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

[Pari Nannapaneni]

You are right, I've just commited a fix for this.
http://lists.digium.com/pipermail/asterisk-gui-commits/2007-March/000415.html

thanks
Pari


dima wrote:
>> 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.
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-gui mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-gui
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-gui mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-gui
> 
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-gui mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-gui