I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is:
asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no alterations to the default configuration, whatever it is. The asterisk host is configured as 192.168.6.122. The softphone is configured on a separate host with a routable IP on our 216.xxx.xxx/24 netblock. Both networks pass though an internal switch and are firewalled from the outside world by a centos-6.4 based gateway host using IPTables. I have no difficulty in connecting to the asterisk host either by ssh or by https. I have initialised the FreePBX config and have selected the user/device approach as this seems to fit our firm's employee requirements more closely than the extension based configuration. We have several employees who frequently telecommute. For the purposes of testing I have created two users, 11 and 12. I have configured a twinkle user profile for user 12. I can place a call to user 11 from twinkle and I get the IVR message for 'the number you have called is not in service'. I have tried to register Twinkle and this always fails. If I do : # asterisk -rvvvvvvvv CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] CLI> sip show users Username Secret Accountcode Def.Context ACL ForcerPort Which seems to say to me that I have nothing configured albeit I have tried to through FreePBX. At this point I am not trying to get a call out to our PTSN, although I have the FXO port plugged into a live analogue line. What I am trying to understand is the relationship between asterisk devices and users. The twinkle softphone has two lines (1 and 2). It seems to me that I should be able to configure each line as a separate extension and to call one from the other. What I cannot seem to discover is how to do it. Is it possible to do this? How is it done? -- *** E-Mail is NOT a SECURE channel *** James B. Byrne mailto:[email protected] Harte & Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
