[Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c

Sat Apr 8 14:38:13 MST 2006

Tzafrir,

How did you set  sip:tzafrir at local.xorcom.com

I use ser----asterisk

look at my sip.conf and extensions.conf

Regards 
Harry 
////////////////////////////////////////////////////
[general]

context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes

rtptimeout=60
rtpholdtimeout=300
useragent=PBX
dtmfmode = rfc2833
checkmwi=20

promiscredir=no
nat=yes

autodomain=no
domain=nxs.yi.org,sip
allowexternalinvites=yes
rtcachefriends=yes

rtupdate=yes
rtautoclear=yes
ignoreregexpire=yes

and extensions.conf
[general]

static=yes
writeprotect=no
autofallthrough=yes
//////////////////////////////////////////////////////

[globals]

[mainmenu]
exten => s,1,Answer()
exten =>
s,n,GotoIfTime(09:30-21:00|mon-sun|*|*?day,s,1)
exten =>
s,n,GotoIfTime(21:01-09:31|mon-sun|*|*?night,s,1)

[night]
exten => s,1,PlayBack(closed)
exten => s,2,Voicemail(u84)
exten => s,3,Hangup

[day]
exten => s,1,BackGround(annoucement)
exten => s,2,WaitExten(10)

exten => t,1,Playback(no-answer)
exten => t,2,Hangup()

exten => *,1,PlayBack(waiting)
exten => *,2,Queue(info|t||)

exten => 1,1,BackGround(ipbx)
exten => 1,2,Goto(s,1)

exten => 2,1,Playback(informations)
exten => 2,2,Goto(music,600,1)

exten => i,1,PlayBack(key-invalide)
exten => i,2,Goto(s,1)

[sip]
include => info
include => support

exten => info,1,Answer()
exten => info,2,Dial(Sip/84,10)
exten => info,3,Dial(Sip/85,10)
exten => info,4,Hangup

exten => support,1,Answer()
exten => support,2,Queue(support|t||)
exten => support,3,Hangup

[pstn]

exten => s,1,Answer()
exten => s,2,NVFaxDetect()
exten => s,3,Goto(mainmenu,s,1)
exten => s,4,Hangup
exten => fax,1,Dial(Zap/g2)
exten => talk,1,Goto(mainmenu,s,1)

exten => t,1,Hangup()

include => outgoing-pstn

[info]

exten => 84,1,Answer()
exten => 84,2,Dial(Sip/84,30,t)
exten => 84,3,VoiceMail(u84)
exten => 84,103,VoiceMail(b84)

exten => 85,1,Answer()
exten => 85,2,Dial(Sip/85,30,t)
exten => 85,3,VoiceMail(u85)
exten => 85,103,VoiceMail(b85)

include => parkedcalls
include => guest
include => agents
include => pstn
include => music
include => mailbox
include => support
include => aliases

[support]

exten => 86,1,Answer()
exten => 86,2,Dial(Sip/86,30,t)
exten => 86,3,VoiceMail(u86)
exten => 86,103,VoiceMail(b86)

exten => 87,1,Answer()
exten => 87,2,Dial(Sip/87,30,t)
exten => 87,3,VoiceMail(u87)
exten => 87,103,VoiceMail(b87)

include => parkedcalls
include => guest
include => agents
include => pstn
include => music
include => mailbox
include => info

[guest]

exten => 88,1,Answer()
exten => 88,2,Dial(Sip/88,30,t)
exten => 88,3,VoiceMail(u88)
exten => 88,103,VoiceMail(b88)

include => music
include => mailbox

[fax]
exten => fax,1,Dial(Zap/2,40)
exten => fax,2,Congestion
exten => fax,102,Congestion

include => outgoing-pstn

[outgoing-pstn]

ingnorepat => 0
exten => _0XXXX,1,ChanIsAvail(Zap/g1, j)
exten => _0XXXX,2,Dial(Zap/g1/${EXTEN:1})
exten => _0XXXX,102,Playback(busy)
exten => _0XXXX,103,Hangup

exten => _0XXXX.,1,Dial(Zap/g1/${EXTEN:1})

[mailbox]
exten => 700,1,Answer()
exten => 700,2,VoiceMailMain()

[music]
exten => 600,1,Answer()
exten => 600,2,WaitMusicOnHold(60)
exten => 600,3,Hangup
exten => music,1,Goto(600,1)

[agents]
;Agent Login
exten=>
501,1,AgentCallbackLogin(||${CALLERIDNUM}@info)
exten=>
502,1,AgentCallbackLogin(||${CALLERIDNUM}@support)

;Agent Logout
exten=> 503,1,AgentCallbackLogin(||l)
exten=> 504,1,AgentCallbackLogin(||l)

[aliases]
exten => alice,1,Goto(info,84,1)
exten => bob,1,Goto(support,86,1)
//////////////////////////////////////////////////////

--- Tzafrir Cohen <tzafrir.cohen at xorcom.com> a écrit :

> On Sat, Apr 08, 2006 at 09:31:43PM +0200,
> hgaillac-sip at yahoo.fr wrote:
> > Hello,
> > 
> > Anybody could explain me why asterisk spend time
> to
> > send back to proxy or sip agent authentication
> > messages 407
> > 
> 
> I believe some people tried. At least when I
> happened to be present on
> #asterisk.
> 
> > nobody can call me from other domains.
> 
> Tough. But this is not asterisk-users. Any relevant
> questions you have
> to the developers?
> 
> > can we disable authentication for none peers or
> users 
> 
> You were told how to do something quite similar
> (allowguests). 
> Is that good enough? If not: why not? Still, a
> -users question.
> 
> > 
> > Asterisk ask authentication 407 for
> > sip:info at nxs.yi.org or sip:support at nxs.yi.org
> 
> And after people tell you that this is a matter of
> setting up sip.conf
> properly, you still only bother quoting yor
> dialplan, rather than
> sip.conf.
> 
> You also post several thread rather than keeping
> everything in one
> thread. You also post to multiple lists and use
> subject lines suuch as
> "HELP !!!!!".
> 
> Please read a bit about nettique. Your behaviour
> does does not encourge
> people to help you.
> 
> -- 
> Tzafrir Cohen      sip:tzafrir at local.xorcom.com
> icq#16849755       iax:tzafrir at local.xorcom.com
> +972-50-7952406           
> tzafrir.cohen at xorcom.com  http://www.xorcom.com
> _______________________________________________
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