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I think that if the TFTP server is unavailable it will just use the previous config, so it’s possible that you can get away without it, but I haven’t tried. It’s a small and full featured DNS server which also includes a DHCP & TFTP server which are easy to configure and it’s almost certainly packaged for your distro.You should also (temporarily) disable any other DHCP servers on your local network so that dnsmasq is the only thing offering DHCP addresses. UTF-8 [email protected]:~ $ sudo vim /etc/[email protected]:~ $ sudo locale-gen Generating locales (this might take a while)... Would you like to answer one of these unanswered questions instead?
I was going to post this all in the forums, but I've been waiting three days now for a moderator to finalize my account so I'm not sure how long that is going to go on. Inside the Elastix GUI under networking it shows the correct IP (10.0.0.4).
But when I SSH into it, the Welcome Message URL is always a different DHCP address. In the SEP, I have set the username, password, ssh Access to 1, and ssh Port to 22 (under vendor Config). I've got an Aastra phone on it's way, I've configed dozens of those and will be assigning it EXT 1002.
But I really need to get these two Cisco's working too.
My eventual goal is to get the "free SIP trunk" with IPComms to work, but I can't get past the physical phone problem. Modified and added: [general] tcpenable=yes tcpbindaddr=0.0.0.0 Watching the elastix*CLI at verbose 3, I never see anything come in for these extensions.
I have gotten a softphone to work (Zoiper) on two extensions, but found using two laptops was too cumbersome so ordered these phones off Ebay. Installed Asterisk and made ext-2-ext calls with Zoiper 2. Got the firmware on the 7942G's to update to SIP42.9-4-2SR3-1S 4. On the phones themselves, there is some strangeness: SIP General Configuration any line: if I use USECALLMANAGER for the line's Proxy, #5 (Proxy Address) and #6 (Proxy Port) both show blank More strangeness: I have the Free PBX VM set to a static IP. I've read that there may need to be some patch for this, but everything I find is 3-4 years old and talking about versions 1.6, even on voip-info.