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I have a problem when dialing from outside line to sip server. I get this output on debug. Could someone give me a hint what could be wrong? == Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to exten 's' == Starting OH323/R30149 at from-pstn,s,1 still failed so falling back to context 'default' How do you make connection between incoming h323 connection and user who is connected to sip server?
I have a problem when dialing from outside line to sip server. I get this output on debug. Could someone give me a hint what could be wrong? == Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to exten 's' == Starting OH323/R30149 at from-pstn,s,1 still failed so falling back to context 'default' How do you make connection between incoming h323 connection and user who is connected to sip server?
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Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060507/b76c113b/attachment.htm
You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribi?:> > Hi all > > I have installed station which support only H323 protocol. I want to > install SIP telephone. Is it possible to call SIP telephone throught > my station > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I don?t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribi?:> > Hi all > > I have installed station which support only H323 protocol. I want to > install SIP telephone. Is it possible to call SIP telephone throught > my station > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribi?:> I don?t have practice to work with Asterisk but I see that is a great soft. > If you have any idea or some config files can you help me > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alberto > Sagredo > Sent: Sunday, May 07, 2006 7:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] H323 to SIP > > You could make a H323 to SIP transport. Before to do that, you need to > have installed and working both chan protocolos on Asterisk. > > aFarhad Ibragimov escribi?: > >> Hi all >> >> I have installed station which support only H323 protocol. I want to >> install SIP telephone. Is it possible to call SIP telephone throught >> my station >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribi?:> I don?t have practice to work with Asterisk but I see that is a greatsoft.> If you have any idea or some config files can you help me > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alberto > Sagredo > Sent: Sunday, May 07, 2006 7:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] H323 to SIP > > You could make a H323 to SIP transport. Before to do that, you need to > have installed and working both chan protocolos on Asterisk. > > aFarhad Ibragimov escribi?: > >> Hi all >> >> I have installed station which support only H323 protocol. I want to >> install SIP telephone. Is it possible to call SIP telephone throught >> my station >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
On Sun, 7 May 2006 19:58:26 +0500, "Farhad Ibragimov" <farhad.i@caspel.com> wrote:> Thanks >Try reading this URL (spanish language): http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 With the page instructions I can call from and to H.323 to every registred SIP/IAX2/H.323 device with my Asterisk box. Good luck,> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alberto > Sagredo > Sent: Sunday, May 07, 2006 7:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] H323 to SIP > > You could begin with: > > http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation > > http://www.voip-info.org/wiki/view/Asterisk+H323+channels > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels > > and much more. > > You need to install chan_h323 module and configure as well as you need > in your application, (if you need gatekeeper functionality maybe you > need to try before GNUGK), and later via extensions make wherever you > need. > > Its a little complicated and you need how to work with asterisk before > doing all this things. > > Regards > > Farhad Ibragimov escribi?: >> I don?t have practice to work with Asterisk but I see that is a great > soft. >> If you have any idea or some config files can you help me >> >> >> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alberto >> Sagredo >> Sent: Sunday, May 07, 2006 7:34 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [Asterisk-Users] H323 to SIP >> >> You could make a H323 to SIP transport. Before to do that, you need to >> have installed and working both chan protocolos on Asterisk. >> >> aFarhad Ibragimov escribi?: >> >>> Hi all >>> >>> I have installed station which support only H323 protocol. I want to >>> install SIP telephone. Is it possible to call SIP telephone throught >>> my station >>> >>> > ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Guillermo V. Salas M Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 1er Piso Tel?fono: 262 8071 Celular : 09 985 5138 Manta - Manab? - Ecuador
Farhad Ibragimov wrote:>I don?t have practice to work with Asterisk but I see that is a great soft. >If you have any idea or some config files can you help me > > > >Asterisk is perfectly documented everywhere on the net. Maybe the first place to visit in order to have working asterisk is www.asterisk.org.Second place is www.voip-info.org If any question arises feel free to email me privately. Tofik Suleymanov