hi guys, i am getting today my dev kit with fxo and fxs boards. i intend to do the following : 1) be able to route an incoming call from the pstn fxo port to an ip (answering with netmeeting or anyother sip softphone) 2) be able to call from netmeeting to my pstn fxo port to place calls. questions : how can i do this ? what are the commands for this simple setup ? how can i place calls using a webbrowser (explorer, etc ?) can i use messenger to call to call my pstn port ? can i translate h323 to sip and viceversa ? thanks, francisco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040202/e06283c4/attachment.htm
On Sun, 1 Feb 2004, Francisco Perez-Landaeta wrote:> i am getting today my dev kit with fxo and fxs boards. i intend to do > the following : > > 1) be able to route an incoming call from the pstn fxo port to an ip > (answering with netmeeting or anyother sip softphone) > > 2) be able to call from netmeeting to my pstn fxo port to place calls. > > questions : > > how can i do this ? what are the commands for this simple setup ?get asterisk (cvs or tarball). make. make install. make samples. Now you'll have a "working" demo setup which you can modify to with with your sip softphone. I don't know what you'll have to do to make the fxo and fxs boards work; I haven't played with any yet. Modifications you'll need to make: - add a service definition in sip.conf so that your softphone can log in to * using SIP. There are examples in the demo sip.conf file. - edit extensions.conf. Write an extension which makes calls arriving on the fxs port get dialed to your sip phone. Also write an extension which lets you dial out the fxs port to the pstn. Again, there are examples in the file. check out the wiki, www.voip-info.org, for information about what goes into sip.conf and extensions.conf.> how can i place calls using a webbrowser (explorer, etc ?)there are IAX and SIP applets around, but I can't recommend any (haven't tried 'em)> can i use messenger to call to call my pstn port ?if it'll connect to *, then the rest of the magic happens when you write the appropriate lines in extensions.conf.> can i translate h323 to sip and viceversa ?I think so.. I believe * accepts h323 connections as well as SIP, and it would have to un-pack one protocol, possibly transcode to a different codec, and then re-pack for transmission in the other protocol. Greg
Date: Sun, 1 Feb 2004 18:24:51 -0400 Subject: [Asterisk-Users] setting up ---- newbie Reply-To: asterisk-users@lists.digium.com This is a multi-part message in MIME format. ------=_NextPart_000_00A4_01C3E8F0.AED1DC90 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable |questions : |how can i do this ? what are the commands for this simple setup ? |how can i place calls using a webbrowser (explorer, etc ?) |can i use messenger to call to call my pstn port ? |can i translate h323 to sip and viceversa ? As a newbie I will introduce you to go to www.voip.org web site to read some of the newbie articles and also the documentation on www.digium.com. There is a lot info which will definitely answer your above questions. When you are stuck with your work please do not hesitate to answer specific questions. No doubt everyone here on the list will lend a helping hand. -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040202/6bc4639a/smime.bin