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<DIV><SPAN class=597055314-15062006><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 class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma
size=2>-----Original Message-----<BR><B>From:</B> Benjamin Stocker
[mailto:bstocker@gmail.com]<BR><B>Sent:</B> Tuesday, June 06, 2006 4:31
AM<BR><B>To:</B> asterisk-users@lists.digium.com<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
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.
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