Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particular). I changed iptel's position from friend to peer. I turned on and off nat, I chose different codecs in first place, entered my outward IP as fromdomain and uncommented the register directive with correct values. All I get is two registrations now, but no calls. get a registration effort every 225secs and it succeeds. But when I make a call; channel originate sip/iptel-out/[email protected] Application playback vm/net_ring The call is onlyleft in state ACK for a while. Then asterisk tells me, that it is destroying the sip dialog (long ID) INVITE. Question: Might it be a problem, that my system only knows itself as 192.168.*. Do I need to set something else than externip? Might it be, that my router really blocks certain ports? I can't check it, since it's heavily javascript based and, since I'm blind and the accessibility software for the GUI never really worked on this distro, I don't have a browser to look at it. Do I need to forward port 5060 to my machine specifically (like it is needed for SSH's port 22), or is the mechanism based on: I talk first and the sever gets back to me based on that. This configuration worked for googletalk. I admit, there were problems, but calls were coming through from both sides. Please can someone help me clear up this mess. I'm completely frustrated and don't know what else to do, where else to look. Kindly yours and thanks in advance JUlien
-------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
