[Asterisk-Users] Asterisk Realtime and SIP Registration

[William Piper]

Just put your carriers in static sip.conf and put your users in the realtime
DB.

bp

On 6/15/06, Douglas Garstang <dgarstang at oneeighty.com> wrote:
>
>  Kevin Fleming has said on numerous ocassions that this is known not to
> work, and is not supported.
>
> -----Original Message-----
> *From:* Benjamin Stocker [mailto:bstocker at gmail.com]
> *Sent:* Tuesday, June 06, 2006 4:31 AM
> *To:* asterisk-users at lists.digium.com
> *Subject:* [Asterisk-Users] Asterisk Realtime and SIP Registration
>
> 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.
>
>
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