asterisk users - Oct 2009 - Searching on how to keep local calls... local

Hi list.

Does anyone know how to keep calls between 2 local SIP-phones on the
local private network when the 2 local IP-phones are registered to an
online public Asterisk-server ??

What network-element / router do I need to install to prevent the
RTP-traffic from flowing via the internet ?

Config :

Asterisk --internet-- > router/firewall --> connected local IP-phones

Internal call :

IP-phone1 --> router/firewall --internet--> Asterisk --internet
(back)--> router/firewall (back) --> IP-phone2


So I don't want an Asterisk server in my company (don't have appropriate
place) and so I place the Asterisk-server in a datacentre. How about
local calls going via the internet and back ?!

Greetingz,
Jonas.
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Your best option without a local asterisk server is to set up the remote
server to do reinvites when calls are going local->local

The calls will end up routed through your internet router, but not beyond
that.

Downside: might have to make each ip phone available via port forwards

If you're really set against a local asterisk server, maybe try some other
sip proxy software running on a small embedded computer (wrt54gl nslu2 )

On Wed, Oct 21, 2009 at 2:47 AM, jonas kellens <jonas.kellens at
telenet.be>wrote:
>  Hi list.
>
> Does anyone know how to keep calls between 2 local SIP-phones on the local
> private network when the 2 local IP-phones are registered to an online
> public Asterisk-server ??
>
> What network-element / router do I need to install to prevent the
> RTP-traffic from flowing via the internet ?
>
> *Config :*
>
> Asterisk --internet-- > router/firewall --> connected local IP-phones
>
> *Internal call :*
>
> *IP-phone1* --> router/firewall --internet--> *Asterisk* --internet
> (back)--> router/firewall (back) --> *IP-phone2*
>
>
> So I don't want an Asterisk server in my company (don't have
appropriate
> place) and so I place the Asterisk-server in a datacentre. How about local
> calls going via the internet and back ?!
>
> Greetingz,
> Jonas.
>
> _______________________________________________
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>
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> To UNSUBSCRIBE or update options visit:
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> Your best option without a local asterisk server is to set up the
> remote server to do reinvites when calls are going local->local
> 
> The calls will end up routed through your internet router, but not
> beyond that.

So by placing "canreinvite=yes" in sip.conf, the RTP-traffic would
flow
between the 2 IP-phones and through the router.
Do I loose music on hold ? I guess I do...

> Downside: might have to make each ip phone available via port forwards

And if I place "nat=yes" in sip.conf ??
Or will IP-phone 1 not know the local IP-address of IP-phone 2 for
sending a re-invite ??


Jonas.
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>
> Your best option without a local asterisk server is to set up the remote
> server to do reinvites when calls are going local->local
>
> The calls will end up routed through your internet router, but not beyond
> that.
>
>
> So by placing "canreinvite=yes" in sip.conf, the RTP-traffic
would flow
> between the 2 IP-phones and through the router.
> Do I loose music on hold ? I guess I do...
>
Try it first, asterisk could just reinvite the audio back to the server
Also you might be able to program a SIP address for music on hold into the
ip phones

exten => moh,1,Answer()
exten => moh,2,MusicOnHold()

>
>  Downside: might have to make each ip phone available via port forwards
>
>
> And if I place "nat=yes" in sip.conf ??
> Or will IP-phone 1 not know the local IP-address of IP-phone 2 for sending
> a re-invite ??
>
The remote asterisk server would be doing the reinvites with what it knows
>
>
> Jonas.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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