Hi I'm trying to configure my Asterisk box to provide a simple sample configuration. It's a mandrake 9.1 box, no cards except a sound card. The config I am trying to achieve is simply one server, with two SIP clients.
Two issues are cropping up - the first, when I start Asterisk, the sound goes nuts and I get an error (below) Jan 25 18:16:44 WARNING[163851]: chan_oss.c:268 sound_thread: Read error on sound device: Resource temporarily unavailable When Asterisk starts, I get the error (below): Jan 25 18:06:33 WARNING[81926]: chan_sip.c:446 __sip_xmit: sip_xmit of 0x80db77c (len 459) to 0.0.0.0 returned -1: Invalid argument I'm pretty confident this second error is because I have misconfigured extensions.conf and sip.conf, but I can't see why. When I try to connect to the server with an XTEN client, I get this error: Jan 25 18:15:38 NOTICE[81926]: chan_sip.c:5548 handle_request: Registration from 'Mike <sip:[EMAIL PROTECTED]>' failed for '203.118.186.16' I've tried looking at the www.automated.it setup information, along with the information on fnords.org - this has gotten me this far, but I can't see for the life of me what I have done wrong. If anyone could provide me some pointers, it would be much appreciated. Regards Mike My SIP conf looks like this: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ; Default for incoming calls [Phone1] type=friend secret=yap auth=md5 nat=yes host=dynamic dtfmmode=inband mailbox=1000 username=mike context=sip disallow=all allow=gsm callerid="Mike Nash" <6969> [Phone2] type=friend secret=yap auth=md5 nat=yes host=dynamic dtfmmode=inband mailbox=1000 username=darryl context=sip disallow=all allow=gsm callerid="Darryl West" <6970> My extensions.conf looks like this: [general] ; ; If static is set to no, or omitted, then the pbx_config will rewrite ; this file when extensions are modified. Remember that all comments ; made in the file will be lost when that happens. ; ; XXX Not yet implemented XXX ; static=yes ; ; if static=yes and writeprotect=no, you can save dialplan by ; CLI command 'save dialplan' too ; writeprotect=no ; For more information on applications, just type "show applications" at your ; friendly Asterisk CLI prompt. ; [sip] exten => 1,1,Dial(SIP/Phone1,20,tr) exten => 2,1,Dial(SIP/Phone2,20,tr) exten => 1000,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) _______________________________________________ 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
