Julien Claassen wrote: > 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? > Does the server see your sip client at 192.168.*.*? that would be a problem. > 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. > It's possible that the router is not SIP friendly or there is a setting to allow sip on it. I can not tell as I don't know what router you are using. > 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. > Should not need any forwards. However the router could be firewalling some ports, like the rtp ports. You need to ask what ports are needed for rtp.
Lyle Giese LCR Computer Services, Inc. > 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
