[asterisk-users] Basic outbound calling issue : a lot closer

[Brad]

This what they sent me
You need to send: 
- 11-digit originating # (i.e., 1-NPA-NXX-0000) 
- 10-digit terminating #

This got me a lot further in extensions.conf

exten => _9.,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)

I am getting a 503 error on the phone and asterisk is giving me:

 == Auto fallthrough, channel 'SIP/100-09ef2cc0' status is 'CONGESTION'
    -- Executing [919544790554 at To_Airspring:1] Dial("SIP/100-09f2ee18", "SIP/19544790554 at 64.211.41.115|30|r") in new stack
    -- Called 19544790554 at 64.211.41.115
    -- Got SIP response 503 "NoCircuitChannelAvailable" back from 64.211.41.115
    -- SIP/64.211.41.115-09ef2cc0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/100-09f2ee18' status is 'CONGESTION'



--- On Fri, 8/15/08, Brad <bcddd214 at yahoo.com> wrote:

> From: Brad <bcddd214 at yahoo.com>
> Subject: Re: [asterisk-users] Basic outbound calling issue
> To: asterisk-users at lists.digium.com
> Cc: "Felippe Silvestre" <Felippe.Silvestre at locaweb.com.br>
> Date: Friday, August 15, 2008, 9:06 PM
> 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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