[Asterisk-Users] Stumped by BroadVoice SIP
Dalon Westergreen
dwesterg at gmail.com
Mon Jan 31 14:42:36 MST 2005
I used the setup described, and it works for me. I do have my box as
a DMZ though i do not think it is necessary.
I will post my sip.conf later this evening.
--Dalon
On Mon, 31 Jan 2005 16:10:21 -0500 (EST), asterisk at stephenamadei.com
<asterisk at stephenamadei.com> wrote:
>
> Unfortunately, it doesn't. I have used your config as a guide, and I
> always get the same problem... No registration.
>
> Well, actually, it does eventually register... according to Asterisk.
> But if I try to call outbound, I get a message from BV saying I am not
> registered.
>
> I can't get BroadVoice to register to save my life. I fear it might be a
> NAT problem. Are you using NAT?
>
> I was able to get BroadVoice working behind NAT with X-Lite, but not with
> Asterisk.
>
> I see alot of notes about SIP behind NAT, and that Asterisk is bad behind
> a NAT device. Can Asterisk work behind a NAT device, like the PIX? Or do
> I have to move heaven and earth to get this network engineered to allow
> Asterisk to live in a DMZ?
>
> ----Stephen
>
> On Thu, 27 Jan 2005, Manjit Riat wrote:
>
> > I had a lot of problem with them to set up..
> >
> > You need to register to sip.broadvoice.com
> >
> > And need to have all of their four servers to listen to incoming calls as
> > ony one can send it in..
> >
> > Just posted my config two days ago.
> >
> > http://lists.digium.com/pipermail/asterisk-users/2005-January/085736.html
> >
> > hope that helps
> >
> > -----Original Message-----
> > From: asterisk at stephenamadei.com [mailto:asterisk at stephenamadei.com]
> > Sent: Thursday, January 27, 2005 2:02 PM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Stumped by BroadVoice SIP
> >
> >
> >
> > Hello guys.
> >
> > I am a fairly new user to Asterisk, and I'm just having a tough time.
> >
> > My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
> > number, and I have three systems with SIP phones.
> >
> > The PBX and the SIP phones are all behind a Cisco PIX running NAT.
> > I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
> > little luck.
> >
> > The SIP phones are two X-Lites on Windows and one Kphone on Linux (running
> > from the same system that Asterisk runs on).
> >
> > It appears that the BroadVoice SIP registers and the SIP phones register,
> > as I can call from one Xlite to the Kphone. However, I cannot get
> > incoming calls from BroadVoice. Calling the BroadVoice number results in
> > a 'The party you wish to reach is busy and cannot...' message. I sniffed
> > packets and I can see packets coming in from BroadVoice on port 5060 to
> > the PBX, but they do not correspond with my call attempts. And debugging
> > the sip session shows alot of '404 Not Found'.
> >
> > Also, even though this is meant as a incoming only PBX, I tried to test
> > outgoing calls from an X-Lite softphone to BroadVoice, but it doesn't
> > work, either.
> >
> > I've probably screwed my configs to hell trying to get this to work, but
> > here they are. Any suggestions would be appreciated.
> >
> > Here are my configs, decrufted...
> >
> > sip.conf
> > ------------------------------------------------------------
> > [general]
> > context=sip
> > recordhistory=yes
> > port = 5060
> > bindaddr = 0.0.0.0
> >
> > allow=gsm
> > allow=alaw
> > allow=ulaw
> > allow=adpcm
> > allow=speex
> > allow=ilbc
> > allow=slinear
> > [general]
> > nat=yes
> >
> > register => 2129999999:<password>:2129999999 at 147.135.8.128:5060
> > register => 2129999999:<password>:2129999999 at 147.135.0.128:5060
> >
> > externip = 208.59.47.2
> >
> > localnet=192.168.1.0/255.255.0.0
> >
> > [sip_proxy]
> > type=user
> > context=from-broadvoice
> >
> > [xlite1]
> > type=friend
> > regexten=101
> > username=xlite1
> > secret=<password>
> > callerid="Stephen's Laptop" <101>
> > host=dynamic
> > nat=no
> > canreinite=yes
> > disallow=all
> > allow=gsm
> > allow=ulaw
> > allow=alaw
> > dtmfmode=inband
> > qualify=yes
> >
> > [xlite2]
> > type=friend
> > regexten=103
> > context=sip
> > username=103
> > secret=<password>
> > callerid="Ben's Laptop" <103>
> > host=dynamic
> > nat=no
> > allow=gsm
> > allow=ulaw
> > allow=alaw
> > dtmfmode=inband
> > quality=yes
> >
> > [kphone1]
> > type=friend
> > username=kphone1
> > secret=<password>
> > callerid="Diablo" <102>
> > host=dynamic
> > allow=gsm
> > qualify=yes
> >
> > [sip.broadvoice.com]
> > type=peer
> > host=proxy.dca.broadvoice.com
> > fromdomain=sip.broadvoice.com
> > fromuser=2129999999
> > secret=<password>
> > context=incoming
> > canreinvite=no
> >
> > [broadvoice-out]
> > type=peer
> > dtmfmode=inband
> > host=147.135.0.128
> > user=2129999999
> > username=2129999999
> > authuser=2129999999
> > fromuser=2129999999
> > fromdomain=sip.broadvoice.com
> > md5secret=<password>
> > qualify=yes
> > canreinvite=no
> > disallow=all
> > allow=ulaw
> >
> > [broadvoice-out2]
> > type=peer
> > dtmfmode=inband
> > host=147.135.8.128
> > user=2129999999
> > username=2129999999
> > authuser=2129999999
> > fromuser=2129999999
> > fromdomain=sip.broadvoice.com
> > md5secret=<password>
> > qualify=yes
> > canreinvite=no
> > disallow=all
> > allow=ulaw
> >
> > [broadvoice-incoming]
> > type=peer
> > dtmfmode=inband
> > host=147.135.8.128
> > context=incoming
> > qualify=yes
> > nat=yes
> > canreinvite=no
> > fromdomain=sip.broadvoice.com
> > username=2129999999
> > fromuser=2129999999
> > insecure=very
> >
> > [broadvoice-incoming2]
> > type=peer
> > dtmfmode=inband
> > host=147.135.0.128
> > context=incoming
> > qualify=yes
> > nat=yes
> > canreinvite=no
> > fromdomain=sip.broadvoice.com
> > username=2129999999
> > fromuser=2129999999
> > insecure=very
> > ---------------------------------------------------------
> >
> > extensions.conf
> > ---------------------------------------------------------
> > [general]
> > static=yes
> > writeprotect=no
> >
> >
> > [globals]
> > CONSOLE=Console/dsp ; Console interface for demo
> > IAXINFO=guest ; IAXtel username/password
> > TRUNK=Zap/g2 ; Trunk interface
> > TRUNKMSD=1 ; MSD digits to strip
> > (usually 1 or 0)
> >
> >
> > [iaxtel700]
> > exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
> >
> > [iaxprovider]
> >
> > [trunkint]
> > exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9011.,2,Congestion
> >
> > [trunkld]
> > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91NXXNXXXXXX,2,Congestion
> >
> > [trunklocal]
> > exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _9NXXXXXX,2,Congestion
> >
> > [trunktollfree]
> > exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91800NXXXXXX,2,Congestion
> > exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91888NXXXXXX,2,Congestion
> > exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91877NXXXXXX,2,Congestion
> > exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
> > exten => _91866NXXXXXX,2,Congestion
> >
> > [international]
> > ignorepat => 9
> > include => longdistance
> > include => trunkint
> >
> > [longdistance]
> > ignorepat => 9
> > include => local
> > include => trunkld
> >
> > [local]
> > ignorepat => 9
> > include => default
> > include => parkedcalls
> > include => trunklocal
> > include => iaxtel700
> > include => trunktollfree
> > include => iaxprovider
> >
> > [macro-stdexten];
> > exten => s,1,Dial(${ARG2},20) ; Ring the
> > interface, 20 seconds maximum
> > exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based
> > on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> >
> > exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable,
> > send to voicemail w/ unavail announce
> > exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #,
> > return to start
> >
> > exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
> > voicemail w/ busy announce
> > exten => s-BUSY,2,Goto(default,s,1) ; If they
> > press #, return to start
> >
> > exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat
> > anything else as no answer
> >
> > exten => a,1,VoicemailMain(${ARG1}) ; If they
> > press *, send the user into VoicemailMain
> >
> > [demo]
> > exten => s,1,Wait,1 ; Wait a second, just for fun
> > exten => s,2,Answer ; Answer the line
> > exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
> > exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
> > exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
> > exten => s,6,BackGround(demo-instruct) ; Play some instructions
> >
> > exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
> > exten => 2,2,Goto(s,6)
> >
> > exten => 3,1,SetLanguage(fr) ; Set language to french
> > exten => 3,2,Goto(s,5) ; Start with the congratulations
> >
> > exten => 1000,1,Goto(default,s,1)
> > exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
> > ; (but skip if channel is not up)
> > exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
> >
> > exten => 1235,1,Voicemail(u1234) ; Right to voicemail
> >
> > exten => 1236,1,Dial(Console/dsp) ; Ring forever
> > exten => 1236,2,Voicemail(u1234) ; Unless busy
> >
> > exten => #,1,Playback(demo-thanks) ; "Thanks for trying the
> > demo"
> > exten => #,2,Hangup ; Hang them up.
> >
> > exten => t,1,Goto(#,1) ; If they take too long, give up
> > exten => i,1,Playback(invalid) ; "That's not valid, try again"
> >
> > exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
> > exten => 500,2,Dial(IAX2/guest at misery.digium.com/s at default) ; Call the
> > Asterisk demo
> > exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
> > exten => 500,4,Goto(s,6) ; Return to the start over message.
> >
> > exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
> > exten => 600,2,Echo ; Do the echo test
> > exten => 600,3,Playback(demo-echodone) ; Let them know it's over
> > exten => 600,4,Goto(s,6) ; Start over
> >
> > exten => 8500,1,VoicemailMain
> > exten => 8500,2,Goto(s,6)
> >
> >
> > [default]
> > include => demo
> >
> > ; I modified stuff from here down...
> >
> > exten=_9NXXNXXXXXX, 1, dial(SIP/${EXTEN}@broadvoice-out,30)
> > exten=_9NXXNXXXXXX, 2, dial(SIP/${EXTEN}@broadvoice-out2,30)
> > exten=_9NXXNXXXXXX, 3, congestion()
> > exten=_9NXXNXXXXXX, 103, busy()
> >
> > [sip]
> > exten => 1,1,Dial(SIP/xlite1,20,tr)
> > exten => 2,1,Dial(SIP/kphone1,20,tr)
> > exten => 3,1,Dial(SIP/xlite2,20,tr)
> > exten => 1000,1,Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,20,tr)
> >
> > [incoming]
> > exten => 1,1,Dial(SIP/xlite1,20,tr)
> > exten => 2,1,Dial(SIP/kphone1,20,tr)
> > exten => 3,1,Dial(SIP/xlite2,20,tr)
> > exten => 1000,1,Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,20,tr)
> >
> > [from-broadvoice]
> > exten => 1,1,Dial(SIP/xlite1,20,tr)
> > exten => 2,1,Dial(SIP/kphone1,20,tr)
> > exten => 3,1,Dial(SIP/xlite2,20,tr)
> > exten => 1000,1,Dial(SIP/xlite1&SIP/kphone1&SIP/xlite2,20,tr)
> >
> > -----------------------------------------------------------
> >
> > ----Steve
> > Stephen Amadei
> > 5114 Harbor Beach Blvd
> > Brigantine Beach, NJ 08203
> > (609) 703-9649
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ----Steve
> Stephen Amadei
> 5114 Harbor Beach Blvd
> Brigantine Beach, NJ 08203
> (609) 703-9649
>
> Current resume at http://www.amadei.com/resume.doc
> _______________________________________________
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