I get congestion (same error) with
exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)
not dialing 1
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)
dialing 1
exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r)
dialing 9
All the same
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Executing [9544790554 at To_Airspring:1]
Dial("SIP/100-b7c03ce8", "SIP/544790554 at
xxx.xxx.xxx|30|r") in new stack
-- Called 544790554 at 64.211.41.115
-- Got SIP response 503 "NoCircuitChannelAvailable" back from
64.211.41.115
-- SIP/64.211.41.115-09f2ee18 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel 'SIP/100-b7c03ce8' status is
'CONGESTION'
--- On Fri, 8/15/08, Brad <bcddd214 at yahoo.com> wrote:
> From: Brad <bcddd214 at yahoo.com>
> Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer
> To: asterisk-users at lists.digium.com
> Date: Friday, August 15, 2008, 9:33 PM
> This what they sent me
> You need to send:
> - 11-digit originating # (i.e., 1-NPA-NXX-0000)
> - 10-digit terminating #
>
> This got me a lot further in extensions.conf
>
> exten => _9.,1,Dial(SIP/${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 [91xxxxxxxxxx4 at To_Airspring:1]
> Dial("SIP/100-09f2ee18",
> "SIP/19544790554 at xxx.xxx.xxx|30|r") in new stack
> -- Called 19544790554 at xxx.xxx.xxx
> -- Got SIP response 503
> "NoCircuitChannelAvailable" back from
> xxx.xxx.xxx
> -- SIP/xxx.xxx.xxx-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 91xxxxxxxxxxx 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.
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by
> > > http://www.api-digital.com --
> > >
> > > AstriCon 2008 - September 22 - 25 Phoenix,
> Arizona
> > Register
> > >
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
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> >
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