I use the following in my Asterisk 1.2.4: [out-sipdiscount] type=peer secret=mypass username=myuser fromuser=myuser host=sip1.sipdiscount.com call-limit=1 disallow=all allow=ulaw allow=g726 allow=g729 allow=g723.1
Using this configuration, I am able to place calls using a Dial statement such as: Dial(SIP/[EMAIL PROTECTED]) HTH, Roshan http://roshan.info On Fri, Feb 17, 2006 at 12:31:17PM +0100, Alejandro Vargas scribbled: > 2006/2/17, Peter Bowyer <[EMAIL PROTECTED]>: > > A pretty simple setup works for me: > > The problem may be the username/password. But the page says this: > > SIP Discount offers the possibility to test our service right away, > for free! No need to sign up: just enter the account details below in > your favorite softphone or ATA and start calling! You can call all > destinations marked with * in our rate list . (Trial calls are limited > to a maximum duration of 1 minute). To enjoy unlimited calls, simply > sign up for SIP Discount. > > User Name: test > Password: test > Domain/Realm: sipdiscount.com > SIP Proxy/registrar: sip1.sipdiscount.com > SIP Outbound Proxy (optional): sip1.sipdiscount.com > STUN server (optional): stun.sipdiscount.com > > The only problem I say is asterisk is not sending stun.sipdiscount.com > or sipdiscount.com as domain. It is sending sip1.sipdiscount.com. > > -- > Alejandro Vargas > _______________________________________________ > --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 > _______________________________________________ --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
