Mike, thank you for your reply.
However I do not have the option of a DHCP-server. On the network where Asterisk needs to be implemented all is configured statically, so also the IP-phones need to be statically assigned an IP-address. Surely this can not be thé problem... Greetingz, Jonas. On Mon, 2009-04-13 at 12:28 -0400, Michael van der Stoop wrote: > > 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 > > I had the same issue. I set the hosts to dynamic and and explicitly set > their IP's via a dhcp server using their MAC addresses. The phones > registered and all is well. > > Regards, > Mike > > _______________________________________________ > -- 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
