[asterisk-users] sip.conf for talking to other Asterisk machines

Forrest Beck jonforrest.beck at gmail.com
Mon Sep 18 18:51:51 MST 2006


I use two user's per host one for user and the other peer.  Sort of
like attahed.

I also prefer IAX for communication between asterisk boxes.  IAX use's
less bandwidth than SIP and it's trunks are alot smaller.  If you look
at SIP traffic, 80% of it is headers.  The headers look just like smtp
headers.

Even if your clients are using SIP to communicate to asterisk using
SIP, the asterisk servers will maintain the trunked connection route
the traffic for your SIP phones.

On 9/18/06, Bill Gibbs <bgibbs at edurotech.com> wrote:
>
>
>
>
> Just curious how most of you are defining SIP peers in sip.conf – for
> Asterisk boxes talking to each other.  Are most of you just making a
> type=friend connection and a single context or are you separating them out
> to in/out definitions and contexts?
>
>
>
> In other words
>
> Where voicegw1 is the Asterisk box with the TDM cards for talking to the
> PSTN, it will receive calls from the PSTN and forward to the appropriate
> Asterisk box as well as receive calls from the other Asterisk boxes to
> forward out to the PSTN.
>
>
>
> Do you on the Asterisk box that contains all the SIP phones define (ie the
> client to the PSTN Asterisk box and voicegw1 is the one with the PSTN
> connection)
>
> [voicegw1-in]
>
> type=user
>
> username=virtualpbx1-in
>
> secret=1234
>
> host=192.168.1.99
>
> context=voicegw1-in
>
> canreinvite=no
>
> nat=no
>
> qualify=yes
>
> allow=all
>
>
>
> [voicegw1-out]
>
> type=peer
>
> username=virtualpbx1-out
>
> secret=1234
>
> host=192.168.1.99
>
> context=voicegw1-out
>
> canreinvite=no
>
> nat=no
>
> qualify=yes
>
> allow=all
>
>
>
> or
>
>
>
> [voicegw1]
>
> Type=friend
>
> Blah
>
> Context=voicegw1
>
>
>
> And use a single context for inbound/outbound routing?
>
>
>
> The same would apply to the PSTN Asterisk server.
>
>
>
>
>
> Bill
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