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: asterisk-users@lists.digium.com > 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" > <asterisk-users@lists.digium.com> > > 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. > > > > > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by > > http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register > > > > Now: http://www.astricon.net > > > > 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 > > 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users