Hi!<br><br>I use the following configuration to register my asterisk server to my SIP provider:<br><br>register => <a href="http://12345:passwd@sip.provider.com/12345">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 href="http://provider.com">provider.com</a><br>secret=passwd<br>insecure=very<br>host=<a href="http://sip.provider.com">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>