I figure it out, asterisk is using the wrong ip address. I have bind address set to the correct ip address. How to I force asterisk to use the correct ip address?
--- On Fri, 8/15/08, Brad <[EMAIL PROTECTED]> wrote: > From: Brad <[EMAIL PROTECTED]> > Subject: Re: [asterisk-users] Basic outbound calling issue : a lot closer > To: [email protected] > 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:[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. > > > > > > > > > > > > > > > _______________________________________________ > > > -- 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 _______________________________________________ -- 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
