EUREKA!!!! I finally solved this problem, I dont know why some of the more experienced people in here haven't answered this question (I guess they dont use Stanaphone but here it is) The problem isn't in how you register with Stanaphone but with the AMP config :( in the sip.conf folder is the line context = from-sip-external ; Send unknown SIP callers to this context what happens when it goes to this context is ;give external sip users congestion and hangup exten => _.,1,AbsoluteTimeout(15) exten => _.,2,Congestion exten => _.,3,Hangup I dont know why this is there when it isn't in the iax.conf folder - I'm assuming it is to stop random SIP calls. So to make Stanaphone work you can do 1 of 3 things 1/ comment out this line 2/ change this line to context = from-pstn 3/ do what I did, add the following 4 lines to your extensions.conf file (the ones in the middle ; ######################################################################## #### ; Inbound Contexts [from] ; ######################################################################## #### [from-sip-external] ;Stanaphone incoming extension exten => 91514413,1,Answer exten => 91514413,2,Goto(from-pstn,s,1) exten => 91514413,3,Hangup ;give external sip users congestion and hangup ; exten => _.,1,AbsoluteTimeout(15) exten => _.,2,Congestion exten => _.,3,Hangup The extension number is obviously the Stanaphone phone number I didn't comment over the 'congestion' section in case it was important. And for the sip trunks configuration section here is the information I put in but you need to customise to your own trunk name Stanaphone-out allow=ulaw auth=md5,plaintext canredirect=no disallow=all dtmfmode=rfc2833 fromuser=91514413 host=sip.stanaphone.com insecure=very qualify=yes secret=******** type=peer username=91514413 User Context 91514413 auth=md5 context=from-sip-external host=sip.stanaphone.com qualify=3000 secret=************* type=peer username=91414413 Register String 91514413:************@sip.stanaphone.com/91514413 Like I said I dont understand the AMP guys set it up this way, might be something I dont understand but at least it is working. Maybe someone else can explain the reason for hanging up on SIP calls. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050409/a16939c0/attachment.htm