[asterisk-gui] IAX and SIP

[Josh Richards]

Hi Trevor,

users.conf should not be involved if I'm understanding your situation and
you should be able to accomplish everything from the GUI without resorting
to the File Editor/shell.  If this is for inbound SIP/IAX calls from a
provider (that would seem to be the case), they should added under Service
Providers.   Then decide what you want done with incoming calls from that
service provider by adding a rule under Incoming Calls.   They could be
routed directly to particular extensions (phones), voicemail, the IVR
(auto-attendant), whatever.

If you have SIP (or even IAX) phones (hardware or software, makes no
difference) those will need to be configured in order to have extensions to
route inbound calls to (unless you'll just keeping all calls into the
IVR).   These will end up in users.conf but, again, there should not be a
reason you'll need to leave the GUI just to get these working initially.
Perhaps later if you want to have a more advanced dialplan or set some less
common sip/users.conf options.

-jr

From: Trevor Benson <tbenson at a-1networks.com
>
> Having a few issues in Custom voip with SNV Revision: 2131.  When inbound
> SIP calls come in I get a "Failed to authenticate user "A-1 NET SOLUTIO"
> <sip:xxxxxxxx" well thats the Caller ID from teliax being sent along.  Not
> really used to setting users.conf manually so not sure what it could be,
> as it seems self explanatory compared to the old sip.conf, and i dont see
> anything obviously wrong.
>
> IAX gives me the error  chan_iax2.c:7344 socket_process: Rejected connect
> attempt from 207.174.202.2, who was trying to reach '8005551234@' (number
> replaced for mailing list) but nothing after the @.  If I set inbound rules
> for ANY unrecognized AND the actual number from this provider to route
> somewhere, it still says rejecting connect attempt.
>
> Willing to debug, but want to bring up the line as well so I will place it
> into the system the old way by hand if need be.
>
> Thanks,
> Trevor
>
> PS.  This is a client that is so gungho about the asterisk-gui and they
> just want it in production in beta anyway.  We can pretty much do any direct
> conf editing to resolve problems, but otherwise I wouldn't be worried about
> connecting this line and getting it working.
>
>
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