just for a test, run "service iptables stop" as root on the asterisk server and then reboot your phones. after that, try again and see if the phones are making communications with asterisk.
you can turn the firewall back on with "service iptables start" jonas kellens wrote: > Hi there, > > this is the first time that I'm building an Asterisk-server. > > I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. > Zaptel is for later, when configuring the POTS-line. Now first internal > communication with SIP. > > Thought it would go easier... > > I have 2 Grandstream IP-phones : BT-201 and GXP-1200. > > These are my settings : > > sip.conf : > /[r...@asterisk asterisk]# cat sip.conf/ > /[general]/ > /bindport=5060/ > /bindaddr = 0.0.0.0/ > > /[BT201]/ > /type=friend/ > /context=intern/ > /host=192.168.4.210/ > /secret=testpaswoord/ > > /[GXP1200]/ > /type=friend/ > /context=intern/ > /host=192.168.4.211/ > /secret=testpaswoord/ > extensions.conf : > /[r...@asterisk asterisk]# cat extensions.conf/ > /[intern]/ > /exten => 210,1,Dial(SIP/BT201)/ > /exten => 211,1,Dial(SIP/GXP1200)/ > Asterisk CLI shows me : > /asterisk*CLI> sip reload/ > /Reloading SIP/ > / == Parsing '/etc/asterisk/sip.conf': Found/ > / == Parsing '/etc/asterisk/users.conf': Found/ > / == Parsing '/etc/asterisk/sip_notify.conf': Found/ > /asterisk*CLI> sip show peers/ > /Name/username Host Dyn Nat ACL Port Status > / > /GXP1200 192.168.4.211 5060 > Unmonitored / > /BT201 192.168.4.210 5060 > Unmonitored / > /2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 > offline]/ > > /asterisk*CLI> dialplan show intern/ > /[ Context 'intern' created by 'pbx_config' ]/ > / '210' => 1. Dial(SIP/BT201) > [pbx_config]/ > / '211' => 1. Dial(SIP/GXP1200) > [pbx_config]/ > > I pick up the phone of the BT201 and dial 211... nothing happens. > I pick up the phone of the GXP1200 and dial 210... nothing happens. > > I would love to have your feedback on this. Where could this problem be > situated ? > > I notice (on the Asterisk CLI) that my SIP-phones do not register. They > have a fixed IP and there account information is set via the web interface. > > Greetingz, > Jonas. > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- 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 _______________________________________________ -- 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
