[asterisk-gui] Linksys ATA SPA3102 Using The GUI

[Fadi Almasalha]

David , Thanks a lot , the settings works with me , but can you explain to
me how did you setup the 102 extension to hit the incoming rules.

 

GUI 2 only let define incoming rules for trunk only . Thanks a lot for your
help  

 

Yours 

Fadi 

 

 

From: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of David Kerr
Sent: Wednesday, September 17, 2008 9:21 PM
To: Asterisk GUI project discussion
Subject: Re: [asterisk-gui] Linksys ATA SPA3102 Using The GUI

 

You have to create it as a trunk, not a user. I have a SPA3000 (earlier
version of the 3102 I think) working this way.  For host URL enter "dynamic"
this tells Asterisk not to register with the service provider, but rather
that the service provider (the SPA3102) will register with Asterisk.  enter
a userid/password that the SPA will use to Authenticate with Asterisk. My
context in Users.conf ended up looking like this... (this was built with the
v1 GUI, but I manually added some lines like qualify. v2 should build a
similar entry for you)

[localPSTN]
allow=ulaw,alaw,gsm,g726
context=DID_localPSTN
dialformat=${EXTEN:1}
canreinvite=no
hasexten=no
hasiax=no
hassip=yes
host=dynamic
type=friend
qualify=yes
registeriax=no
registersip=no
secret=password
trunkname=Custom - localPSTN
trunkstyle=customvoip
username=localPSTN
insecure=very
dtmfmode=auto

You also need to configure the PSTN Line of the SPA to enable the PSTN to
VoIP Gateway and set a PSTN answer delay of about 3 seconds (to give time
for the CallerID to come through) and set the PSTN Caller Default DP to one
of the dial plans (I used "1").  You need to set up this Dial Plan to tell
the SPA to call through to Asterisk on incoming calls. In my case I set
"(S0<:102>)"  which means zero seconds delay, try and call extension 102.
But the extension number is actually not used by asterisk, as it hits the
incoming call rules logic rather than go to the extension. 

Hope this helps.

David

On Wed, Sep 17, 2008 at 4:42 PM, Fadi Almasalha <falmas2 at uic.edu> wrote:

You can't Create an analog trunk without any hardware FXO PCI Cards


-----Original Message-----
From: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of Pari Nannapaneni
Sent: Wednesday, September 17, 2008 3:39 PM
To: Asterisk GUI project discussion
Subject: Re: [asterisk-gui] Linksys ATA SPA3102 Using The GUI

> I have created a regular user account and linked it to the SPA3102

you should probably create an analog trunk and connect the SPA3102
like just any other PSTN connection.

-Pari

----- Original Message -----
From: "Fadi Almasalha" <falmas2 at uic.edu>
To: asterisk-gui at lists.digium.com
Sent: Wednesday, September 17, 2008 3:23:33 PM GMT -06:00 US/Canada Central
Subject: [asterisk-gui] Linksys ATA SPA3102 Using  The GUI

Hi ,

I am not able to create a sip trunk using the asterisk GUI 2.0 for Linksys
SPA 3102 . Please any ideas how to create a sip trunk on the GUI .

I have created a regular user account and linked it to the SPA3102 , It
receive calls and make calls with no problem . but the only problem that I
can't configure the incoming call rules , as it only support the trunk
method .

Yours Fadi

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