[asterisk-gui] Asterisk GUI no time based rule menu HELP!

[bkruse]

Alek Katamail wrote:
> Hi Brandon,
> Thank you for the answer, ports opened and now audio it's ok.
> Regarding time based rules I can place them directly on the extensions.conf
> file then?
>   
Yes
> When a new version of the gui will be out?
>   
not sure
> I got 3 more problems, maybe you can help me:
>
> 1 - if I put more than one sip trunk from the same provider when calling a
> number always rings the same extension. Maybe I should open more than just
> 5060 sip port?
>   
No, you should use one line to one provider, and then match the DID's in 
the incoming
calling rules.
> 2 - Voicemail is configured for extension 999 but it doesn't works, this is
> what it says: Executing [s-CONGESTION at macro-trunkdial:1]
> NoOp("SIP/102-0854e380", "") in new stack
>   == Auto fallthrough, channel 'SIP/102-0854e380' status is 'CONGESTION'
> In my SNOM360 phone when I hit the mail key it doesn't work but no message
>
>   
That is NOT a voicemail extension. Obviously it is trying to dial a trunk.

snom360's have to be programmed to dial a certain extension when you hit
the mail but, it isn't automagical.
> 3 - should I put IAX to yes in all the users?
>
>   
Depends, are they using IAX2? In the case of a snom/polycom/other sip phone
then no, but if they want to also register iax2 to that account (say 
from their
own asterisk box) then sure.

For your situation, probably not.

> Bye Alek
>
> -----Original Message-----
> From: asterisk-gui-bounces at lists.digium.com
> [mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of bkruse
> Sent: martedì 3 giugno 2008 20.46
> To: Asterisk GUI project discussion
> Subject: Re: [asterisk-gui] Asterisk GUI no time based rule menu HELP!
>
> Use those ports in rtp.conf (the rtp ports forwarded)
>
> Other than that, if it is not behind a nat, you do not need those fields.
>
> The time based rules were pulled out because they were "unstable", this
> is still being worked on.
>
> -brandon
>
> Alek Katamail wrote:
>   
>> Hi Guys,
>>
>> I’m using Asterisk 1.4.current with addons and Asterisk GUI taken from:
>>
>> svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
>>
>> Is this the latest correct version? Because everything seems to work 
>> but I can’t see any time based rule in the menu
>>
>> And another question, in my asterisk GUI I have this fields:
>>
>> Extern ip:
>> Extern Host:
>> Extern Refresh:
>> Local Network Address:
>> NAT mode: yes
>> Allow RTP Reinvite
>>
>> How do I have to fill them?
>> My dedicated server is remote and has his own IP.
>>
>> Firewall ports opened are:
>>
>> UDP 5060
>> TCP 8088 (for the asterisk's GUI)
>> UDP 13456:16482 (rtp ports)
>>
>> Bye Alek
>>
>> ------------------------------------------------------------------------
>>
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>
>
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-brandon