
> causing the problem if there is jitter or packet loss. Your ISP
> probably won't have any idea how to deal with it, because for
> traditional data these types of packet problems do not have much
> consequence.
>
> If you're cool with that, there are hundreds of providers of varying
> quality.
>
> The suggestion is still to go with the data line from the SIP
> provider. You may be able to save some money on equipment
> consolidation or pricing depending on your volume / area as well.
> It's not the best scenario for every case, but there are certainly
> cases where it makes since and these cases are growing.
>
>
> -nick
>
> On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
> > We dont have too many SIP providers here in Oz at the moment anyway.
> > We were talking about just using SIP between CCM and the Gateway. Vs MGCP
> > and H323.
> >
> > Fax / modem could definitely be a good point though.
> >
> > Cheers,
> >
> > Tim.
> >
> > On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca>
> wrote:
> >>
> >> From our initial conversations with our PSTN providers, SIP was a few
> >> years away with feature parity with H323/MGCP/PRI trunks.
> >>
> >> FAX support was definately out of the question, and there were crazy
> >> requirements about not being able to do voice only on the ethernet
> trunk. We
> >> had to buy a data package that was no more than 50% voice traffic. For
> us,
> >> we get our internet through our regional network at dirt cheap prices
> >> because we basically run a co-op. For others it might make sense to move
> to
> >> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our
> backup
> >> internet link is cheaper than the PSTN provider could price I believe.
> >>
> >> The other thing was route diversity and multiple demarcs. I think those
> >> were quite expensive where as now, we get it at no extra cost.
> >>
> >> I've long been a proponent of if it ain't broke, don't fix it. Even when
> >> we went to tender and ended up switching our PRIs to another local
> carrier,
> >> it was a LOT of work. I understood it saved us quite a bit of money, so
> it
> >> was worth it in the end for a three year contract. That being said,
> don't
> >> expect that SIP will be cheaper than PRIs and/or without it's own
> problems.
> >>
> >> Caveat Emptor as my friend Caesar said.
> >>
> >>
> >> ----- Original Message -----
> >> From: Tim Smith
> >> To: STEVEN CASPER
> >> Cc: CiscosupportUpuck
> >> Sent: Tuesday, November 03, 2009 8:46 PM
> >> Subject: Re: [cisco-voip] SIP as a gateway Protocol
> >> Also, SIP is slightly easier to troubleshoot than H323, much more so
> than
> >> MGCP. (And I also dont like MGCP anyway :)
> >>
> >> Cheers,
> >>
> >> Tim.
> >>
> >> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
> >>>
> >>> I like the idea.
> >>>
> >>> More and more SIP trunks will be turning up. Why bother having to go
> from
> >>> H323 to SIP. Simpler just to run SIP.
> >>>
> >>> I also like SIP and how you can set it up to monitor the destination of
> >>> your dial-peers. Shut them down if a CCM is down.
> >>>
> >>> Cheers,
> >>>
> >>> Tim
> >>>
> >>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com>
> wrote:
> >>>>
> >>>> I assume you are talking traditional analog and digital PSTN
> >>>> gateways, why are you considering migrating to SIP to control these as
> >>>> opposed to H323? .
> >>>>
> >>>> Steve
> >>>>
> >>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
> >>>> Has anyone started using SIP on the PSTN gateway? I want to use it
> >>>> instead of H.323 or MGCP and start migrating it to SIP on the gateway.
> Any
> >>>> experience with this? Can I get Calling Name and Number from the PSTN
> side?
> >>>>
> >>>> ************************************
> >>>> This email may contain privileged and/or confidential information that
> >>>> is intended solely for the use of the addressee. If you are not the
> >>>> intended recipient or entity, you are strictly prohibited from
> disclosing,
> >>>> copying, distributing or using any of the information contained in the
> >>>> transmission. If you received this communication in error, please
> contact
> >>>> the sender immediately and destroy the material in its entirety,
> whether
> >>>> electronic or hard copy. This communication may contain nonpublic
> personal
> >>>> information about consumers subject to the restrictions of the
> >>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
> directly or
> >>>> indirectly reuse or disclose such information for any purpose other
> than to
> >>>> provide the services for which you are receiving the information.
> >>>> There are risks associated with the use of electronic transmission.
> The
> >>>> sender of this information does not control the method of transmittal
> or
> >>>> service providers and assumes no duty or obligation for the security,
> >>>> receipt, or third party interception of this transmission.
> >>>> ************************************
> >>>>
> >>>> _______________________________________________
> >>>> cisco-voip mailing list
> >>>> cisco-voip[at]puck.nether.net
> >>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>
> >>>
> >>>
> >>>
> >>> --
> >>>
> >>> Cheers,
> >>>
> >>> Tim
> >>>
> >>>
> >>> Sent from Sydney, Nsw, Australia
> >>
> >>
> >> --
> >>
> >> Cheers,
> >>
> >> Tim
> >>
> >>
> >> Sent from Sydney, Nsw, Australia
> >>
> >> ________________________________
> >>
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip[at]puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
> >
> > --
> >
> > Cheers,
> >
> > Tim
> >
> >
> > Sent from Sydney, Nsw, Australia
> > _______________________________________________
> > cisco-voip mailing list
> > cisco-voip[at]puck.nether.net
> > https://puck.nether.net/mailman/listinfo/cisco-voip
> >
> >
>



--

Cheers,

Tim


voicenoob at gmail

Nov 4, 2009, 5:37 AM

Post #11 of 17 (220 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Nick that is what I am asking. I in no way want to go with a SIP trunk to
the PSTN I just want to use SIP as my gateway protocol. So the Telco still
hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use
SIP. As far as why drop H.323 I don't have a reason to but when doing new
customer deployments I don't want to put one thing in and then migrate to
something else two years down the road.



So I ask my question again has anyone used SIP as their GW protocol instead
of H.323? Any problems or things I should look for? Should I just not do it
yet.



From: cisco-voip-bounces[at]puck.nether.net
[mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
Sent: Tuesday, November 03, 2009 10:07 PM
To: Nick Matthews
Cc: CiscosupportUpuck
Subject: Re: [cisco-voip] SIP as a gateway Protocol



Hi Nick,



What about using SIP just as protocol to replace H323 / MGCP between CCM and
your Voice Gateway?



Cheers,



Tim

On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com> wrote:

You can get an over-the-top SIP provider, but if you get voice quality
problems you'll have some trouble getting your ISP and SIP provider to
play nicely. Once it leaves your gateway you can't prove who may be
causing the problem if there is jitter or packet loss. Your ISP
probably won't have any idea how to deal with it, because for
traditional data these types of packet problems do not have much
consequence.

If you're cool with that, there are hundreds of providers of varying
quality.

The suggestion is still to go with the data line from the SIP
provider. You may be able to save some money on equipment
consolidation or pricing depending on your volume / area as well.
It's not the best scenario for every case, but there are certainly
cases where it makes since and these cases are growing.


-nick


On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
> We dont have too many SIP providers here in Oz at the moment anyway.
> We were talking about just using SIP between CCM and the Gateway. Vs MGCP
> and H323.
>
> Fax / modem could definitely be a good point though.
>
> Cheers,
>
> Tim.
>
> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca> wrote:
>>
>> From our initial conversations with our PSTN providers, SIP was a few
>> years away with feature parity with H323/MGCP/PRI trunks.
>>
>> FAX support was definately out of the question, and there were crazy
>> requirements about not being able to do voice only on the ethernet trunk.
We
>> had to buy a data package that was no more than 50% voice traffic. For
us,
>> we get our internet through our regional network at dirt cheap prices
>> because we basically run a co-op. For others it might make sense to move
to
>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup
>> internet link is cheaper than the PSTN provider could price I believe.
>>
>> The other thing was route diversity and multiple demarcs. I think those
>> were quite expensive where as now, we get it at no extra cost.
>>
>> I've long been a proponent of if it ain't broke, don't fix it. Even when
>> we went to tender and ended up switching our PRIs to another local
carrier,
>> it was a LOT of work. I understood it saved us quite a bit of money, so
it
>> was worth it in the end for a three year contract. That being said, don't
>> expect that SIP will be cheaper than PRIs and/or without it's own
problems.
>>
>> Caveat Emptor as my friend Caesar said.
>>
>>
>> ----- Original Message -----
>> From: Tim Smith
>> To: STEVEN CASPER
>> Cc: CiscosupportUpuck
>> Sent: Tuesday, November 03, 2009 8:46 PM
>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>> Also, SIP is slightly easier to troubleshoot than H323, much more so than
>> MGCP. (And I also dont like MGCP anyway :)
>>
>> Cheers,
>>
>> Tim.
>>
>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
>>>
>>> I like the idea.
>>>
>>> More and more SIP trunks will be turning up. Why bother having to go
from
>>> H323 to SIP. Simpler just to run SIP.
>>>
>>> I also like SIP and how you can set it up to monitor the destination of
>>> your dial-peers. Shut them down if a CCM is down.
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:
>>>>
>>>> I assume you are talking traditional analog and digital PSTN
>>>> gateways, why are you considering migrating to SIP to control these as
>>>> opposed to H323? .
>>>>
>>>> Steve
>>>>
>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
>>>> instead of H.323 or MGCP and start migrating it to SIP on the gateway.
Any
>>>> experience with this? Can I get Calling Name and Number from the PSTN
side?
>>>>
>>>> ************************************
>>>> This email may contain privileged and/or confidential information that
>>>> is intended solely for the use of the addressee. If you are not the
>>>> intended recipient or entity, you are strictly prohibited from
disclosing,
>>>> copying, distributing or using any of the information contained in the
>>>> transmission. If you received this communication in error, please
contact
>>>> the sender immediately and destroy the material in its entirety,
whether
>>>> electronic or hard copy. This communication may contain nonpublic
personal
>>>> information about consumers subject to the restrictions of the
>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
directly or
>>>> indirectly reuse or disclose such information for any purpose other
than to
>>>> provide the services for which you are receiving the information.
>>>> There are risks associated with the use of electronic transmission.
The
>>>> sender of this information does not control the method of transmittal
or
>>>> service providers and assumes no duty or obligation for the security,
>>>> receipt, or third party interception of this transmission.
>>>> ************************************
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip[at]puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>>
>>> Sent from Sydney, Nsw, Australia
>>
>>
>> --
>>
>> Cheers,
>>
>> Tim
>>
>>
>> Sent from Sydney, Nsw, Australia
>>
>> ________________________________
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> --
>
> Cheers,
>
> Tim
>
>
> Sent from Sydney, Nsw, Australia
> _______________________________________________
> cisco-voip mailing list
> cisco-voip[at]puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>




--

Cheers,

Tim


matthnick at gmail

Nov 4, 2009, 6:57 AM

Post #12 of 17 (219 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

I think there are a few different factors - but it's the protocol I
would use if I was administering my network.

We see a lot of SIP gateways, and it's definitely being deployed.

Some of the advantages:
-Easy to troubleshoot. You can read up on SIP and learn the basics
2-3x faster than other protocols. It's clear and concise for the most
part.
-Interop. Most of the new devices coming out are all running SIP.
You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
experience with it already.
-Easier transition to SIP as your PSTN connection (last post) if/when
you decide to make that jump.
-If you're already running H323, switching over is pretty easy.

Other considerations:
-H323 is still the best at video, and for a while, there doesn't
appear to be any real alternatives.
-MGCP is still the only 'centralized dial plan' protocol where you
don't have to do anything on your gateways at all. If you're not good
with IOS and just 'want it to work', this is still the protocol to
look at. It comes with it's own troubles, bugs, and instability
because of it.
-Some older devices don't support SIP yet, and you may still be
running H323 in the network anyways.
-For more advanced call flows and designs, you may run into some
unsupported features. (Like using ANN for ringback, I think that is
still H323 only).
-I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
If you have older platforms like the 3700, CMM, that won't run 20.T, I
would stick to your existing protocol. Likewise for CUCM versions
prior to 6.x. The SIP stacks in the versions prior just aren't as
stable or have as many features.


-nick


On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob[at]gmail.com> wrote:
> Nick that is what I am  asking. I in no way want to go with a SIP trunk to
> the PSTN I just want to use SIP as my gateway protocol. So the Telco still
> hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use
> SIP. As far as why drop H.323 I don’t have a reason to but when doing new
> customer deployments I don’t want to put one thing in and then migrate to
> something else two years down the road.
>
>
>
> So I ask my question again has anyone used SIP as their GW protocol instead
> of H.323? Any problems or things I should look for? Should I just not do it
> yet.
>
>
>
> From: cisco-voip-bounces[at]puck.nether.net
> [mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
> Sent: Tuesday, November 03, 2009 10:07 PM
> To: Nick Matthews
> Cc: CiscosupportUpuck
> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>
>
>
> Hi Nick,
>
>
>
> What about using SIP just as protocol to replace H323 / MGCP between CCM and
> your Voice Gateway?
>
>
>
> Cheers,
>
>
>
> Tim
>
> On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com> wrote:
>
> You can get an over-the-top SIP provider, but if you get voice quality
> problems you'll have some trouble getting your ISP and SIP provider to
> play nicely.  Once it leaves your gateway you can't prove who may be
> causing the problem if there is jitter or packet loss.  Your ISP
> probably won't have any idea how to deal with it, because for
> traditional data these types of packet problems do not have much
> consequence.
>
> If you're cool with that, there are hundreds of providers of varying
> quality.
>
> The suggestion is still to go with the data line from the SIP
> provider.  You may be able to save some money on equipment
> consolidation or pricing depending on your volume / area as well.
> It's not the best scenario for every case, but there are certainly
> cases where it makes since and these cases are growing.
>
>
> -nick
>
> On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
>> We dont have too many SIP providers here in Oz at the moment anyway.
>> We were talking about just using SIP between CCM and the Gateway. Vs MGCP
>> and H323.
>>
>> Fax / modem could definitely be a good point though.
>>
>> Cheers,
>>
>> Tim.
>>
>> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca> wrote:
>>>
>>> From our initial conversations with our PSTN providers, SIP was a few
>>> years away with feature parity with H323/MGCP/PRI trunks.
>>>
>>> FAX support was definately out of the question, and there were crazy
>>> requirements about not being able to do voice only on the ethernet trunk.
>>> We
>>> had to buy a data package that was no more than 50% voice traffic. For
>>> us,
>>> we get our internet through our regional network at dirt cheap prices
>>> because we basically run a co-op. For others it might make sense to move
>>> to
>>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup
>>> internet link is cheaper than the PSTN provider could price I believe.
>>>
>>> The other thing was route diversity and multiple demarcs. I think those
>>> were quite expensive where as now, we get it at no extra cost.
>>>
>>> I've long been a proponent of if it ain't broke, don't fix it. Even when
>>> we went to tender and ended up switching our PRIs to another local
>>> carrier,
>>> it was a LOT of work. I understood it saved us quite a bit of money, so
>>> it
>>> was worth it in the end for a three year contract. That being said, don't
>>> expect that SIP will be cheaper than PRIs and/or without it's own
>>> problems.
>>>
>>> Caveat Emptor as my friend Caesar said.
>>>
>>>
>>> ----- Original Message -----
>>> From: Tim Smith
>>> To: STEVEN CASPER
>>> Cc: CiscosupportUpuck
>>> Sent: Tuesday, November 03, 2009 8:46 PM
>>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>>> Also, SIP is slightly easier to troubleshoot than H323, much more so than
>>> MGCP. (And I also dont like MGCP anyway :)
>>>
>>> Cheers,
>>>
>>> Tim.
>>>
>>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
>>>>
>>>> I like the idea.
>>>>
>>>> More and more SIP trunks will be turning up. Why bother having to go
>>>> from
>>>> H323 to SIP. Simpler just to run SIP.
>>>>
>>>> I also like SIP and how you can set it up to monitor the destination of
>>>> your dial-peers. Shut them down if a CCM is down.
>>>>
>>>> Cheers,
>>>>
>>>> Tim
>>>>
>>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com> wrote:
>>>>>
>>>>> I assume you are talking traditional analog and digital PSTN
>>>>> gateways, why are you considering migrating to SIP to control these as
>>>>> opposed to H323? .
>>>>>
>>>>> Steve
>>>>>
>>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
>>>>> instead of H.323 or MGCP and start migrating it to SIP on the gateway.
>>>>> Any
>>>>> experience with this? Can I get Calling Name and Number from the PSTN
>>>>> side?
>>>>>
>>>>> ************************************
>>>>> This email may contain privileged and/or confidential information that
>>>>> is intended solely for the use of the addressee.  If you are not the
>>>>> intended recipient or entity, you are strictly prohibited from
>>>>> disclosing,
>>>>> copying, distributing or using any of the information contained in the
>>>>> transmission.  If you received this communication in error, please
>>>>> contact
>>>>> the sender immediately and destroy the material in its entirety,
>>>>> whether
>>>>> electronic or hard copy.  This communication may contain nonpublic
>>>>> personal
>>>>> information about consumers subject to the restrictions of the
>>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act.  You may not
>>>>> directly or
>>>>> indirectly reuse or disclose such information for any purpose other
>>>>> than to
>>>>> provide the services for which you are receiving the information.
>>>>> There are risks associated with the use of electronic transmission.
>>>>>  The
>>>>> sender of this information does not control the method of transmittal
>>>>> or
>>>>> service providers and assumes no duty or obligation for the security,
>>>>> receipt, or third party interception of this transmission.
>>>>> ************************************
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip[at]puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Cheers,
>>>>
>>>> Tim
>>>>
>>>>
>>>> Sent from Sydney, Nsw, Australia
>>>
>>>
>>> --
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>>
>>> Sent from Sydney, Nsw, Australia
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip[at]puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>> --
>>
>> Cheers,
>>
>> Tim
>>
>>
>> Sent from Sydney, Nsw, Australia
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
> --
>
> Cheers,
>
> Tim
>
>
_______________________________________________
cisco-voip mailing list
cisco-voip[at]puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip


voicenoob at gmail

Nov 4, 2009, 8:26 AM

Post #13 of 17 (217 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Thanks for your response. You mentioned SIP trunks in your respoonse. Just
to verify again I am NOT talking about a SIP trunk. I am talking at a SIP
gateway.

On Wed, Nov 4, 2009 at 8:57 AM, Nick Matthews <matthnick[at]gmail.com> wrote:

> I think there are a few different factors - but it's the protocol I
> would use if I was administering my network.
>
> We see a lot of SIP gateways, and it's definitely being deployed.
>
> Some of the advantages:
> -Easy to troubleshoot. You can read up on SIP and learn the basics
> 2-3x faster than other protocols. It's clear and concise for the most
> part.
> -Interop. Most of the new devices coming out are all running SIP.
> You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
> experience with it already.
> -Easier transition to SIP as your PSTN connection (last post) if/when
> you decide to make that jump.
> -If you're already running H323, switching over is pretty easy.
>
> Other considerations:
> -H323 is still the best at video, and for a while, there doesn't
> appear to be any real alternatives.
> -MGCP is still the only 'centralized dial plan' protocol where you
> don't have to do anything on your gateways at all. If you're not good
> with IOS and just 'want it to work', this is still the protocol to
> look at. It comes with it's own troubles, bugs, and instability
> because of it.
> -Some older devices don't support SIP yet, and you may still be
> running H323 in the network anyways.
> -For more advanced call flows and designs, you may run into some
> unsupported features. (Like using ANN for ringback, I think that is
> still H323 only).
> -I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
> If you have older platforms like the 3700, CMM, that won't run 20.T, I
> would stick to your existing protocol. Likewise for CUCM versions
> prior to 6.x. The SIP stacks in the versions prior just aren't as
> stable or have as many features.
>
>
> -nick
>
>
> On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob[at]gmail.com> wrote:
> > Nick that is what I am asking. I in no way want to go with a SIP trunk
> to
> > the PSTN I just want to use SIP as my gateway protocol. So the Telco
> still
> > hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use
> > SIP. As far as why drop H.323 I don’t have a reason to but when doing new
> > customer deployments I don’t want to put one thing in and then migrate to
> > something else two years down the road.
> >
> >
> >
> > So I ask my question again has anyone used SIP as their GW protocol
> instead
> > of H.323? Any problems or things I should look for? Should I just not do
> it
> > yet.
> >
> >
> >
> > From: cisco-voip-bounces[at]puck.nether.net
> > [mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
> > Sent: Tuesday, November 03, 2009 10:07 PM
> > To: Nick Matthews
> > Cc: CiscosupportUpuck
> > Subject: Re: [cisco-voip] SIP as a gateway Protocol
> >
> >
> >
> > Hi Nick,
> >
> >
> >
> > What about using SIP just as protocol to replace H323 / MGCP between CCM
> and
> > your Voice Gateway?
> >
> >
> >
> > Cheers,
> >
> >
> >
> > Tim
> >
> > On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com>
> wrote:
> >
> > You can get an over-the-top SIP provider, but if you get voice quality
> > problems you'll have some trouble getting your ISP and SIP provider to
> > play nicely. Once it leaves your gateway you can't prove who may be
> > causing the problem if there is jitter or packet loss. Your ISP
> > probably won't have any idea how to deal with it, because for
> > traditional data these types of packet problems do not have much
> > consequence.
> >
> > If you're cool with that, there are hundreds of providers of varying
> > quality.
> >
> > The suggestion is still to go with the data line from the SIP
> > provider. You may be able to save some money on equipment
> > consolidation or pricing depending on your volume / area as well.
> > It's not the best scenario for every case, but there are certainly
> > cases where it makes since and these cases are growing.
> >
> >
> > -nick
> >
> > On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
> >> We dont have too many SIP providers here in Oz at the moment anyway.
> >> We were talking about just using SIP between CCM and the Gateway. Vs
> MGCP
> >> and H323.
> >>
> >> Fax / modem could definitely be a good point though.
> >>
> >> Cheers,
> >>
> >> Tim.
> >>
> >> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca>
> wrote:
> >>>
> >>> From our initial conversations with our PSTN providers, SIP was a few
> >>> years away with feature parity with H323/MGCP/PRI trunks.
> >>>
> >>> FAX support was definately out of the question, and there were crazy
> >>> requirements about not being able to do voice only on the ethernet
> trunk.
> >>> We
> >>> had to buy a data package that was no more than 50% voice traffic. For
> >>> us,
> >>> we get our internet through our regional network at dirt cheap prices
> >>> because we basically run a co-op. For others it might make sense to
> move
> >>> to
> >>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our
> backup
> >>> internet link is cheaper than the PSTN provider could price I believe.
> >>>
> >>> The other thing was route diversity and multiple demarcs. I think those
> >>> were quite expensive where as now, we get it at no extra cost.
> >>>
> >>> I've long been a proponent of if it ain't broke, don't fix it. Even
> when
> >>> we went to tender and ended up switching our PRIs to another local
> >>> carrier,
> >>> it was a LOT of work. I understood it saved us quite a bit of money, so
> >>> it
> >>> was worth it in the end for a three year contract. That being said,
> don't
> >>> expect that SIP will be cheaper than PRIs and/or without it's own
> >>> problems.
> >>>
> >>> Caveat Emptor as my friend Caesar said.
> >>>
> >>>
> >>> ----- Original Message -----
> >>> From: Tim Smith
> >>> To: STEVEN CASPER
> >>> Cc: CiscosupportUpuck
> >>> Sent: Tuesday, November 03, 2009 8:46 PM
> >>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
> >>> Also, SIP is slightly easier to troubleshoot than H323, much more so
> than
> >>> MGCP. (And I also dont like MGCP anyway :)
> >>>
> >>> Cheers,
> >>>
> >>> Tim.
> >>>
> >>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com>
> wrote:
> >>>>
> >>>> I like the idea.
> >>>>
> >>>> More and more SIP trunks will be turning up. Why bother having to go
> >>>> from
> >>>> H323 to SIP. Simpler just to run SIP.
> >>>>
> >>>> I also like SIP and how you can set it up to monitor the destination
> of
> >>>> your dial-peers. Shut them down if a CCM is down.
> >>>>
> >>>> Cheers,
> >>>>
> >>>> Tim
> >>>>
> >>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com>
> wrote:
> >>>>>
> >>>>> I assume you are talking traditional analog and digital PSTN
> >>>>> gateways, why are you considering migrating to SIP to control these
> as
> >>>>> opposed to H323? .
> >>>>>
> >>>>> Steve
> >>>>>
> >>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
> >>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
> >>>>> instead of H.323 or MGCP and start migrating it to SIP on the
> gateway.
> >>>>> Any
> >>>>> experience with this? Can I get Calling Name and Number from the PSTN
> >>>>> side?
> >>>>>
> >>>>> ************************************
> >>>>> This email may contain privileged and/or confidential information
> that
> >>>>> is intended solely for the use of the addressee. If you are not the
> >>>>> intended recipient or entity, you are strictly prohibited from
> >>>>> disclosing,
> >>>>> copying, distributing or using any of the information contained in
> the
> >>>>> transmission. If you received this communication in error, please
> >>>>> contact
> >>>>> the sender immediately and destroy the material in its entirety,
> >>>>> whether
> >>>>> electronic or hard copy. This communication may contain nonpublic
> >>>>> personal
> >>>>> information about consumers subject to the restrictions of the
> >>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
> >>>>> directly or
> >>>>> indirectly reuse or disclose such information for any purpose other
> >>>>> than to
> >>>>> provide the services for which you are receiving the information.
> >>>>> There are risks associated with the use of electronic transmission.
> >>>>> The
> >>>>> sender of this information does not control the method of transmittal
> >>>>> or
> >>>>> service providers and assumes no duty or obligation for the security,
> >>>>> receipt, or third party interception of this transmission.
> >>>>> ************************************
> >>>>>
> >>>>> _______________________________________________
> >>>>> cisco-voip mailing list
> >>>>> cisco-voip[at]puck.nether.net
> >>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>>
> >>>> Cheers,
> >>>>
> >>>> Tim
> >>>>
> >>>>
> >>>> Sent from Sydney, Nsw, Australia
> >>>
> >>>
> >>> --
> >>>
> >>> Cheers,
> >>>
> >>> Tim
> >>>
> >>>
> >>> Sent from Sydney, Nsw, Australia
> >>>
> >>> ________________________________
> >>>
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip[at]puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
> >>
> >> --
> >>
> >> Cheers,
> >>
> >> Tim
> >>
> >>
> >> Sent from Sydney, Nsw, Australia
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip[at]puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
> >
> >
> > --
> >
> > Cheers,
> >
> > Tim
> >
> >
>


rratliff at cisco

Nov 4, 2009, 8:31 AM

Post #14 of 17 (217 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

You define the SIP gateway in CUCM as a SIP trunk. That's what he is
referring to.

-Ryan

On Nov 4, 2009, at 11:26 AM, Voice Noob wrote:

Thanks for your response. You mentioned SIP trunks in your respoonse.
Just to verify again I am NOT talking about a SIP trunk. I am talking
at a SIP gateway.

On Wed, Nov 4, 2009 at 8:57 AM, Nick Matthews <matthnick[at]gmail.com>
wrote:
I think there are a few different factors - but it's the protocol I
would use if I was administering my network.

We see a lot of SIP gateways, and it's definitely being deployed.

Some of the advantages:
-Easy to troubleshoot. You can read up on SIP and learn the basics
2-3x faster than other protocols. It's clear and concise for the most
part.
-Interop. Most of the new devices coming out are all running SIP.
You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
experience with it already.
-Easier transition to SIP as your PSTN connection (last post) if/when
you decide to make that jump.
-If you're already running H323, switching over is pretty easy.

Other considerations:
-H323 is still the best at video, and for a while, there doesn't
appear to be any real alternatives.
-MGCP is still the only 'centralized dial plan' protocol where you
don't have to do anything on your gateways at all. If you're not good
with IOS and just 'want it to work', this is still the protocol to
look at. It comes with it's own troubles, bugs, and instability
because of it.
-Some older devices don't support SIP yet, and you may still be
running H323 in the network anyways.
-For more advanced call flows and designs, you may run into some
unsupported features. (Like using ANN for ringback, I think that is
still H323 only).
-I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
If you have older platforms like the 3700, CMM, that won't run 20.T, I
would stick to your existing protocol. Likewise for CUCM versions
prior to 6.x. The SIP stacks in the versions prior just aren't as
stable or have as many features.


-nick


On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob[at]gmail.com> wrote:
> Nick that is what I am asking. I in no way want to go with a SIP
trunk to
> the PSTN I just want to use SIP as my gateway protocol. So the
Telco still
> hands me a PRI / FXO lines and instead of using MGCP or H.323 I
would use
> SIP. As far as why drop H.323 I don’t have a reason to but when
doing new
> customer deployments I don’t want to put one thing in and then
migrate to
> something else two years down the road.
>
>
>
> So I ask my question again has anyone used SIP as their GW protocol
instead
> of H.323? Any problems or things I should look for? Should I just
not do it
> yet.
>
>
>
> From: cisco-voip-bounces[at]puck.nether.net
> [mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
> Sent: Tuesday, November 03, 2009 10:07 PM
> To: Nick Matthews
> Cc: CiscosupportUpuck
> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>
>
>
> Hi Nick,
>
>
>
> What about using SIP just as protocol to replace H323 / MGCP
between CCM and
> your Voice Gateway?
>
>
>
> Cheers,
>
>
>
> Tim
>
> On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com>
wrote:
>
> You can get an over-the-top SIP provider, but if you get voice
quality
> problems you'll have some trouble getting your ISP and SIP provider
to
> play nicely. Once it leaves your gateway you can't prove who may be
> causing the problem if there is jitter or packet loss. Your ISP
> probably won't have any idea how to deal with it, because for
> traditional data these types of packet problems do not have much
> consequence.
>
> If you're cool with that, there are hundreds of providers of varying
> quality.
>
> The suggestion is still to go with the data line from the SIP
> provider. You may be able to save some money on equipment
> consolidation or pricing depending on your volume / area as well.
> It's not the best scenario for every case, but there are certainly
> cases where it makes since and these cases are growing.
>
>
> -nick
>
> On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com>
wrote:
>> We dont have too many SIP providers here in Oz at the moment anyway.
>> We were talking about just using SIP between CCM and the Gateway.
Vs MGCP
>> and H323.
>>
>> Fax / modem could definitely be a good point though.
>>
>> Cheers,
>>
>> Tim.
>>
>> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca>
wrote:
>>>
>>> From our initial conversations with our PSTN providers, SIP was a
few
>>> years away with feature parity with H323/MGCP/PRI trunks.
>>>
>>> FAX support was definately out of the question, and there were
crazy
>>> requirements about not being able to do voice only on the
ethernet trunk.
>>> We
>>> had to buy a data package that was no more than 50% voice
traffic. For
>>> us,
>>> we get our internet through our regional network at dirt cheap
prices
>>> because we basically run a co-op. For others it might make sense
to move
>>> to
>>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even
our backup
>>> internet link is cheaper than the PSTN provider could price I
believe.
>>>
>>> The other thing was route diversity and multiple demarcs. I think
those
>>> were quite expensive where as now, we get it at no extra cost.
>>>
>>> I've long been a proponent of if it ain't broke, don't fix it.
Even when
>>> we went to tender and ended up switching our PRIs to another local
>>> carrier,
>>> it was a LOT of work. I understood it saved us quite a bit of
money, so
>>> it
>>> was worth it in the end for a three year contract. That being
said, don't
>>> expect that SIP will be cheaper than PRIs and/or without it's own
>>> problems.
>>>
>>> Caveat Emptor as my friend Caesar said.
>>>
>>>
>>> ----- Original Message -----
>>> From: Tim Smith
>>> To: STEVEN CASPER
>>> Cc: CiscosupportUpuck
>>> Sent: Tuesday, November 03, 2009 8:46 PM
>>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>>> Also, SIP is slightly easier to troubleshoot than H323, much more
so than
>>> MGCP. (And I also dont like MGCP anyway :)
>>>
>>> Cheers,
>>>
>>> Tim.
>>>
>>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com>
wrote:
>>>>
>>>> I like the idea.
>>>>
>>>> More and more SIP trunks will be turning up. Why bother having
to go
>>>> from
>>>> H323 to SIP. Simpler just to run SIP.
>>>>
>>>> I also like SIP and how you can set it up to monitor the
destination of
>>>> your dial-peers. Shut them down if a CCM is down.
>>>>
>>>> Cheers,
>>>>
>>>> Tim
>>>>
>>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com>
wrote:
>>>>>
>>>>> I assume you are talking traditional analog and digital PSTN
>>>>> gateways, why are you considering migrating to SIP to control
these as
>>>>> opposed to H323? .
>>>>>
>>>>> Steve
>>>>>
>>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>>>>> Has anyone started using SIP on the PSTN gateway? I want to use
it
>>>>> instead of H.323 or MGCP and start migrating it to SIP on the
gateway.
>>>>> Any
>>>>> experience with this? Can I get Calling Name and Number from
the PSTN
>>>>> side?
>>>>>
>>>>> ************************************
>>>>> This email may contain privileged and/or confidential
information that
>>>>> is intended solely for the use of the addressee. If you are
not the
>>>>> intended recipient or entity, you are strictly prohibited from
>>>>> disclosing,
>>>>> copying, distributing or using any of the information contained
in the
>>>>> transmission. If you received this communication in error,
please
>>>>> contact
>>>>> the sender immediately and destroy the material in its entirety,
>>>>> whether
>>>>> electronic or hard copy. This communication may contain
nonpublic
>>>>> personal
>>>>> information about consumers subject to the restrictions of the
>>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
>>>>> directly or
>>>>> indirectly reuse or disclose such information for any purpose
other
>>>>> than to
>>>>> provide the services for which you are receiving the information.
>>>>> There are risks associated with the use of electronic
transmission.
>>>>> The
>>>>> sender of this information does not control the method of
transmittal
>>>>> or
>>>>> service providers and assumes no duty or obligation for the
security,
>>>>> receipt, or third party interception of this transmission.
>>>>> ************************************
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip[at]puck.nether.net
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Cheers,
>>>>
>>>> Tim
>>>>
>>>>
>>>> Sent from Sydney, Nsw, Australia
>>>
>>>
>>> --
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>>
>>> Sent from Sydney, Nsw, Australia
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip[at]puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>> --
>>
>> Cheers,
>>
>> Tim
>>
>>
>> Sent from Sydney, Nsw, Australia
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
> --
>
> Cheers,
>
> Tim
>
>

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


voicenoob at gmail

Nov 4, 2009, 8:31 AM

Post #15 of 17 (217 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Huh. Thanks for the clarification. I thought it was just a gateway option.

On Wed, Nov 4, 2009 at 10:31 AM, Ryan Ratliff <rratliff[at]cisco.com> wrote:

> You define the SIP gateway in CUCM as a SIP trunk. That's what he is
> referring to.
>
> -Ryan
>
> On Nov 4, 2009, at 11:26 AM, Voice Noob wrote:
>
> Thanks for your response. You mentioned SIP trunks in your respoonse. Just
> to verify again I am NOT talking about a SIP trunk. I am talking at a SIP
> gateway.
>
> On Wed, Nov 4, 2009 at 8:57 AM, Nick Matthews <matthnick[at]gmail.com> wrote:
>
>> I think there are a few different factors - but it's the protocol I
>> would use if I was administering my network.
>>
>> We see a lot of SIP gateways, and it's definitely being deployed.
>>
>> Some of the advantages:
>> -Easy to troubleshoot. You can read up on SIP and learn the basics
>> 2-3x faster than other protocols. It's clear and concise for the most
>> part.
>> -Interop. Most of the new devices coming out are all running SIP.
>> You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
>> experience with it already.
>> -Easier transition to SIP as your PSTN connection (last post) if/when
>> you decide to make that jump.
>> -If you're already running H323, switching over is pretty easy.
>>
>> Other considerations:
>> -H323 is still the best at video, and for a while, there doesn't
>> appear to be any real alternatives.
>> -MGCP is still the only 'centralized dial plan' protocol where you
>> don't have to do anything on your gateways at all. If you're not good
>> with IOS and just 'want it to work', this is still the protocol to
>> look at. It comes with it's own troubles, bugs, and instability
>> because of it.
>> -Some older devices don't support SIP yet, and you may still be
>> running H323 in the network anyways.
>> -For more advanced call flows and designs, you may run into some
>> unsupported features. (Like using ANN for ringback, I think that is
>> still H323 only).
>> -I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
>> If you have older platforms like the 3700, CMM, that won't run 20.T, I
>> would stick to your existing protocol. Likewise for CUCM versions
>> prior to 6.x. The SIP stacks in the versions prior just aren't as
>> stable or have as many features.
>>
>>
>> -nick
>>
>>
>> On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob[at]gmail.com> wrote:
>> > Nick that is what I am asking. I in no way want to go with a SIP trunk
>> to
>> > the PSTN I just want to use SIP as my gateway protocol. So the Telco
>> still
>> > hands me a PRI / FXO lines and instead of using MGCP or H.323 I would
>> use
>> > SIP. As far as why drop H.323 I don’t have a reason to but when doing
>> new
>> > customer deployments I don’t want to put one thing in and then migrate
>> to
>> > something else two years down the road.
>> >
>> >
>> >
>> > So I ask my question again has anyone used SIP as their GW protocol
>> instead
>> > of H.323? Any problems or things I should look for? Should I just not do
>> it
>> > yet.
>> >
>> >
>> >
>> > From: cisco-voip-bounces[at]puck.nether.net
>> > [mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
>> > Sent: Tuesday, November 03, 2009 10:07 PM
>> > To: Nick Matthews
>> > Cc: CiscosupportUpuck
>> > Subject: Re: [cisco-voip] SIP as a gateway Protocol
>> >
>> >
>> >
>> > Hi Nick,
>> >
>> >
>> >
>> > What about using SIP just as protocol to replace H323 / MGCP between CCM
>> and
>> > your Voice Gateway?
>> >
>> >
>> >
>> > Cheers,
>> >
>> >
>> >
>> > Tim
>> >
>> > On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com>
>> wrote:
>> >
>> > You can get an over-the-top SIP provider, but if you get voice quality
>> > problems you'll have some trouble getting your ISP and SIP provider to
>> > play nicely. Once it leaves your gateway you can't prove who may be
>> > causing the problem if there is jitter or packet loss. Your ISP
>> > probably won't have any idea how to deal with it, because for
>> > traditional data these types of packet problems do not have much
>> > consequence.
>> >
>> > If you're cool with that, there are hundreds of providers of varying
>> > quality.
>> >
>> > The suggestion is still to go with the data line from the SIP
>> > provider. You may be able to save some money on equipment
>> > consolidation or pricing depending on your volume / area as well.
>> > It's not the best scenario for every case, but there are certainly
>> > cases where it makes since and these cases are growing.
>> >
>> >
>> > -nick
>> >
>> > On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
>> >> We dont have too many SIP providers here in Oz at the moment anyway.
>> >> We were talking about just using SIP between CCM and the Gateway. Vs
>> MGCP
>> >> and H323.
>> >>
>> >> Fax / modem could definitely be a good point though.
>> >>
>> >> Cheers,
>> >>
>> >> Tim.
>> >>
>> >> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca>
>> wrote:
>> >>>
>> >>> From our initial conversations with our PSTN providers, SIP was a few
>> >>> years away with feature parity with H323/MGCP/PRI trunks.
>> >>>
>> >>> FAX support was definately out of the question, and there were crazy
>> >>> requirements about not being able to do voice only on the ethernet
>> trunk.
>> >>> We
>> >>> had to buy a data package that was no more than 50% voice traffic. For
>> >>> us,
>> >>> we get our internet through our regional network at dirt cheap prices
>> >>> because we basically run a co-op. For others it might make sense to
>> move
>> >>> to
>> >>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our
>> backup
>> >>> internet link is cheaper than the PSTN provider could price I believe.
>> >>>
>> >>> The other thing was route diversity and multiple demarcs. I think
>> those
>> >>> were quite expensive where as now, we get it at no extra cost.
>> >>>
>> >>> I've long been a proponent of if it ain't broke, don't fix it. Even
>> when
>> >>> we went to tender and ended up switching our PRIs to another local
>> >>> carrier,
>> >>> it was a LOT of work. I understood it saved us quite a bit of money,
>> so
>> >>> it
>> >>> was worth it in the end for a three year contract. That being said,
>> don't
>> >>> expect that SIP will be cheaper than PRIs and/or without it's own
>> >>> problems.
>> >>>
>> >>> Caveat Emptor as my friend Caesar said.
>> >>>
>> >>>
>> >>> ----- Original Message -----
>> >>> From: Tim Smith
>> >>> To: STEVEN CASPER
>> >>> Cc: CiscosupportUpuck
>> >>> Sent: Tuesday, November 03, 2009 8:46 PM
>> >>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>> >>> Also, SIP is slightly easier to troubleshoot than H323, much more so
>> than
>> >>> MGCP. (And I also dont like MGCP anyway :)
>> >>>
>> >>> Cheers,
>> >>>
>> >>> Tim.
>> >>>
>> >>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com>
>> wrote:
>> >>>>
>> >>>> I like the idea.
>> >>>>
>> >>>> More and more SIP trunks will be turning up. Why bother having to go
>> >>>> from
>> >>>> H323 to SIP. Simpler just to run SIP.
>> >>>>
>> >>>> I also like SIP and how you can set it up to monitor the destination
>> of
>> >>>> your dial-peers. Shut them down if a CCM is down.
>> >>>>
>> >>>> Cheers,
>> >>>>
>> >>>> Tim
>> >>>>
>> >>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com>
>> wrote:
>> >>>>>
>> >>>>> I assume you are talking traditional analog and digital PSTN
>> >>>>> gateways, why are you considering migrating to SIP to control these
>> as
>> >>>>> opposed to H323? .
>> >>>>>
>> >>>>> Steve
>> >>>>>
>> >>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
>> >>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
>> >>>>> instead of H.323 or MGCP and start migrating it to SIP on the
>> gateway.
>> >>>>> Any
>> >>>>> experience with this? Can I get Calling Name and Number from the
>> PSTN
>> >>>>> side?
>> >>>>>
>> >>>>> ************************************
>> >>>>> This email may contain privileged and/or confidential information
>> that
>> >>>>> is intended solely for the use of the addressee. If you are not the
>> >>>>> intended recipient or entity, you are strictly prohibited from
>> >>>>> disclosing,
>> >>>>> copying, distributing or using any of the information contained in
>> the
>> >>>>> transmission. If you received this communication in error, please
>> >>>>> contact
>> >>>>> the sender immediately and destroy the material in its entirety,
>> >>>>> whether
>> >>>>> electronic or hard copy. This communication may contain nonpublic
>> >>>>> personal
>> >>>>> information about consumers subject to the restrictions of the
>> >>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
>> >>>>> directly or
>> >>>>> indirectly reuse or disclose such information for any purpose other
>> >>>>> than to
>> >>>>> provide the services for which you are receiving the information.
>> >>>>> There are risks associated with the use of electronic transmission.
>> >>>>> The
>> >>>>> sender of this information does not control the method of
>> transmittal
>> >>>>> or
>> >>>>> service providers and assumes no duty or obligation for the
>> security,
>> >>>>> receipt, or third party interception of this transmission.
>> >>>>> ************************************
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> cisco-voip mailing list
>> >>>>> cisco-voip[at]puck.nether.net
>> >>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>>
>> >>>> Cheers,
>> >>>>
>> >>>> Tim
>> >>>>
>> >>>>
>> >>>> Sent from Sydney, Nsw, Australia
>> >>>
>> >>>
>> >>> --
>> >>>
>> >>> Cheers,
>> >>>
>> >>> Tim
>> >>>
>> >>>
>> >>> Sent from Sydney, Nsw, Australia
>> >>>
>> >>> ________________________________
>> >>>
>> >>> _______________________________________________
>> >>> cisco-voip mailing list
>> >>> cisco-voip[at]puck.nether.net
>> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
>> >>
>> >>
>> >> --
>> >>
>> >> Cheers,
>> >>
>> >> Tim
>> >>
>> >>
>> >> Sent from Sydney, Nsw, Australia
>> >> _______________________________________________
>> >> cisco-voip mailing list
>> >> cisco-voip[at]puck.nether.net
>> >> https://puck.nether.net/mailman/listinfo/cisco-voip
>> >>
>> >>
>> >
>> >
>> > --
>> >
>> > Cheers,
>> >
>> > Tim
>> >
>> >
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip[at]puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>


thsglobal at gmail

Nov 4, 2009, 2:07 PM

Post #16 of 17 (210 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

Thanks Nick, that is really great info!

On Thu, Nov 5, 2009 at 1:57 AM, Nick Matthews <matthnick[at]gmail.com> wrote:

> I think there are a few different factors - but it's the protocol I
> would use if I was administering my network.
>
> We see a lot of SIP gateways, and it's definitely being deployed.
>
> Some of the advantages:
> -Easy to troubleshoot. You can read up on SIP and learn the basics
> 2-3x faster than other protocols. It's clear and concise for the most
> part.
> -Interop. Most of the new devices coming out are all running SIP.
> You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
> experience with it already.
> -Easier transition to SIP as your PSTN connection (last post) if/when
> you decide to make that jump.
> -If you're already running H323, switching over is pretty easy.
>
> Other considerations:
> -H323 is still the best at video, and for a while, there doesn't
> appear to be any real alternatives.
> -MGCP is still the only 'centralized dial plan' protocol where you
> don't have to do anything on your gateways at all. If you're not good
> with IOS and just 'want it to work', this is still the protocol to
> look at. It comes with it's own troubles, bugs, and instability
> because of it.
> -Some older devices don't support SIP yet, and you may still be
> running H323 in the network anyways.
> -For more advanced call flows and designs, you may run into some
> unsupported features. (Like using ANN for ringback, I think that is
> still H323 only).
> -I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
> If you have older platforms like the 3700, CMM, that won't run 20.T, I
> would stick to your existing protocol. Likewise for CUCM versions
> prior to 6.x. The SIP stacks in the versions prior just aren't as
> stable or have as many features.
>
>
> -nick
>
>
> On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob[at]gmail.com> wrote:
> > Nick that is what I am asking. I in no way want to go with a SIP trunk
> to
> > the PSTN I just want to use SIP as my gateway protocol. So the Telco
> still
> > hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use
> > SIP. As far as why drop H.323 I don’t have a reason to but when doing new
> > customer deployments I don’t want to put one thing in and then migrate to
> > something else two years down the road.
> >
> >
> >
> > So I ask my question again has anyone used SIP as their GW protocol
> instead
> > of H.323? Any problems or things I should look for? Should I just not do
> it
> > yet.
> >
> >
> >
> > From: cisco-voip-bounces[at]puck.nether.net
> > [mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
> > Sent: Tuesday, November 03, 2009 10:07 PM
> > To: Nick Matthews
> > Cc: CiscosupportUpuck
> > Subject: Re: [cisco-voip] SIP as a gateway Protocol
> >
> >
> >
> > Hi Nick,
> >
> >
> >
> > What about using SIP just as protocol to replace H323 / MGCP between CCM
> and
> > your Voice Gateway?
> >
> >
> >
> > Cheers,
> >
> >
> >
> > Tim
> >
> > On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com>
> wrote:
> >
> > You can get an over-the-top SIP provider, but if you get voice quality
> > problems you'll have some trouble getting your ISP and SIP provider to
> > play nicely. Once it leaves your gateway you can't prove who may be
> > causing the problem if there is jitter or packet loss. Your ISP
> > probably won't have any idea how to deal with it, because for
> > traditional data these types of packet problems do not have much
> > consequence.
> >
> > If you're cool with that, there are hundreds of providers of varying
> > quality.
> >
> > The suggestion is still to go with the data line from the SIP
> > provider. You may be able to save some money on equipment
> > consolidation or pricing depending on your volume / area as well.
> > It's not the best scenario for every case, but there are certainly
> > cases where it makes since and these cases are growing.
> >
> >
> > -nick
> >
> > On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com> wrote:
> >> We dont have too many SIP providers here in Oz at the moment anyway.
> >> We were talking about just using SIP between CCM and the Gateway. Vs
> MGCP
> >> and H323.
> >>
> >> Fax / modem could definitely be a good point though.
> >>
> >> Cheers,
> >>
> >> Tim.
> >>
> >> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca>
> wrote:
> >>>
> >>> From our initial conversations with our PSTN providers, SIP was a few
> >>> years away with feature parity with H323/MGCP/PRI trunks.
> >>>
> >>> FAX support was definately out of the question, and there were crazy
> >>> requirements about not being able to do voice only on the ethernet
> trunk.
> >>> We
> >>> had to buy a data package that was no more than 50% voice traffic. For
> >>> us,
> >>> we get our internet through our regional network at dirt cheap prices
> >>> because we basically run a co-op. For others it might make sense to
> move
> >>> to
> >>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our
> backup
> >>> internet link is cheaper than the PSTN provider could price I believe.
> >>>
> >>> The other thing was route diversity and multiple demarcs. I think those
> >>> were quite expensive where as now, we get it at no extra cost.
> >>>
> >>> I've long been a proponent of if it ain't broke, don't fix it. Even
> when
> >>> we went to tender and ended up switching our PRIs to another local
> >>> carrier,
> >>> it was a LOT of work. I understood it saved us quite a bit of money, so
> >>> it
> >>> was worth it in the end for a three year contract. That being said,
> don't
> >>> expect that SIP will be cheaper than PRIs and/or without it's own
> >>> problems.
> >>>
> >>> Caveat Emptor as my friend Caesar said.
> >>>
> >>>
> >>> ----- Original Message -----
> >>> From: Tim Smith
> >>> To: STEVEN CASPER
> >>> Cc: CiscosupportUpuck
> >>> Sent: Tuesday, November 03, 2009 8:46 PM
> >>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
> >>> Also, SIP is slightly easier to troubleshoot than H323, much more so
> than
> >>> MGCP. (And I also dont like MGCP anyway :)
> >>>
> >>> Cheers,
> >>>
> >>> Tim.
> >>>
> >>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com>
> wrote:
> >>>>
> >>>> I like the idea.
> >>>>
> >>>> More and more SIP trunks will be turning up. Why bother having to go
> >>>> from
> >>>> H323 to SIP. Simpler just to run SIP.
> >>>>
> >>>> I also like SIP and how you can set it up to monitor the destination
> of
> >>>> your dial-peers. Shut them down if a CCM is down.
> >>>>
> >>>> Cheers,
> >>>>
> >>>> Tim
> >>>>
> >>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com>
> wrote:
> >>>>>
> >>>>> I assume you are talking traditional analog and digital PSTN
> >>>>> gateways, why are you considering migrating to SIP to control these
> as
> >>>>> opposed to H323? .
> >>>>>
> >>>>> Steve
> >>>>>
> >>>>> >>> Voice Noob <voicenoob[at]gmail.com> 11/3/2009 6:09 PM >>>
> >>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
> >>>>> instead of H.323 or MGCP and start migrating it to SIP on the
> gateway.
> >>>>> Any
> >>>>> experience with this? Can I get Calling Name and Number from the PSTN
> >>>>> side?
> >>>>>
> >>>>> ************************************
> >>>>> This email may contain privileged and/or confidential information
> that
> >>>>> is intended solely for the use of the addressee. If you are not the
> >>>>> intended recipient or entity, you are strictly prohibited from
> >>>>> disclosing,
> >>>>> copying, distributing or using any of the information contained in
> the
> >>>>> transmission. If you received this communication in error, please
> >>>>> contact
> >>>>> the sender immediately and destroy the material in its entirety,
> >>>>> whether
> >>>>> electronic or hard copy. This communication may contain nonpublic
> >>>>> personal
> >>>>> information about consumers subject to the restrictions of the
> >>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
> >>>>> directly or
> >>>>> indirectly reuse or disclose such information for any purpose other
> >>>>> than to
> >>>>> provide the services for which you are receiving the information.
> >>>>> There are risks associated with the use of electronic transmission.
> >>>>> The
> >>>>> sender of this information does not control the method of transmittal
> >>>>> or
> >>>>> service providers and assumes no duty or obligation for the security,
> >>>>> receipt, or third party interception of this transmission.
> >>>>> ************************************
> >>>>>
> >>>>> _______________________________________________
> >>>>> cisco-voip mailing list
> >>>>> cisco-voip[at]puck.nether.net
> >>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>>
> >>>> Cheers,
> >>>>
> >>>> Tim
> >>>>
> >>>>
> >>>> Sent from Sydney, Nsw, Australia
> >>>
> >>>
> >>> --
> >>>
> >>> Cheers,
> >>>
> >>> Tim
> >>>
> >>>
> >>> Sent from Sydney, Nsw, Australia
> >>>
> >>> ________________________________
> >>>
> >>> _______________________________________________
> >>> cisco-voip mailing list
> >>> cisco-voip[at]puck.nether.net
> >>> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
> >>
> >> --
> >>
> >> Cheers,
> >>
> >> Tim
> >>
> >>
> >> Sent from Sydney, Nsw, Australia
> >> _______________________________________________
> >> cisco-voip mailing list
> >> cisco-voip[at]puck.nether.net
> >> https://puck.nether.net/mailman/listinfo/cisco-voip
> >>
> >>
> >
> >
> > --
> >
> > Cheers,
> >
> > Tim
> >
> >
>



--

Cheers,

Tim


Sent from Sydney, Nsw, Australia


Dennis.Heim at cdw

Nov 4, 2009, 7:02 PM

Post #17 of 17 (202 views)
Permalink
Re: SIP as a gateway Protocol [In reply to]

SIP is like pre-draft 802.11n. Until SIP progresses back each vendor bucket of RFCs that they choose to support then it can be a standard based protocol. But for now there is Nortel SIP, Cisco SIP.

I wish cisco would allow some of the POTS commands on the VOIP dial-peers such as forward digits etc. I know you can do it all with translation profiles/reg-ex but typing a command that resembles English would be appreciated.

Dennis Heim
Network Voice Engineer
CDW Advanced Technology Services
11711 N. Meridian Street, Suite 225
Carmel, IN 46032

317.569.4255 Office
317.569.4201 Fax
317.694.6070 Cell
dennis.heim[at]cdw.com<mailto:dennis.heim[at]cdw.com>
www.berbee.com<http://www.berbee.com/>

From: cisco-voip-bounces[at]puck.nether.net [mailto:cisco-voip-bounces[at]puck.nether.net] On Behalf Of Tim Smith
Sent: Wednesday, November 04, 2009 5:08 PM
To: Nick Matthews
Cc: CiscosupportUpuck
Subject: Re: [cisco-voip] SIP as a gateway Protocol

Thanks Nick, that is really great info!
On Thu, Nov 5, 2009 at 1:57 AM, Nick Matthews <matthnick[at]gmail.com<mailto:matthnick[at]gmail.com>> wrote:
I think there are a few different factors - but it's the protocol I
would use if I was administering my network.

We see a lot of SIP gateways, and it's definitely being deployed.

Some of the advantages:
-Easy to troubleshoot. You can read up on SIP and learn the basics
2-3x faster than other protocols. It's clear and concise for the most
part.
-Interop. Most of the new devices coming out are all running SIP.
You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
experience with it already.
-Easier transition to SIP as your PSTN connection (last post) if/when
you decide to make that jump.
-If you're already running H323, switching over is pretty easy.

Other considerations:
-H323 is still the best at video, and for a while, there doesn't
appear to be any real alternatives.
-MGCP is still the only 'centralized dial plan' protocol where you
don't have to do anything on your gateways at all. If you're not good
with IOS and just 'want it to work', this is still the protocol to
look at. It comes with it's own troubles, bugs, and instability
because of it.
-Some older devices don't support SIP yet, and you may still be
running H323 in the network anyways.
-For more advanced call flows and designs, you may run into some
unsupported features. (Like using ANN for ringback, I think that is
still H323 only).
-I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
If you have older platforms like the 3700, CMM, that won't run 20.T, I
would stick to your existing protocol. Likewise for CUCM versions
prior to 6.x. The SIP stacks in the versions prior just aren't as
stable or have as many features.


-nick


On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob[at]gmail.com<mailto:voicenoob[at]gmail.com>> wrote:
> Nick that is what I am asking. I in no way want to go with a SIP trunk to
> the PSTN I just want to use SIP as my gateway protocol. So the Telco still
> hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use
> SIP. As far as why drop H.323 I don't have a reason to but when doing new
> customer deployments I don't want to put one thing in and then migrate to
> something else two years down the road.
>
>
>
> So I ask my question again has anyone used SIP as their GW protocol instead
> of H.323? Any problems or things I should look for? Should I just not do it
> yet.
>
>
>
> From: cisco-voip-bounces[at]puck.nether.net<mailto:cisco-voip-bounces[at]puck.nether.net>
> [mailto:cisco-voip-bounces[at]puck.nether.net<mailto:cisco-voip-bounces[at]puck.nether.net>] On Behalf Of Tim Smith
> Sent: Tuesday, November 03, 2009 10:07 PM
> To: Nick Matthews
> Cc: CiscosupportUpuck
> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>
>
>
> Hi Nick,
>
>
>
> What about using SIP just as protocol to replace H323 / MGCP between CCM and
> your Voice Gateway?
>
>
>
> Cheers,
>
>
>
> Tim
>
> On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick[at]gmail.com<mailto:matthnick[at]gmail.com>> wrote:
>
> You can get an over-the-top SIP provider, but if you get voice quality
> problems you'll have some trouble getting your ISP and SIP provider to
> play nicely. Once it leaves your gateway you can't prove who may be
> causing the problem if there is jitter or packet loss. Your ISP
> probably won't have any idea how to deal with it, because for
> traditional data these types of packet problems do not have much
> consequence.
>
> If you're cool with that, there are hundreds of providers of varying
> quality.
>
> The suggestion is still to go with the data line from the SIP
> provider. You may be able to save some money on equipment
> consolidation or pricing depending on your volume / area as well.
> It's not the best scenario for every case, but there are certainly
> cases where it makes since and these cases are growing.
>
>
> -nick
>
> On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal[at]gmail.com<mailto:thsglobal[at]gmail.com>> wrote:
>> We dont have too many SIP providers here in Oz at the moment anyway.
>> We were talking about just using SIP between CCM and the Gateway. Vs MGCP
>> and H323.
>>
>> Fax / modem could definitely be a good point though.
>>
>> Cheers,
>>
>> Tim.
>>
>> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio[at]uoguelph.ca<mailto:lelio[at]uoguelph.ca>> wrote:
>>>
>>> From our initial conversations with our PSTN providers, SIP was a few
>>> years away with feature parity with H323/MGCP/PRI trunks.
>>>
>>> FAX support was definately out of the question, and there were crazy
>>> requirements about not being able to do voice only on the ethernet trunk.
>>> We
>>> had to buy a data package that was no more than 50% voice traffic. For
>>> us,
>>> we get our internet through our regional network at dirt cheap prices
>>> because we basically run a co-op. For others it might make sense to move
>>> to
>>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup
>>> internet link is cheaper than the PSTN provider could price I believe.
>>>
>>> The other thing was route diversity and multiple demarcs. I think those
>>> were quite expensive where as now, we get it at no extra cost.
>>>
>>> I've long been a proponent of if it ain't broke, don't fix it. Even when
>>> we went to tender and ended up switching our PRIs to another local
>>> carrier,
>>> it was a LOT of work. I understood it saved us quite a bit of money, so
>>> it
>>> was worth it in the end for a three year contract. That being said, don't
>>> expect that SIP will be cheaper than PRIs and/or without it's own
>>> problems.
>>>
>>> Caveat Emptor as my friend Caesar said.
>>>
>>>
>>> ----- Original Message -----
>>> From: Tim Smith
>>> To: STEVEN CASPER
>>> Cc: CiscosupportUpuck
>>> Sent: Tuesday, November 03, 2009 8:46 PM
>>> Subject: Re: [cisco-voip] SIP as a gateway Protocol
>>> Also, SIP is slightly easier to troubleshoot than H323, much more so than
>>> MGCP. (And I also dont like MGCP anyway :)
>>>
>>> Cheers,
>>>
>>> Tim.
>>>
>>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal[at]gmail.com<mailto:thsglobal[at]gmail.com>> wrote:
>>>>
>>>> I like the idea.
>>>>
>>>> More and more SIP trunks will be turning up. Why bother having to go
>>>> from
>>>> H323 to SIP. Simpler just to run SIP.
>>>>
>>>> I also like SIP and how you can set it up to monitor the destination of
>>>> your dial-peers. Shut them down if a CCM is down.
>>>>
>>>> Cheers,
>>>>
>>>> Tim
>>>>
>>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER[at]mtb.com<mailto:SCASPER[at]mtb.com>> wrote:
>>>>>
>>>>> I assume you are talking traditional analog and digital PSTN
>>>>> gateways, why are you considering migrating to SIP to control these as
>>>>> opposed to H323? .
>>>>>
>>>>> Steve
>>>>>
>>>>> >>> Voice Noob <voicenoob[at]gmail.com<mailto:voicenoob[at]gmail.com>> 11/3/2009 6:09 PM >>>
>>>>> Has anyone started using SIP on the PSTN gateway? I want to use it
>>>>> instead of H.323 or MGCP and start migrating it to SIP on the gateway.
>>>>> Any
>>>>> experience with this? Can I get Calling Name and Number from the PSTN
>>>>> side?
>>>>>
>>>>> ************************************
>>>>> This email may contain privileged and/or confidential information that
>>>>> is intended solely for the use of the addressee. If you are not the
>>>>> intended recipient or entity, you are strictly prohibited from
>>>>> disclosing,
>>>>> copying, distributing or using any of the information contained in the
>>>>> transmission. If you received this communication in error, please
>>>>> contact
>>>>> the sender immediately and destroy the material in its entirety,
>>>>> whether
>>>>> electronic or hard copy. This communication may contain nonpublic
>>>>> personal
>>>>> information about consumers subject to the restrictions of the
>>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not
>>>>> directly or
>>>>> indirectly reuse or disclose such information for any purpose other
>>>>> than to
>>>>> provide the services for which you are receiving the information.
>>>>> There are risks associated with the use of electronic transmission.
>>>>> The
>>>>> sender of this information does not control the method of transmittal
>>>>> or
>>>>> service providers and assumes no duty or obligation for the security,
>>>>> receipt, or third party interception of this transmission.
>>>>> ************************************
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip[at]puck.nether.net<mailto:cisco-voip[at]puck.nether.net>
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Cheers,
>>>>
>>>> Tim
>>>>
>>>>
>>>> Sent from Sydney, Nsw, Australia
>>>
>>>
>>> --
>>>
>>> Cheers,
>>>
>>> Tim
>>>
>>>
>>> Sent from Sydney, Nsw, Australia
>>>
>>> ________________________________
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip[at]puck.nether.net<mailto:cisco-voip[at]puck.nether.net>
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>> --
>>
>> Cheers,
>>
>> Tim
>>
>>
>> Sent from Sydney, Nsw, Australia
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip[at]puck.nether.net<mailto:cisco-voip[at]puck.nether.net>
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
>
> --
>
> Cheers,
>
> Tim
>
>



--

Cheers,

Tim


Sent from Sydney, Nsw, Australia

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