[Asterisk-Users] Stumped by BroadVoice SIP {Scanned}
Manjit Riat
manjit at riat.net
Fri Jan 28 09:48:11 MST 2005
None just one single line.
-----Original Message-----
From: David Shaw [mailto:asterisk at ke6upi.com]
Sent: Friday, January 28, 2005 6:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Stumped by BroadVoice SIP {Scanned}
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: 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
>
>
>
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--
David Shaw <asterisk at ke6upi.com>
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