[Asterisk-Users] Asterisk Realtime and SIP Registration

[Benjamin Stocker]

Hi!

I use the following configuration to register my asterisk server to my SIP
provider:

register => 12345:passwd at sip.provider.com/12345

sip.conf:
[sipout-test]
type=peer
username=12345
fromuser=12345
fromdomain=provider.com
secret=passwd
insecure=very
host=sip.provider.com
qualify=yes
context=test-incoming

extensions.conf:
exten => 12345,1,Dial(SIP/10)
exten => _0NXZXXXXXX,1,Dial(SIP/${EXTEN}@sipout-test)

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

  * I make at least one outgoing call
  - or -
  * Somebody calls me twice

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':

sipout-test/12345  IP.AD.DR.ESS                 5060     UNKNOWN

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.

I don't know wheter this is RTA-  or a config-problem.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060606/c8aa04aa/attachment.htm