[Asterisk-Users] Grandstream Source?

Steve Totaro stotaro at seepu.com
Fri Sep 19 05:02:29 MST 2003


Look at all the time you are wasting flaming people.  just ignore these
questions and get off the high horse.  Do you maintain this list?  If not
then you have no say whatsoever.


----- Original Message ----- 
From: "Steve Creel" <screel at turbs.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, September 18, 2003 6:09 PM
Subject: Re: [Asterisk-Users] Grandstream Source?


> I am NOT a VoIP guru.  I am NOT an Asterisk guru.  I am NOT a telephony
> guru.  Take that as a disclaimer for the information below, as well as to
> say that the best learning comes from reading anything you can get your
> hands on.  The idea of "post any question to the mailing list" works well
> with 10 people.  It scales horribly.  Reading through the archives, you
> will see the same questions asked (and answered) over and over.  At _some_
> point, it's okay to say "I've answered it 15 times, YOU can go look it
> up on YOUR time".  Besides, I'd rather spend 3 hours looking for the
> answer than just ask my question, because I hate looking like an idiot.
>
> This isn't a flame, nor a sarcastic, snide response.  I don't want to
> complain about people asking "what is a ____" if I've never made an
> attempt to answer that question for someone.
>
> On Thu, 18 Sep 2003, PJ Welsh wrote:
>
> >I have to defend us newbies on this.
> >
> >This environment does not facilitate sequential knowledge building! Based
> >on my entry to Asterisk, I should have already known
> >T1/E1/VOIP/SIP/FreeWorld/H.232/X100P/PBX/FXO/FXS/channel bank etc you get
> >the idea (still trying to figure out "skinny"...cisco something, I know).
> >Heck, I'm struggling to get a grip on what and how to use/confiure SIP
> >for linux and keep my hair.
>
> A T1 is technology used to deliver digital data from one device to
> another.  Most of us are familiar with data T1s - 1.544mbps.  When used
> for voice, they can be PRI (primary rate interface) or Channelized T1.  A
> PRI has 23 voice channels and a bearer channel.  The Channelized T1 has 24
> voice channels.  Depending on the specific application, one may be better
> suited than another (or depending on the price).  There are many other
> technical characteristics about a T1, but know we've established what it
> is.
>
> An E1 is used for the same purposes as a T1.  "Which one is it" depends on
> your geographic location - T1 in US, Canada, and Japan (according to a
> telecom dictionary on the shelf here, sorry if misinformed).  Other parts
> of the world use E1.
>
> VoIP refers to the high-level use of an IP network (or IP equipment) to
> deliver telephone service.  Sometimes this means telephone calls from a
> software app on one machine to another software app.  It could mean a call
> from one physical analog phone to another that was connected by way of an
> IP network.  It could refer to an off-premise extension of your desk phone
> to home.
>
> SIP is "session initiated protocol".  There are two parts to VoIP
> protocols - the call setup and the audio stream.  All of the audio is
> handled similarly with most protocols.  The difference is usually in call
> setup.  You can use SIP to call from one phone to another directly,
> without a callmanager, gatekeeper, or any other VoIP equipment.  SIP
> allows IP addresses to be entered and called directly.  SIP seems to be
> best for single-line extensions, "I want to call my brother in _____ ",
> and for most consumer-grade VoIP for home use.  The biggest "user
> experience" thing I can think to mention about SIP is that dialing
> _usually_ (excluding "early dial") works like a cellphone - dial number &
> press "send".
>
> Skinny (or SCCP used interchangably) is Cisco's "Skinny Client Control
> Protocol".  It is a proprietary protocol that Cisco uses in their Call
> Manager system.  The Cisco phones use SCCP to talk to the server (yes,
> like how a SIP phone would use SIP to talk to another phone, or to a SIP
> server).  Because Cisco is Cisco, there is a certain demand to use their
> devices.  To accomodate this, they have offered SIP firmware to load on
> some of their phones.  However, the SIP firmware does not offer all of the
> features of the firmware for SCCP.  Some of this is protocol limitations,
> some is because they didn't include it.  Asterisk's support for SCCP is
> beginning to be functional (no disrespect to those who have put tons of
> time in on it already - "beginning" in that it's beginning to be offered,
> not beginning to be worked on).
>
> FreeWorld is Free World Dialup, or FWD.  Their website,
> www.freeworlddialup.com, says the following:
> Free World Dialup (FWD)  allows you to make free phone calls over
> the Internet using a 'regular' telephone or a computer program.
>
> Free World Dialup does not directly provide access to the
> traditional telephone networks or cellular networks. FWD members
> can only call other FWD members and customers of IP-based service
> providers who have a business relationship with FWD. If you are
> interested in learning about VoIP and would like to setup your own
> personal PBX, give Asterisk a try.
>
> H.232 is a typo, the protocol is H.323.  My understanding is that it is
> essentially the "first-generation" VoIP protocol.  Generally this is
> associated with "older" equipment, or a last-resort for interfacing
> otherwise incompatible equipment.  Netmeeting used to use it, and still
> may.  That's all I know about H.323, and I may be wrong about all of it.
>
>
> An X100P is a Digium FXO card (see FXO quickly explained below).  Digium's
> hardware list (from
> http://www.digium.com/index.php?menu=hardware_products):
> TE410P - 4 port T1/E1 (individually togglable) PCI card
> TDM400P - 4 port FXS PCI card
> T400P - 4 port T1 PCI card
> T100P - Single T1 PCI card
> E100P - Single E1 PCI card
> X100P - Single FXO PCI card
> S100U - Single FXS USB device
>
>
> PBX is an acronym for "Private Branch Exchange".  Consider the phone
> company...  You plug your phone into a circuit from the phone company and
> get dialtone.  You can dial digits and have your call sent to the right
> place.  You can subscribe to various services (call waiting, callerid,
> voicemail, hunting, etc).  If your phone company has a connection to other
> phone companies (which they must), they can connect your call to someone
> outside of their system.  Apply "Private Branch" to that - you provide
> dialtone, you setup and offer services, you handle what happens when you
> dial a number.  If the PBX is connected to the phone company (which it may
> or may not be), you can place a call out to the phone company through the
> PBX.  Generally a PBX is a physical thing, to which other equipment
> connects.  It isn't a phone, it isn't a service you get from your telco,
> it isn't an idea, it isn't a protocol, it isn't a wiring scheme.
>
>
> FXO/FXS - Have you used the phones with lighted keypads?  There must be
> some electricity somewhere to light the keypad, right?  Your phone
> connects to the phone company, and only one of you can put power on the
> line (the phone company does).  Look at it like an electrical extension
> cord (I only know US electric equipment, sorry if the analogy doesn't
> translate).  The electric company supplies electricity to the outlet on
> the wall.  You have two ends on an extension cord - one to connect to the
> electric company, and one that emulates what you got from them.  FXO is
> "Foreign Exchange Office".  Here, office is the same office in "Central
> Office" - the telco.  The phone company provides the FXO signal (the
> electric company provides the outlet) and your phone provides FXS signal
> (your radio has a plug to connect to the electric company's outlet).
>
> To receive an existing analog line, you need an FXO card (like the X100P).
> When you set it up, you configure it to use FXS signalling (it should look
> like a phone to the other end).  To plug in an analog phone, you need an
> FXS card (and you configure it to use FXO signalling - it should look like
> the phone company to the phone).
>
> If I want 24 analog extensions, there is no change I'll put 8 TDM400P PCI
> cards in a chassis.  Remember that the T1 holds 24? (Yes, channelized T1
> will hold 24, PRI will hold 23 - don't worry about that here)  A channel
> bank (generally) has 24 analog ports and 1 T1 port.  It is essentially a
> breakout box, used to assemble the 24 channels on a T1.  You can get a
> combination of FXO and FXS devices in the channel bank.  Maybe you want to
> bring in 8 analog lines from the phone company (The phone company uses FXS
> hardware and FXO signalling - you'll need an FXO device and FXS
> signalling) and 16 analog extensions.  You can get a channel bank that has
> 8 FXO ports and 16 FXS ports.
>
>
> >You don't start off with a prerequisite of knowledge to join like a
> >class/school. You don't have the you-must-have-asterisk-101-before going
> >to asterisk-102 before you can join this list. You have a forum that is
> >GENERAL.
>
> No, but no one wakes up every morning hoping to answer the same 5
> questions they answered yesterday.  The prerequisite is that people be
> interested and motivated to learn.
>
> >I would like to a better effort to provide a more sensible way to start
> >helping us newbies. I have to say that the Digium handbook helped a
> >little, but not much. I have googled till I couldn't see straight. I just
> >don't yet have the "big picture" that most of you do. I couldn't even
> >tell you if I need a channel bank or a channel changer ;) at this point.
> >
> >A group of you seem to expect people to have a knowledge base that allows
> >for entering keywords to google. I don't know those keywords. You know
> >the context to search for when someone says I'm having a problem with
> >insert-thing-here.
>
> Yes, if a certain question comes up, someone on the list might be able to
> easily send you in the right direction.  Asking to be sent in the right
> direction, not to be handed the solution, would probably yield good
> responses.  I'd rather say "search for canreinvite and 7960" than feel
> like if I offer any help, it's going to be expected that I produce a
> complete explanation of the problem, why it happens, and to fix it for
> someone.
>
> >Instead of the usual, "Search the archives". It would be more helpfull to
> >give a hint on what to search for. I could search for SIP and get back
> >several hundred "answers". Then I have to figure out where that answer
> >lies in the series of possible answers. Then I have to somehow figure out
> >if it works.
>
> Describe _exactly_ what you've run up against, offer relevant information,
> and ask where to look for more information.  Sometimes I realize the
> solution when I'm putting all the information together to email someone
> about a problem.
>
> >As most of you teachers (past and present) should know, not all of us
> >learn the same. Some people just "get" written material. Some NEED the
> >"spoon" to make it to the next level. Some need the hands-on experience
> >and other's just can't learn any more than they have already know(those
> >people are not likely on this list, however).
>
> In a group of people, some will prefer to sit down by themselves and read
> every word written about a topic.  Others would prefer to sit down and
> talk about it with other people.  Others would prefer to sit and play
> solitare on the computer while someone talks at them about it.
>
> If you've done alot of reading and want to talk about what you've read and
> questions you have, use the IRC channel.  Until using Asterisk, I had
> never used IRC, despite being around it (and involved with linux
> projects) for years.  Seriously, look at the IRC channel to talk about
> things.
>
> >You do realize that the http://www.asterisk.org/index.php?menu=support
> >lists the mailing list first for support, don't you. In fact, you have to
> >go to the second page before you even see the google reference. More a
> >few people tend to look for the FIRST way to get help not ALL ways to get
> >help...
> ><flame suit on>
> >
> >
> >On Thu, Sep 18, 2003 at 08:31:59PM +0200, Dave Cotton wrote:
> >...
> >> Absolutely agree with you Steve.  I left teachers training college in
> >> 1970. I shock some teachers when I said that in all the years since I
> >> haven't taught anyone anything. I've just enabled them to learn.
> >> The problem is that in most national education systems the teacher is
> >> expected to provide the answers to pass some test at the end of the
> >> course. Thinking is not part of the curriculum.
> >> --
> >> Dave Cotton <dcotton at linuxautrement.com>
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___________________________________________________________
> Steve Creel                                screel at turbs.com
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>




More information about the asterisk-users mailing list