Am 16.10.2014 um 23:25 schrieb Michael Knill 
<[email protected]>:

> Thanks Michael. That sounds perfect.
> I am thinking though that the SIP trunk does not really need to go through 
> the VPN e.g. no NAT problems as its on the external Astlinux addresses.
> The only traffic going through the VPN is the console extension traffic (and 
> BLF).
> 
> What do you think?
> 
> Regards
> Michael Knill

For me it works perfectly fine with the SIP trunk going through the VPN tunnel.

> On 17 Oct 2014, at 8:01 am, Michael Keuter <[email protected]> wrote:
> 
> 
> Am 16.10.2014 um 22:20 schrieb Michael Knill 
> <[email protected]>:
> 
>> Hi group
>> 
>> I have been struggling to work out how to set up a multisite company with 
>> Astlinux.
>> Basically they want to have a central answering point but essentially all 
>> sites are autonomous e.g. each site has their own SIP service which is 
>> routed to the main site but used for outgoing calls from that site. I have 
>> SIP trunks set up between them and thats working fine however as their 
>> Internet connections are slow, I cannot afford to have hair pinning of calls 
>> or at least one hop only.
>> 
>> I first tried routing via the SIP trunk, but when the call is transferred 
>> (usually back to the original site) the call is hair pinned all the way to 
>> the Head Office and back. direct media does not fix the problem either.
>> I then though that I could remotely register extensions to each site on the 
>> head office console phone so the transfer is local to the remote system. 
>> Good thought but Im having lots of trouble having both a SIP trunk and a 
>> passthrough SIP extension on the same box.
>> My next thought was using the above idea but getting rid of the NAT problem 
>> with a VPN.
>> 
>> Does anyone have any other ideas. Im losing my hair.
>> 
>> Regards
>> Michael Knill
> 
> Hi Michael,
> 
> I have a customer with a similar setup: one main office with 4 branch 
> offices. Every site has its own SIP trunk to an VoIP provider for outgoing 
> calls.
> Most incoming calls are going through the main office with has 3 queues. Each 
> branch office has an OpenVPN client connection to the main office and a SIP 
> trunk to the main office.
> The important point is: the host (IP) of the main office SIP trunk in the 
> branch offices needs to be the virtual VPN IP address (e.g. 10.8.x.1) not the 
> LAN address!
> 
> All SIP endpoints (of the branch offices) can login as agents into each of 
> the 3 queues of the main office. That works fine in combination with FOP2. 
> Prosody is needed for distributed device status (if needed).
> I never had any routing or NAT issue.
> 
> Michael
> 
> http://www.mksolutions.info
> 
> 
> 
> 
> 
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> 
> 
> ------------------------------------------------------------------------------
> Comprehensive Server Monitoring with Site24x7.
> Monitor 10 servers for $9/Month.
> Get alerted through email, SMS, voice calls or mobile push notifications.
> Take corrective actions from your mobile device.
> http://p.sf.net/sfu/Zoho
> _______________________________________________
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> https://lists.sourceforge.net/lists/listinfo/astlinux-users
> 
> Donations to support AstLinux are graciously accepted via PayPal to 
> [email protected].


Michael

http://www.mksolutions.info





------------------------------------------------------------------------------
Comprehensive Server Monitoring with Site24x7.
Monitor 10 servers for $9/Month.
Get alerted through email, SMS, voice calls or mobile push notifications.
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http://p.sf.net/sfu/Zoho
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