Hi! <br> <p>Im looking for a very basic example for the following simple problem. <br> I've been searching <a href="http://voip-info.org">voip-info.org</a> and looked in the ORA book without a <br> clue. I have a SIP account at
<a href="http://sip.provider.com">sip.provider.com</a> and my own asterisk <br> server. What I want is the following: <br> </p><p> I. Register my phone to my asterisk server, not directly to <a href="http://provider.com">
provider.com</a> <br> II. My asterisk server should ring my phone when somebody calls me on <mynumber>@<a href="http://provider.com">provider.com</a> <br> III. Asterisk forwards my outgoing calls to <a href="http://provider.com">
provider.com</a> <br> </p><p>I found a lot of sample snippets but none of them really works. The two <br> main problems are: <br> </p><p>A. When somebody calls me, he get's a "user unavailable" from <br> <a href="http://provider.com">
provider.com</a>, but my asterisk server successfully registered at <br> <a href="http://provider.com">provider.com</a>: <br> </p><p> (sip.conf) <br> register => <user>:<pwd>@<a href="http://sip.provider.com/">
sip.provider.com/</a><user> <br> </p><p>B. When I call a number, my asterisk server says: " Failed to <br> authenticate on INVITE". But all login informations for <a href="http://provider.com">provider.com
</a> <br> are correct. <br> </p><p> (sip.conf) <br> [<user>] <br> type=friend <br> secret=<pwd> <br> username=<user> <br> fromuser=<user> <br> canreinvite=yes <br> </p><p> (extensions.conf
) <br> exten => 0041321112233,1,Dial(SIP/${EXT<a target="_parent" href="http://groups.google.ch/groups/unlock?msg=c9756294de75b896&hl=de&_done=/group/Asterisk-users/browse_thread/thread/de1e6e8328c2a255/c9756294de75b896%3Fhl%3Dde">
...</a>@<a href="http://sip.provider.com">sip.provider.com</a>,60,r) <br> </p>Thanks for any help!