IAX has some pretty severe limitations when it comes to trunking calls between
Asterisk boxes. It can't pass variables for example, and any calls to SIP
phones at the far end will be treated as IAX calls, which is just nuts. This
means you lose a lot of SIP features, like transferring and forwarding. We had
to drop IAX and go back to SIP, which is pretty ironic considering IAX stands
for Inter Asterisk Exchange.
-----Original Message-----
From: Forrest Beck [mailto:[EMAIL PROTECTED]
Sent: Mon 9/18/2006 7:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] sip.conf for talking to other Asterisk
machines
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 <[EMAIL PROTECTED]> 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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>
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