[Asterisk-Users] parameters documentation

Steve Totaro asterisk at totarotechnologies.com
Wed Oct 12 08:17:01 MST 2005


There is plenty of documentation online for both the 3com and *.  You have
to have good search skills I guess.

3com has the best knowledge base I have seen.
http://knowledgebase.3com.com/ and there are tons of 3com dealers that can
help.

I think you may need to learn some basic networking before learning
asterisk.  NAT is a very basic concept in networking as well as ports such
as 5060 (standard port for SIP).

There is a very steep learning curve for asterisk and networking in general.
If you want to learn it then you need to dig into the wiki and read all the
posts that come across the user's list (well maybe not all  of them).

There are plenty of consultants that you can hire if you are not up to it.


----- Original Message ----- 
From: <asterisk at frameweb.it>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>; <asterisk-users-bounces at lists.digium.com>
Sent: Wednesday, October 12, 2005 10:34 AM
Subject: Re: [Asterisk-Users] parameters documentation


> 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
>              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
>              13/10/2005 16.13          Re: [Asterisk-Users] parameters
>                                        documentation
>
>              Please respond to
>               Asterisk Users
>               Mailing List -
>               Non-Commercial
>                 Discussion
>              <asterisk-users at l
>              ists.digium.com>
>
>
>
>
>
>
> "> 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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