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), [EMAIL PROTECTED] <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] > > Sent: Thursday, January 27, 2005 2:02 PM > > To: [email protected] > > 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>:[EMAIL PROTECTED]:5060 > > register => 2129999999:<password>:[EMAIL PROTECTED]: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/[EMAIL PROTECTED]/${EXTEN:[EMAIL > > PROTECTED]) > > > > [iaxprovider] > > > > [trunkint] > > exten => _9011.,1,Dial(${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/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; 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/[EMAIL PROTECTED],30) > > exten=_9NXXNXXXXXX, 2, dial(SIP/[EMAIL PROTECTED],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 > > [email protected] > > 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 > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
