[Asterisk-Users] Re: Starting with Asterisk-SIP

jltaylor jltaylor at metrotel.net
Thu Apr 21 13:30:47 MST 2005 

Don't get many hugs around here...

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of ruben
cuevas rumin
Sent: Thursday, April 21, 2005 2:28 PM
To: Moises Silva
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Starting with Asterisk-SIP


Hi Mosies,

Thanks for your help, now I have a SIP server using asterisk and I can
communicate my two SIP clients with asterisk in the middle :).

This is the first step, but I have to work a lot of yet, so I think I
will disturbe you and the other people in the list (I'm sorry).

Thank you very much for your help.

"Un saludo y un abrazo" ;). (It's an spanish expresion)

                               Rubén.

On 4/20/05, Moises Silva <moises.silva at gmail.com> wrote:
> Hi again Ruben. Well, it would be good idea to put here what do you
> have in your extensions.conf. Actually i have only "includes" in this
> file, several statements like this:
>
> #include /var/lib/pavoz/extengeneral.iss
>
> So, its easier its administration, but for a simple test you can do
> this in extensions.conf:
>
> [testdialplan]
> exten => _.,1,Dial(SIP/${EXTEN},40,r)
> exten => _.,2,Hangup();
>
> then, in sip.conf:
> [general]
> port=5060
> bindaddr=0.0.0.0
> localnet=192.168.1.0/24 ; here you need your net config net_addr/mask
> tos=lowdelay
> tos=184
> defaultexpirey=120
> disallow=all
> allow=ilbc
> allow=alaw
> allow=ulaw
> defaultcontext=incoming_iss
>
> [15]
> type=friend
> secret=adminpass
> host=dynamic
> nat=no
> dtmfmode=info
> canreinvite=yes
> qualify=yes
> context=testdialplan
>
> [12]
> type=friend
> secret=adminpass
> host=dynamic
> nat=no
> dtmfmode=info
> canreinvite=yes
> qualify=yes
> context=testdialplan
>
> So, you need 2 sip phones (can use kphone) with SIP username 12 and
> 15, using password 'adminpass'.
>
> This is what will happend:
>
> - When you start kphone's, or any other SIP phone, the phones will,
> they will try to make a SIP register with the server that you specify,
> so you have to configure the phones yo try a register in the Asterisk
> Box IP. Asterisk will receive its request for registry and will check
> that the username and secret exists in the file sip.conf, if exists,
> will save the registry and then Asterisk and the phones will be
> connected. Now, when you dial from any sip user, the number will be
> sent to Asterisk, and asterisk will try to find a match in the dialed
> pattern in the context that the SIP entry specifies (in this case the
> parameter context=testdialplan), so , for example, if you dial "12"
> from sip user "15", the 12 will match in the pattern "_.", because the
> dot match anything, you can be more specific an put in extensions.conf
> "_XX", instead of "_.", and Asterisk will only match when you dial a
> number of 2 digits length, and that digits are 0-9 (the X means 0-9).
> You can read more about this in:
>
> http://voip-info.org/wiki-Asterisk+config+extensions.conf
>
> Once the pattern is matched, Asterisk will attempt to execute the
> commands that are there, in this case a Dial() command, that say "Open
> a Channel type SIP, and try to dial to the ${EXTENSION}", ${EXTENSION}
> is a special var, you can read more about asterisk vars in:
>
> http://voip-info.org/wiki-Asterisk+variables
>
> So it will try to dial to a SIP user with the dialed extensión.
>
> So that all, it should work for a small test.
>
> I have studied in Universidad de Guadalajara, in Guadalajara, México.
> Any other people from México here???? :-)
>
> Good Look!
>
>
> On 4/19/05, ruben cuevas rumin <rcuevasrumin at gmail.com> wrote:
> > Hi Moises,
> >
> > Thanks for the reply, and thanks Dana too.
> >
> > I  know that I can to communicate two SIPs phones without Asterisk in
> > the middle. But this isn't my final objective, This is the first step
> > in my project, it mean, I firstly want make works a simple testbed
> > (the one I described in the previous mail), and then step by step
> > configure more difficult testbed.
> > So if you, please, could help me to configure this simple test, I'm
> > will be happy :).
> > I think my problem is the dial plan in the extensions.conf.
> >
> > Ah, I'm studing electronics and comunnication eng, in the University
> > Carlos III of Madrid. Congratulations for your graduation, I hope end
> > in September of this year.
> > Which University do you have study?
> >
> > Best Regards and thank you for your help.
> >
> > On 4/19/05, Moises Silva <moises.silva at gmail.com> wrote:
> > > Hi Ruben. You can make a direct IP call. If the 2 sip phones can ping
> > > each other (that is, both are reachable in the network), then in
> > > kphone select the option File > New Call, then type
> > > sip:number at sipdeviceip , the 'number' is the number wich is configured
> > > in kphone, sipdeviceip will be the IP of the machine that is running
> > > the kphone application. Note that this kind of call does not have
> > > nothing to do with Asterisk, the phones are using sip protocol without
> > > asterisk in the middle. When kphone makes a register to asterisk, then
> > > you dont need to specify sip:blahblah at blahbla... you only dial a
> > > number and the number is immediatly sent to asterisk wich routes the
> > > call where the dialplan says.
> > >
> > > Ah, and by the way, where do you study? i just graduate of electronics
> > > and como eng. too :-)
> > >
> > > Good look.
> > > - Moisés Silva
> > >
> > > On 4/19/05, ruben cuevas rumin <rcuevasrumin at gmail.com> wrote:
> > > > Hi Flavio,
> > > >
> > > > I asked for help to start with asterisk some weeks ago.
> > > > Thanks for your help and thanks to other people who reply my mail.
> > > >
> > > > At this moment I have configured asterisk and I have two clients (
I'm
> > > > using Kphone software like SIP client), the asterisk regist
correctly
> > > > the clients, it's mean, the SIP register works fine. But I can't
> > > > stablish a connection between client 1 and client 2.
> > > >
> > > > Mi test is very simple, I have the clients and the asterisk in the
> > > > same LAN. I would like to stablish a SIP connection between the
> > > > clients. So in kphone at client 1 I execute: sip:client2 at ip_client2.
> > > > But this doesn't work. I think my problem is the dialplan.
> > > >
> > > > I would like to know if for this simple test (communication using IP
> > > > address directly) , need I a dialplan or no??? And if I need a
> > > > dialplan, where I could obtain any example of a extension.conf file
> > > > for this simple test. (because I only find examples for other more
> > > > difficult implementations).
> > > >
> > > > It would be great if you, flavio, or other people could help me.
> > > >
> > > > Thanks in advance.
> > > >
> > > > Best Regards.
> > > >
> > > >                              Rubén.
> > > >
> > > > On 4/2/05, flavio patria <flavio.patria at gmail.com> wrote:
> > > > > At the URL http://www.voip-info.org may find some examples.
> > > > >
> > > > > Gettin' started
> > > > > First of all you must define a "possible" dialplan that you can
> > > > > configure in the file extensions.conf. Dialplan may include
several
> > > > > options, just like a simple comunication between two softphone(for
> > > > > example Sjphone) using SIP through the Asterisk PBX.
> > > > > After this, you must define setting about the other configuration
> > > > > files (.conf, like sip.conf.. etc..)related to the dialplan
defined..
> > > > > and so on...
> > > > >
> > > > > However you must easily find several interesting examples over
> > > > > Internet if you search them^_^
> > > > >
> > > > > I am an Electronic Engineer student too ^_^
> > > > >
> > > > > bye
> > > > > flx
> > > > >
> > > > > On Sat, 2 Apr 2005 18:24:17 +0200, ruben cuevas rumin
> > > > > <rcuevasrumin at gmail.com> wrote:
> > > > > > Hi all,
> > > > > >
> > > > > > I'm a Telecomunication Engeenering student. I have to develop a
VoIP
> > > > > > apliccation using SIP protocol. I have to develop the SIP
Server, and
> > > > > > the SIP clients.
> > > > > >
> > > > > > I think I can use Asterisk for this issue. I have installed it
and I
> > > > > > have run it, but I don't know how I have to configure it.
> > > > > >
> > > > > > I have read the documentation, but It's so much big and I don't
know
> > > > > > what I have to do.
> > > > > >
> > > > > > Someone could tell me what configuration files have I to use,
and what
> > > > > > have I to put in this files?. If is it posible, I would like
someone
> > > > > > send me some simple examples of this files.
> > > > > >
> > > > > > It would be wonderful if someone could help me.
> > > > > >
> > > > > > Thanks in advance.
> > > > > >
> > > > > > Best Regards,
> > > > > >
> > > > > >                       Rubén.
> > > > > > _______________________________________________
> > > > > > Asterisk-Users mailing list
> > > > > > Asterisk-Users at lists.digium.com
> > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > > > > >
> > > > >
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> > >
> > > --
> > > "Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"
> > >
> >
>
> --
> "Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"
>
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