The main reason is that we have a couple of phones at remote locations 
connected via VPN. Since we are in a rural area with slow ADSL1 as our only 
Internet connection option, having the audio from these offices come in through 
the VPN and then back out to the ITSP is less than ideal. With QOS on the links 
quality is okay but latency is terrible and our bandwidth is very limited.

Regards,
Andrew Radke

On 01/12/2010, at 7:40 PM, "Nikolay Kondratyev" <[email protected]> wrote:

> "Proxy authentication" with itsp for outgoing calls is also performed by 
> sipxbridge.
> That is, it will not work via 'unmanaged gw'.
> By the way, why don't you like media relay? Is there any real problem because 
> of that?
> Just in case it concerns you:
> If there is no NAT between you and ITSP you can configure your siptrunk gw to 
> use internal ip address (uncheck "use public address for call setup" 
> checkbox, which is checked by default).
> But media relay will be used anyway.
> Rgds,
> Nikolay.
> 
>  
> From: [email protected] 
> [mailto:[email protected]] On Behalf Of Andrew Radke
> Sent: Wednesday, December 01, 2010 12:21 PM
> To: Discussion list for users of sipXecs software
> Subject: Re: [sipx-users] SipX 4.2 media relay
> 
> At this stage we don't need to register since we don't accept any incoming 
> calls from our ITSP, but we do need to authenticate outgoing calls via them.
> 
> Regards,
> Andrew Radke
> 
> On 01/12/2010, at 7:00 PM, "Nikolay Kondratyev" <[email protected]> wrote:
> 
>> You mean register to ITSP?
>> No, it can't (afaik).
>> Rgds,
>> Nikolay.
>> 
>> From: [email protected] 
>> [mailto:[email protected]] On Behalf Of Andrew Radke
>> Sent: Wednesday, December 01, 2010 11:23 AM
>> To: Discussion list for users of sipXecs software
>> Subject: Re: [sipx-users] SipX 4.2 media relay
>> 
>> Can an unmanaged gateway authenticate with an ITSP though? Sorry, I didn't 
>> even look in 4.2 and I've left work for the day now.
>> 
>> Regards,
>> Andrew Radke
>> 
>> On 01/12/2010, at 4:37 PM, "Nikolay Kondratyev" <[email protected]> wrote:
>> 
>>> Andrew,
>>> do you use "sip trunk" to connect to ITSP?
>>> my understanding is that, when you use siptrunk (sipxbridge), you get the 
>>> following:
>>> 1. Refer messages are converted to re-Invite.
>>> 2. media relay
>>> 3. shortened headers (not so many via's).
>>> And you can not make sipxbriddge not to do it.
>>> If you really want your media streams go directly from the phones to ITSP, 
>>> just use "unmanaged" gaeway.
>>> Rgds,
>>> Nikolay. 
>>>  
>>> 
>>> From: [email protected] 
>>> [mailto:[email protected]] On Behalf Of Andrew Radke
>>> Sent: Wednesday, December 01, 2010 9:07 AM
>>> To: [email protected]
>>> Subject: [sipx-users] SipX 4.2 media relay
>>> 
>>> Hi all,
>>> 
>>> Something that I have noticed in 4.2 is that calls made via an ITSP have 
>>> their audio routed via the SipX media relay. I have been looking at how to 
>>> turn this off by reading previous discussions on this list but it seems 
>>> that since SipX notices that there is a NATing firewall between it and the 
>>> ITSP it will relay the media anyway.
>>> 
>>> Under Internet Calling I have the correct intranet subnets defined and have 
>>> tried all combinations of "Enable NAT Traversal" and "Server behind NAT". 
>>> The notes on the right of this screen states: "Whenever it is detected that 
>>> the two endpoints that are starting a media session are separated by one or 
>>> more NATs, the server will insert its Media Relay into the media path." 
>>> which seems to define my issue.
>>> 
>>> Is it possible to disable this behaviour?
>>> 
>>> Regards,
>>> Andrew Radke
>>> Yuruga Nursery Pty Ltd
>>> Clonal Solutions Australia Pty Ltd
>>> PO Box 220
>>> Walkamin Qld 4872
>>> Phone: (07) 4093 3826
>>> Fax: (07) 4093 3869
>>> Email: [email protected]
>>> Web: www.yuruga.com.au
>>> 
>> 
>>> _______________________________________________
>>> sipx-users mailing list
>>> [email protected]
>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/
> 
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