[Asterisk-Users] parameters documentation

Wed Oct 12 07:34:07 MST 2005

I come from a NBX100
No documentation available.
1 day it starts saying: "syslog full" and voicemail stop working
No one was able to tell me what was the meaning of that alert
.
3COM NBX anyway is a good product, but the price is too high, especially 4
years ago, and especially the price of the telephone is very high.

Andrea

             "asterisk"                                                    
             <asterisk at totarot                                             
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                                                                   Subject 
             13/10/2005 16.13          Re: [Asterisk-Users] parameters     
                                       documentation                       

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"> I really hope this project will be implemented, without documentation
evrything is too hard"

Not for the thousands of people that have figured it out.

3Com NBX might be more your speed and plenty of documentation.

> Really strange answer. I am non used to search on playboy.com.
>
> Anyway, if you try to search
> insecure=very
> on www.voip-info.org, you find 742 links , a bit more for me. (I just
want
> to know what it means)
>
> Moreovere, the first 20 links are non accessible at all
>
>
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecure&diff=6

>
> they speak about tiki-pagehistory.php, which appears not to exist.
>
> no other comments about this.
> ************************************************************
>
> I know about one project , "asterisk documentation project"
>
> http://www.asteriskdocs.org
>
> in its home page, the first line is
>
>
>
>
>
>  Great software needs great documentation.
>
>
>
>
> I really hope this project will be implemented, without documentation
> evrything is too hard
>
> Andrea
>
>
>
>
>              "Steve Totaro"
>              <asterisk at totarot
>              echnologies.com>
To
>              Sent by:                  "Asterisk Users Mailing List -
>              asterisk-users-bo         Non-Commercial Discussion"
>              unces at lists.digiu         <asterisk-users at lists.digium.com>
>              m.com
cc
>
>
Subject
>              12/10/2005 14.53          Re: [Asterisk-Users] parameters
>                                        documentation
>
>              Please respond to
>               Asterisk Users
>               Mailing List -
>               Non-Commercial
>                 Discussion
>              <asterisk-users at l
>              ists.digium.com>
>
>
>
>
>
>
>
> www.voip-info.org
>
> > Another trivial question:
> >
> > Is there a "place" where all the parameters are documented ?
> > In example (my example!) I would like to know the meaning of a lot of
> > parameter that can be used in sip.conf,
> >
> > A lot of these keywords are intuitive keywords (i.e.
> NAT=YES/NO;PORT=5060;
> > context=xxxxx) but other are not (at least for me)
> > i.e.:
> >
> > type = peer, friend
> > insecure=very
> > host=dynamic
> >
> > and so on.
> >
> > At last, my need is:
> >
> > Accept a non-registerd sip-strem from a well known ip address (and only
> > from that ip address....)
> >
> > I tried to add a
> >
> > [testsip]
> > ;username=testsip
> > type=friend
> > ;secret=testsip
> > qualify=no
> > port=5060
> > nat=no
> > host=x.y.z.w
> > dtmfmode=rfc2833
> > context=from-internal
> > canreinvite=no
> > callerid="test  sip " <testsip>
> >
> > that would work if the sip would be registered. But the SIP client is
not
> > able to register.
> >
> > I solved using the
> > context = from-sip-external ; Send unknown SIP callers to this context
> >
> > and it works, but I have no more the control about who is sending me
SIP
> > stream (anybody now can use my asterisk box...)
> >
> > any help will be greatly appreciated
> >
> > Andrea
>
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