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 <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] , 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:[EMAIL PROTECTED] 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. > - Mois�s Silva > > On 4/19/05, ruben cuevas rumin <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > > 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. > > > > Rub�n. > > > > On 4/2/05, flavio patria <[EMAIL PROTECTED]> 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 > > > <[EMAIL PROTECTED]> 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, > > > > > > > > Rub�n. > > > > _______________________________________________ > > > > Asterisk-Users mailing list > > > > [email protected] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"; > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
