Buongiorno a tutti;
Come da titolo, non riesco a configurare il mio centralino con un FTA 5120 che altro non è che un gateway VoIP ovvero, ha in ingresso 2 porte FXO e una WAN dove collegarci il centralino. Ho provato a creare un account come se il gateway fosse un interno ma anche se forse riesco a farlo registrare, non riesco ad instradare chiamate in entrata ed uscita.
Set up a pjsip trunk. Trunk Name should match the username on the FTA.
Secret should match the SIP password on the FTA.
Authentication: Both
Registration: Receive
Match Inbound Authentication: Auth Username
Rewrite Contact: Yes
Force Rport: Yes
If you have trouble, turn on pjsip logger and paste the Asterisk log for a failing registration (if the FTA fails to register), or a failing incoming or outgoing call at pastebin.com and post the link here.
hi, thank you… now i’m able to do the outgoing call, but now i have a problem whit the incoming calls, that’s the error on freepbx
this is the log from fta 5120
thank you for reply
I know nothing about the FTA, so here is some general info:
The FTA should be configured with a username that matches the Trunk Name on FreePBX. If there is a separate setting called “auth username”, “authentication name” or similar, that should have the same value.
When the FTA receives a call on an analog line, it sends an INVITE. The From header should contain the caller’s number (if so configured), or the username (if the caller ID is blocked or the device can’t parse it). What is 6004? It seems too short to be a caller ID, unless the calling device is a PBX or you’re in a country such as Niue where 6004 is a valid mobile number.
In either case, Asterisk will challenge the INVITE with a 401 and the FTA should resend it with an Authorization header containing a username= parameter with the authentication name (that matches the Trunk Name). Because the trunk has Match Inbound Authentication set to Auth Username, it should match the call from the FTA.
If all this is correctly set and the calls still fail, at the Asterisk command prompt (not a shell prompt) type
pjsip set logger on
call in again, paste the Asterisk log for the call (which will now include the SIP trace) and post the link.
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