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

[Douglas Garstang]

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:jonforrest.beck at gmail.com] 
	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 <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
	> _______________________________________________
	> --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
	>
	>
	>