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:[EMAIL PROTECTED],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 [EMAIL PROTECTED]:1] Dial("SIP/100-09f2ee18", "SIP/[EMAIL
PROTECTED]|30|r") in new stack
-- Called [EMAIL PROTECTED]
-- 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 <[EMAIL PROTECTED]> wrote:
> From: Brad <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Basic outbound calling issue
> To: [email protected]
> Cc: "Felippe Silvestre" <[EMAIL PROTECTED]>
> 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
> <[EMAIL PROTECTED]> wrote:
>
> > From: Felippe Silvestre
> <[EMAIL PROTECTED]>
> > Subject: RE: [asterisk-users] Basic outbound calling
> issue
> > To: [EMAIL PROTECTED], "Asterisk Users Mailing
> List - Non-Commercial Discussion"
> <[email protected]>
> > Date: Friday, August 15, 2008, 12:25 PM
> > Check if you have some rule to dial under brad1
> context
> >
> > dialplan [EMAIL PROTECTED]
> >
> > Regards
> >
> > Felippe Silvestre
> >
> >
> > -----Original Message-----
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] 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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> >
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>
>
>
>
> _______________________________________________
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