Manjit, Do you have 3 lines with BroadVoice? If so how do you tell which number is ring in on or which line to dial out on???? I have on line with him now and would like to add two lines..
Thanks, David. On Thu, 2005-01-27 at 14:14 -0800, 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 > -- David Shaw <[EMAIL PROTECTED]> _______________________________________________ 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
