Hi I would like to make point to point SIP calls between two digium with asterisk. I have got the asterisk working with the digium fxs interface but canīt get the SIP session working yet. I have configured the sip.conf and extensions.conf files the way that I can have calls between the extensions 3002 (ip 10.5.0.4), located in one asterisk, and 2008(10.5.0.2) located in the other one. When I try to make calls between the two extensions, from 3002 to 2008, for example, I got the following message:
-- Got SIP response 404 "Not Found" back from 10.5.0.2 Bellow thereīs the sip.conf and extensions.conf files I have modified for the 2008 extension (the same way Iīve done for the 3002 one): ======= sip.conf ========== [general] context=default port=5060 localnet=10.5.0.2 maxexpirey=33600 defaultexpirey=33600 allow=g729 language=en relaxdtmf=yes localnet=0.0.0.0/0.0.0.0 [3002] context=from-sip type=friend username=3002 fromuser=2008 host=10.5.0.4 allow=g729 [2008] type=friend context=2008 host=10.05.0.2 username=2008 disallow=all allow=g729 ======== extensions.conf ========= [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp [from-sip] exten => 3002,1,Dial(SIP/3002,20,tr) [default] include => from-sip What is wrong with these config files? Could anyone help me? What am I supposed to do or change on these files to get the SIP working so I could make point to point calls? Does anyone have sip.conf and extensions.conf examples to handle point to point calls? Thanks in advance and best regards __________________________________ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users