>----- Original Message ----- >From: "Herv� Thibaud" <[EMAIL PROTECTED]> >To: "asterisk-users" <[EMAIL PROTECTED]> >Sent: Sunday, June 22, 2003 8:13 AM >Subject: [Asterisk-Users] asteisk, sip & NAT
>hi >My stations are behinds a firewall, the system is windows 2000 & 98, i >use sjphone >aterisk is on the internet gateway where is the firewall Shorewall the >system is linux debian (sid) kernel 2.4.20 >j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) >to write my sip.conf but i can't call an external sip user. (an external >user can call me) This sound like the problem that I've been having this weekend. My setup is a Snom100 and X-Lite connected to my * box, and the same box is the NAT gateway for the devices. I could have external users call in no problem at all, but when I tried to call out I got about 1/1.5 seconds of audio and then all incoming audio died. the other end could hear me, however. It turned out to be the fact that * sending reinvite requests to fwd, which was then trying to connect directly to the snom100, and, obviously, failing because it's behind NAT. After much hair-pulling from myself and Andy, I stumbled across an unrelated post that pointed me to the 'canreinvite=no' option. I stuck this in the [fwd.pulver.com] section of the sip.conf file and magically, it all worked! Maybe, just maybe, it'll work for you too :) Jon _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
