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Grandstream-Guru
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blackvel schrieb:Another update
Also don't worry, my first test trying to redirect Asterisk callee from a SIP client (SJPhone) to a handy number with SIP 302 Moved temporarily
showed us the following result:
1. Asterisk is being redirected from 99-number to username@nikotel
2. Asterisk will get a new 302 SIP reply and will be forwarded to to sip:<local IP of SJPhone PC>
3. Asterisk is being redirected from sip:<local IP of SJPhone PC> to local extension sip:49<handy nr.>
Asterisk does NOT try to contact 49<handy nr.>@nikotel which would mean YOU would have to pay for being redirected to 0190 or any other PSTN number.
Instead Asterisk follows the behavior for searching the extension in it's local dial plan (LOCAL/) as before as we never would have set promiscredir=yes.
This is the case, because the SIP client directly tries to forward (redirect comes from different IP) and not Nikotel.
And since promiscredir is pinned to Nikotel (IP), only Nikotel can redirect you. But Nikotel does NOT redirect you to any PSTN number.
Das habe ich ja verstanden - aber diese gesamte Weiterleitungslogik scheint mit SIP-Endgeräten (und IAX wahrscheinlich auch nicht ...) überhaupt nicht zu funktionieren :-( auf jeden Fall habe ich das bisher mit keiner der beschriebenen Einstellungen zum Laufen bekommen.
Ich kann alles anrufen - nur keine internen Nikotel-Nummern.
Da bekomme ich dauerend einen "Proxy Authorization Required" auf mein SIP-Endgerät, der permanent fehlschlägt.
Code:
Sip read:
SIP/2.0 407 Proxy Authentication Required "Account-user must match from-user"
Via: SIP/2.0/UDP 217.20.120.121;branch=z9hG4bK528cbde1
From: "Corinna Fiedler" <sip:[email protected]>;tag=as0ab9d493
To: <sip:[email protected]>
CSeq: 103 INVITE
Call-ID: [email protected]
Expires: 31
Content-Length: 0