<div>Just put your carriers in static sip.conf and put your users in the realtime DB. <br> </div>
<div>bp<br> </div>
<div><span class="gmail_quote">On 6/15/06, <b class="gmail_sendername">Douglas Garstang</b> <<a href="mailto:dgarstang@oneeighty.com">dgarstang@oneeighty.com</a>> wrote:</span>
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<div><span><font face="Arial" color="#0000ff" size="2">Kevin Fleming has said on numerous ocassions that this is known not to work, and is not supported.</font></span></div>
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<div dir="ltr" align="left"><font face="Tahoma" size="2">-----Original Message-----<br><b>From:</b> Benjamin Stocker [mailto:<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:bstocker@gmail.com" target="_blank">
bstocker@gmail.com</a>]<br><b>Sent:</b> Tuesday, June 06, 2006 4:31 AM<br><b>To:</b> <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com
</a><br><b>Subject:</b> [Asterisk-Users] Asterisk Realtime and SIP Registration<br><br></font></div>Hi!<br><br>I use the following configuration to register my asterisk server to my SIP provider:<br><br>register => <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://12345:passwd@sip.provider.com/12345" target="_blank">
12345:passwd@sip.provider.com/12345</a><br><br>sip.conf :<br>[sipout-test]<br>type=peer<br>username=12345<br>fromuser=12345<br>fromdomain=<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://provider.com/" target="_blank">
provider.com</a><br>secret=passwd<br>insecure=very<br>host=<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://sip.provider.com/" target="_blank">sip.provider.com </a><br>qualify=yes<br>context=test-incoming
<br><br>extensions.conf:<br>exten => 12345,1,Dial(SIP/10)<br>exten => _0NXZXXXXXX,1,Dial(SIP/${EXTEN}@sipout-test)<br><br>This works fine when I put it into the config files. I can dial other numbers via my provider and receive calls. Wenn I put everything into Realtime tables (except the register command), incoming calls work only after
<br><br> * I make at least one outgoing call<br> - or -<br> * Somebody calls me twice<br><br>On incoming calls, the caller first gets a 'user unavailale' from my SIP provider. When hanging up and calling again, the connection establishes successfully and I see this when entering 'sip show peers':
<br><br>sipout-test/12345 IP.AD.DR.ESS 5060 UNKNOWN<br><br>This line does not show up when I registering my phone to my asterisk server. But it shows up immediately after registerung the phone when I use config files instead of RTA.
<br><br>I don't know wheter this is RTA- or a config-problem. <br><br></blockquote></div></div><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">
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