[asterisk-gui] incomming calls are routed to "demo"

[dima]

> if you are using r421 or later,
> you can right click on a trunk - click the 'advanced' menu
> and edit some key parameters like 
> trunkname, insecure, port, fromdomain, fromuser, contact etc.

Thanks for your reply.
I'm using
Asterisk SVN-trunk-r58881
Asterisk GUI-version Revision: 431

When I click on 'advanced', edit some values and press 'update', those
values are not being saved. I'm getting 'No changes made'.
I've tried to change them around directly in the config file, but still
no effect. 

> Can you provide us with what asterisk is saying during an incoming call.
> You can try increasing the debug level and see if that helps 
> in locating where the problem is. 
This is what I see in the console while making an incomming call to
924980040
    -- Executing [s at DID_trunk_1:1] Wait("SIP/924980040-b7d38b70", "1") in new stack
    -- Executing [s at DID_trunk_1:2] Answer("SIP/924980040-b7d38b70", "")
in new stack
    -- Executing [s at DID_trunk_1:3] Set("SIP/924980040-b7d38b70",
"TIMEOUT(digit)=5") in new stack
    -- Digit timeout set to 5
    -- Executing [s at DID_trunk_1:4] Set("SIP/924980040-b7d38b70",
"TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing [s at DID_trunk_1:5] BackGround("SIP/924980040-b7d38b70",
"demo-congrats") in new stack
    -- <SIP/924980040-b7d38b70> Playing 'demo-congrats.gsm' (language
'en')
    -- Got SIP response 400 "Subscription-State header missing" back
from 10.0.0.32
  == Spawn extension (DID_trunk_1, s, 5) exited non-zero on
'SIP/924980040-b7d38b70'

What is not clear to me is if I have to do anything extra to make a call
arriving as 924980040 at DID_trunk_1 and not as s at DID_trunk_1

> 
> ----- Original Message -----
> From: "dima" <dima at scancom.es>
> To: asterisk-gui at lists.digium.com
> Sent: Thursday, March 15, 2007 11:24:47 AM (GMT-0600) America/Chicago
> Subject: [asterisk-gui] incomming calls are routed to "demo"
> 
> Hello everyone
> I've installed asterisk and asterisk-gui trunks from svn today
> (15,march). The thing I couldn't get working properly was incomming
> calls. No matter what incomming pattern I specify, all calls that match
> the pattern are routed to demo. However if I specify "all unmatched
> calls" as an option instead, I do receive incomming calls on extension I
> specify.
> I finally made it with adding this line to sip.conf
> register => myusername:mypass at voip.provider.com/MY_LOCAL_EXTENSION_HERE
> 
> I guess it has something to do with my sip provider and how they route
> calls back to me. However if anyone have faced the same problem and
> knows the solution, please respond.
> Thanks in advance.
> 
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