canreinvite defaults to yes, whether specified or not. http://www.voip-info.org/wiki/view/tips
If you follow these directions adapting to your particular circumstances and it doesn't work, post your whole sip.conf Start asterisk with verbose set to 3 or so and turn on sip debugging. I get somewhere in the debug, you will see local NAT IPs that don't belong there, or it will just work. Thanks, Steve Totaro On Thu, Oct 16, 2008 at 12:12 AM, Steve Totaro <[EMAIL PROTECTED]> wrote: > Change all canreinvites to no. > > > > On Wed, Oct 15, 2008 at 9:37 PM, GNUbie <[EMAIL PROTECTED]> wrote: >> Hello Karsten, >> >> On Tue, Oct 14, 2008 at 12:26 AM, Karsten Wemheuer <[EMAIL PROTECTED]> wrote: >>> >>> Please post Your sip.conf. >>> Which IP-Address do You configure in the snom for Your asterisk? (eth0 >>> or eth1)? >> >> The SNOM 300 is using the NET interface beside the DC 5V port to >> connect to the LAN. >> >> The Asterisk box is using the eth1 to connect to the LAN. >> >> As per your instruction, below is my /etc/asterisk/sip.conf : >> >> - - - < s n i p > - - - >> >> [general] >> realm=pbx.domain.com >> bindport=5060 >> bindaddr=0.0.0.0 >> rtptimeout=60 >> disallow=all >> allow=ulaw >> allow=alaw >> allow=gsm >> externip=pbx.domain.com >> localnet=192.168.101.0/255.255.255.0 >> jbforce=yes >> allowtransfers=yes >> maxexpiry=3600 >> minexpiry=1800 >> videosupport=no >> >> [internal-phones](!) >> type=friend >> host=dynamic >> context=family >> dtmfmode=rfc2833 >> insecure=port,invite >> canreinvite=no >> nat=no >> qualify=yes >> port=5060 >> >> [102](internal-phones) >> username=102 >> secret=102 >> callerid="GNUbie"<102> >> [EMAIL PROTECTED] >> >> - - - < s n i p > - - - >> >> Thank you in advance. >> >> Regards, >> >> GNUbie >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users