Asterweb is a web based gui for configuring Asterisk's realtime configuration. Specifically, it
will help manage the sip_buddies, voicemail_users, and extensions tables. It is written in PHP.
- Re-Orders extension priorities when one is inserted in place of another.
- Add/Edit SIP user and their voicemail box in one screen.
- Supports modules. Currently there is a simple techsupport. Polycom phone provisioning is in the works.
- Has a contacts page that can read a custom table and dial the number for you. (best with Cisco or polycom phone)
- SIP Users can login with their username/password to check voicemail. (requires some chmod stuff)
In order for a user to have access to their voicemail, something must be done to the files in /var/spool/asterisk/voicemail/.
I added the command "chmod -R a+rw /var/spool/asterisk/voicemail/default/" in a script that is called whenever a
voicemail is left. That script is referenced in voicemail.conf as "externnotify=". This will allow anyone to read/write
the voicemail files. No one has a login to my asterisk server so I don't care. You might want to set up a group or something.
The contacts section is driven by a table that does not come with asterisk. Any table with a name and number column can
be configured to work. I created a table that also streams a directory listing for my polycom phones. This will be incorporated
into the polycom provisioning module I am working on. The contacts page is only for SIP users. It requires for someone
to login so that it knows who wants to make the call. When the user clicks "call" it rings their phone, then calls the
number. If you are using a polycom phone, it will ring-answer the phone. Meaning the phone will ring once, then
automatically pick up. It saves the user from having to answer their phone. It is done by adding
SetVar: ALERT_INFO="Ring Answer" to the call file. I believe this will work with Cisco phone, but have not tried.