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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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