Kevin
Fleming has said on numerous ocassions that this is known not to work, and is
not supported.
-----Original Message-----Hi!
From: Benjamin Stocker [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 06, 2006 4:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Realtime and SIP Registration
I use the following configuration to register my asterisk server to my SIP provider:
register => 12345:[EMAIL PROTECTED]/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/[EMAIL PROTECTED])
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.
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