[asterisk-users] Basic outbound calling issue

[Brad]

extensions.conf

[To_Airspring]
exten => 55,1,Playback(demo-echotest) ; Let them know what's going on
exten => 55,2,Echo ; Do the echo test
exten => 55,3,Playback(demo-echodone) ; Let them know it's over

exten => 100,1,Dial(SIP/100,20)

sip.conf

;; twinkle softphone
[100]
user=100
nat=yes
type=friend
secret=andreasd
host=dynamic
context=To_Airspring


This should ba all I need

exten => 100,1,Dial(SIP/100,20) should catch it and send it to Sip????


--- On Fri, 8/15/08, Felippe Silvestre <Felippe.Silvestre at locaweb.com.br> wrote:

> From: Felippe Silvestre <Felippe.Silvestre at locaweb.com.br>
> Subject: RE: [asterisk-users] Basic outbound calling issue
> To: bcddd214 at yahoo.com, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> Date: Friday, August 15, 2008, 12:25 PM
> Check if you have some rule to dial under brad1 context
> 
> dialplan 919544790554 at your_context
> 
> Regards
> 
> Felippe Silvestre
>  
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf
> Of Brad
> Sent: Friday, August 15, 2008 12:09
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Basic outbound calling issue
> 
> I am trying to lauch a first outbound call.
> I am connected to my telco via a peer which is a little 
> different from what I consider the norm.
> 
> extinsions.conf
> 
> [To_Bandwidth]
> ignorepat => 9
> exten => 9,1,Dial(Sip/g2/)
> exten => 9,2,Congestion
> 
> sip.conf
> 
> [To_Bandwidth]
> canreinvite=yes
> context=from-pstn
> dtmfmode=rfc2833
> host=xxxx.com
> nat=no
> outboundproxy=xxx.com
> qualify=no
> type=peer
> 
> 
> error
> 
> [Aug 15 11:02:27] NOTICE[25791]: chan_sip.c:14035 
> handle_request_invite: Call from 'brad1' to
> extension 
> '919544790554' rejected because extension not
> found.
> 
> 
>       
> 
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