Hi Noah -
I got everything to load via ftp. The phone appears to correctly boot from the config files. I also put the latest firmware there and the phone sucessfully loaded it.
For some reason, the phone and * don't see each other. This is the part
that confuses me. Any clues as to why the phone won't register?
It's not often I get to address someone with my own name. Hi Noah, I'm Noah. I apologize if someone has already answered your question - I'm writing this on a train with no connection.
There are a few things you need to do:
1. Set up a <macaddress>.cfg file for each phone that you want to configure.
2. Set up a separate phone.cfg file on your FTP server for each of your phones.
3. Set up the sip.cfg file (probably just one) on your FTP server
4. Add a configuration for each of your phones to /etc/asterisk/sip.conf
5. Add an extension to dial each one of your phones in /etc/asterisk/extensions.conf
In detail:
1. On the the FTP server, your should have a file for each one of your phones named <macaddress>.cfg. You can copy from the default 000000000000.cfg file. In that file, you should have a line that looks like:
<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone1.cfg, sip.cfg, ipmid.cfg" MISC_FILES="" LOG_FILE_DIRECTORY=""/>
Adjust the phone1.cfg file to be something descriptive for each phone that you want to configure.
2. In the phone1.cfg file you should have a line that looks like:
<reg reg.1.displayName="" reg.1.address="" reg.1.label="" reg.1.type="private" reg.1.thirdPartyName="" reg.1.auth.userId="" reg.1.auth.password="" reg.1.server.1.address="" .../>
The three things you NEED to change are:
reg.X.auth.userId <-------- This should correspond to what you set later in sip.conf
reg.X.auth.password <--------- This should correspond to what you set later in sip.conf
reg.X.server.1.address <--------- This is the address of your asterisk server - make sure it is reachable by your phones
3. In the sip.cfg file, you should have a line that looks like this:
<server voIpProt.server.1.address="" voIpProt.server.1.port="" ... />
Change these two values to the address of your asterisk server, and the sip port (5060 by default)
4. In /etc/asterisk/sip.conf you'll need to add an entry for each phone. It should look something like this:
[device_name] <--------- whatever you want, but needs to be the same as auth.userId above
type=friend
secret=<password> <-------- whatever you want, but the same as auth.password above
callerid=<whatever>
host=dynamic <------------- If you're using DHCP, or the address if it's static
dtmfmode=inband
mailbox=<mailboxnumber>@<voicemailcontext> <-------- should correspond to setup in voicemail.conf
context=<context_name> <---------whatever you want, but you must have it in extensions.conf - see below
disallow=all
allow=ulaw <------ or whatever codecs you want to use
5. In /etc/asterisk/extensions.conf you'll need to add an extension for each phone device that you set in sip.conf (You can also use regex and/or macros to write just one entry that will match all your phones). A basic entry will look like this:
[context_name] <---------- the same one you specified above in sip.conf
exten => 100,1,Dial(SIP/100,20) <------ Dial the sip device for 20 seconds
exten => 100,2,Hangup
After this, you'll want to add voicemail and you'll probably want to set up your other line appearances (hint: you'll probably want to use CheckGroup and SetGroup to disable call waiting). After all that you can do fun things like set up MWI and intercom. Those are detailed on the WIKI.
Thanks, Noah
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