Understood! and lot of work to be honest.
My question now is: which polycom file needs to be tweacked and what
parameter?

Thank you
Saad

On Tue, Aug 21, 2012 at 10:30 AM, Tony Graziano <
[email protected]> wrote:

> You can either nat this to a firewall (FTP) to sipx or pass it through the
> SBC to sipx (FTP). As long as the SBC sees ANY traffic from a UA, it should
> pass it through to sipx (essentially pass all traffic on one interface to
> sipx, leaving the other to trunking FROM sipx as an unmanaged gateway).
>
> i.e.
>
> Interface_ITSP - unmanaged gateway and handles all trunking in to ITSP and
> sipx simply sends calls there as a result of its own dialplans.
> Interface_USERS - All user/registrations, outbound calls, etc. get passed
> to sipx from the UA. Inbound calls come from Interface_ITSP and get sent to
> the registered UA accordingly.
>
> In your case split DNS may not help since everything is actually private.
>
> If your clients (UA's do not traverse a firewall) then you can simply use
> the Server Menu on the handset to input the TFTP address there, as an
> example, but if no nat is involed you should be able to send that via dhcp
> and not get involved in configuring the phones manually.
>
> On Tue, Aug 21, 2012 at 10:13 AM, Saad <[email protected]> wrote:
>
>> So in this case we cannot rely on SIPX as a file server but we need to
>> setup an external TFTP/FTP correct? Could you please advise the file name
>> that need to be changed for the DNS SRV ?
>>
>>
>> On Tue, Aug 21, 2012 at 10:02 AM, Tony Graziano <
>> [email protected]> wrote:
>>
>>> Nothing should be changed on sipx or in the phones from a default
>>> installation, at least that has been my experience with other SBC's.
>>>
>>> Perhaps you need a dialplan rule in the Acme SBC to to pass the MOH uri
>>> from it to sipx?
>>>
>>> On Tue, Aug 21, 2012 at 9:27 AM, Saad <[email protected]> wrote:
>>>
>>>> You are absolutely right as I have learned this the hard way.. The
>>>> outbound calls experienced many troubles when sent via the same SBC
>>>> interface that the phones registration coming from. Internal calls stopped
>>>> to work ( SIPX didn't like to have the unmanaged GW talking to the same
>>>> interface that all phones showing registered with)..... What I did as a
>>>> work around was created a separate interface on the SBC to talk to the
>>>> unmanaged GW on SIPX and this worked fine.
>>>>
>>>> I believe, I will open a ticket with ACME dig further on the MOH
>>>> treatment with their NN3820.
>>>>
>>>> Just to recap, are you saying that nothing at all needs to be changed
>>>> on SIPX ( other than using unmanaged GW) ? should everything remain as is?
>>>>
>>>>
>>>>
>>>> Thank you
>>>>
>>>> Saad
>>>>
>>>>
>>>> On Tue, Aug 21, 2012 at 9:01 AM, Tony Graziano <
>>>> [email protected]> wrote:
>>>>
>>>>> Assuming the phone is remote:
>>>>>
>>>>> The DNS SRV records need to point to the public IP of the SBC. The SBC
>>>>> needs to see the registration and pass it through to the sipx server. All
>>>>> registered phones should pass through to the sipx server, period.
>>>>>
>>>>> Outbound calls should not be to the SBC, they should be to the sipx
>>>>> server (otherwise you have a security management headache with call
>>>>> permissions). The SBC should see the phone as registered and send all
>>>>> communications to the sipx server and let the sipx server handle all
>>>>> registrations, permissions, etc. Outbound calls should be sent from sipx 
>>>>> to
>>>>> the SBC via an unmanaged gateway/dialplan entry(ies).
>>>>>
>>>>> The most consistent way to make this happen is to:
>>>>>
>>>>> 1. Have an SBC that can perform these functions.
>>>>> 2. Use two different interfaces in the SBC (One facing the public
>>>>> Internet/Where the clients are registering from), and another facing the
>>>>>  ITSP.
>>>>>
>>>>> In this way MOH is a function of an ongoing media transaction with the
>>>>> sipx server, it has always worked for me with other SBC's (Ingate, as an
>>>>> example, you can specify the sipx MOH uri so it can utilize it), but MOH 
>>>>> is
>>>>> tricky even for SBC's which is why other (like Karoo Bridge) allow you to
>>>>> specify it's own MOH as a workaround.
>>>>>
>>>>> Long story short, MOH may not work depending on the SBC in use, but it
>>>>> is important to be able to segregate traffic for users versus trunking in
>>>>> order to make the routing logic in the SBC work appropriately.
>>>>> On Tue, Aug 21, 2012 at 8:44 AM, Saad <[email protected]> wrote:
>>>>>
>>>>>> Hello All,
>>>>>>
>>>>>> I know many of you succeeded with registering phones VIA SBC. Are you
>>>>>> aware of any SIPX documentation on the proper configuration to have 
>>>>>> Polycom
>>>>>> phones to register via SBC?
>>>>>>
>>>>>> From the SBC side, the configuration was completed; however, still
>>>>>> not sure if we are doing it the right way on SIPX... What we did on SIPX
>>>>>> was to point the outbound proxy (Phone registration form)  to the SBC
>>>>>> (Registration point) and this worked with some minor intermittent 
>>>>>> troubles
>>>>>> like (Silent on MOH, attended transfer failure.), if we point both the
>>>>>> outbound proxy and the primary registration server to the SBC, then the
>>>>>> unmanaged GW Caller ID doesn't override the users numbers , MWI update
>>>>>> stops working, the attended AND Unattended transfer failed to work...
>>>>>>
>>>>>>
>>>>>> Please let me know if there is any documents that explains how to
>>>>>> work the phone registration via SBC on SIPX / IP Phones sides..
>>>>>>
>>>>>> Thank you
>>>>>> Saad Khankan, P.Eng.
>>>>>>
>>>>>> _______________________________________________
>>>>>> sipx-users mailing list
>>>>>> [email protected]
>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> ~~~~~~~~~~~~~~~~~~
>>>>> Tony Graziano, Manager
>>>>> Telephone: 434.984.8430
>>>>> sip: [email protected]
>>>>> Fax: 434.465.6833
>>>>> ~~~~~~~~~~~~~~~~~~
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>>>>> Ask about our Internet Fax services!
>>>>> ~~~~~~~~~~~~~~~~~~
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>>>>>
>>>>>
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>>>>
>>>>
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>>>
>>>
>>>
>>> --
>>> ~~~~~~~~~~~~~~~~~~
>>> Tony Graziano, Manager
>>> Telephone: 434.984.8430
>>> sip: [email protected]
>>> Fax: 434.465.6833
>>> ~~~~~~~~~~~~~~~~~~
>>> Linked-In Profile:
>>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
>>> Ask about our Internet Fax services!
>>> ~~~~~~~~~~~~~~~~~~
>>>
>>> Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
>>> 2013!
>>>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>>>
>>>
>>> LAN/Telephony/Security and Control Systems Helpdesk:
>>> Telephone: 434.984.8426
>>> sip: [email protected].**net<[email protected]>
>>>
>>> Helpdesk Customers: 
>>> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
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>>
>>
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>>
>
>
>
> --
> ~~~~~~~~~~~~~~~~~~
> Tony Graziano, Manager
> Telephone: 434.984.8430
> sip: [email protected]
> Fax: 434.465.6833
> ~~~~~~~~~~~~~~~~~~
> Linked-In Profile:
> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
> Ask about our Internet Fax services!
> ~~~~~~~~~~~~~~~~~~
>
> Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab
> 2013!
>  <http://sipxcolab2013.eventbrite.com/?discount=tony2013>
>
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> sip: [email protected].**net<[email protected]>
>
> Helpdesk Customers: 
> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net>
> Blog: http://blog.myitdepartment.net
>
> _______________________________________________
> sipx-users mailing list
> [email protected]
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>
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