Hi! You started out with a much too complex setup. Start small, test, and then add things step by step - don't configure everything at once!
> Asterisk and 2 Grandstrem phones on a LAN. The * box (liza) is RH9 and > happens to have a firewall connected to the outside but * and the SIP > phones are all on the same LAN. Then remove the nat=yes statements (and also the reinvite= and canreinvite= parts) - for Asterisk your phones are not network address translated. Also I guess you already read somewhere that having Asterisk behind NAT is cause of trouble - but that doesn't matter for your internal testing with a basic setup. After adjusting sip.conf don't forget to "reload" Asterisk. > externip = 10.0.1.198 Remove this for the time being to get your internal test running. Leave messing with externip= for later. > [5702] > type=friend > host=dynamic > context=johnhome > qualify=yes > callerid="John workroom #1" <5702> > mailbox=5702 > disallow=all > allow=ulaw > allow=alaw > [5703] > type=friend > host=dynamic > context=johnhome > reinvite=no > canreinvite=no > qualify=300 > callerid="John workroom #2" <5703> > mailbox=5703 > disallow=all > allow=ulaw > allow=alaw > Found audio format UNKN > Found description format PCMU > Capabilities: us - 524302, them - 285/0, combined - 12 > Non-codec capabilities: us - 1, them - 0, combined - 0 See the disallow and allow statements I added in sip.conf. Cheers, Philipp _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
