Hi Marcus, On Thu, 2006-09-14 at 16:43 +0200, Marcus Carlson wrote: > Try canreinvite=no in your sip.conf file. Then all calls will go via > asterisk.
this solved the "proxy" problem. Now I can see in the logs that all connections goes thru asterisk. The calls are up to 1 ~ 2 minutes, but they still mute after that. Continuing with my setup: > > [ Voip Provider ] ------ (XX) XXXX-XXXX > > x.x.x.x (real world phone number) > > | > > { The Internet } > > | > > 200.x.x.x (Internet IP) > > [linux router] > > 10.0.51.1 > > | > > ------------------------- -> (The Lan) > > | | > > [sip peer 1/client] [asterisk server] > > 10.0.51.3 10.0.51.2 > > > > > > The linux router does Nat/firewall for The Lan. > > sip clients inside the Lan can talk to each other (and asterisk) fine. > > > > The router has port forwarding for IAX[2], SIP, RTP (10000-20000) and > > MGCP to the asterisk box (10.0.51.2). My setup is pretty much the sample config with little things altered. This is the source asterisk 1.2.12. I compiled and packaged (rpm) it myself. No patch or addon package added (like sounds, addons, festival and so forth). Commented options are not included. --------In extension.conf:---------- [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=Zap/g2 TRUNKMSD=1 [from-external] exten => s,1,Answer exten => s,2,Wait(2) exten => s,3,Background(swi/swi) exten => s,4,Goto(from-external,s,1) ; This is for debugging ; Voip line1 (more testing) exten => voip,1,Goto(from-external,s,1) ; Internal Peers #include swi-ramais.conf ; internal use [internal] exten => s,1,Answer() ; Make sure we have ansered the call, Playback will do it, ; but we need to ensure the 2s Wait exten => s,2,Wait(2) ; Wait 2s for a SIP session stablish ;welcome title exten => s,3,Playback(swi/info) ; internal peers #include swi-ramais.conf exten => _9NXXXXXXX,1,Dial(SIP/voip1/${EXTEN:1}) ----------->8------------------- -------------- Now in my sip.conf: ------------------- [general] context=default allowguest=no realm=swi bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=lowdelay externip=<external NAT ip 200.x.x.x> register=<user>:<pass>@voip.net.br:5060/voip1 [voip1] username=<user> type=peer secret=<pass> port=5060 insecure=very host=voip.net.br fromuser=<user> fromdomain=voip.net.br dtmfmode=rfc2833 disallow=all context=from-external allow=ilbc allow=alaw allow=g729 canreinvite=no ; add to keep asterisk owning the call ;internal peers [peer1] context=internal type=friend secret=<some pass> host=dynamic ; more just like above ------------------------>8------------------------------- If this is enough, I can paste the full debug with "sip debug" activated or even paste the sniffing result with ethereal (if necessary). I just dont want to flood the list unnecessary. - Raul Dias _______________________________________________ --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