Late Night PC

One last short one about my Asterisk configuration files. I hate it when someone pastes a few lines and calls it an example, so I'm going to give the supporting parts of other config files that are needed to make my extensions.conf work. I don't think the complete files are necessary, since they're almost exactly the same as the samples that come with Asterisk. I authenticate all users based on the login and password they use for voicemail. Here's the section of voicemail.conf that creates the mailboxes and assigns the passwords. [default] 77 => 71284,Rob's Mailbox,rob@example.com 63 => 71284,Alex' Mailbox,alex@example.com 71 => 71284,Candace's Mailbox,candace@example.com 72 => 71284,Jeff's Mailbox,jeff@example.com I've started them all off with the same password but apparently the voicemail module can rewrite voicemail.conf to change passwords. The email addresses don't do anything here yet. Apparently there's a simple configuartion change you can make to enable email notifications or even enable sending the entire voicemail over email but I have yet to try that out. Here's the entirety of my meetme.conf file. ; ; Configuration file for MeetMe simple conference rooms for Asterisk of course. ; ; This configuration file is read every time you call app meetme() [general] ;audiobuffers=32 [rooms] ; ; Usage is conf => confno[,pin][,adminpin] conf => 101 Remember that in order MeetMe to work, you need a timer source, either a hardware solution or the ztdummy software timer module. I think this set up for MeetMe could be useful for people with a scheduled conference call but could also be useful to connect with over a softphone. Then people could hang out in the conference room and possibly use it as a substitute for Ventrillo or TeamSpeak. My sip.conf file is pretty much the standard sample sip.conf for now as well. The one change I made is the one that Les.net tells you to make in their sample code for peering with Asterisk. It looks something like this [general] ... some stuff ... register => <some code>:@did.voip.les.net/<special code> ... some stuff ... [lesnet_peer] type=friend host=did.voip.les.net dtmfmode=rfc2833 insecure=very disallow=all allow=ulaw context=lesnet-incoming ; incoming DID calls will arrive in the lesnet-incoming context Of course this is censored but you get the idea - if you use Les.net then you can look up the exact code for your [general] context and if you don't use Les.net then it's going to be specific to your service provider. This does tie back in to a section in my extensions.conf from yesterday though. The context=lesnet-incoming line brings incoming SIP calls from the DID to the [lesnet-incoming] context in my extensions.conf. From there I send the calls to [incoming] with a Goto() like this. [lesnet-incoming] exten => _X.,1,Answer exten => _X.,n,Goto(incoming,s,1) I didn't mention it yesterday because it ties back to sip.conf. This kind of separation lets me have a different context to answer calls from each different source. Once anything unique to that source is taken care of, I can route the call in to the [incoming] context so that everything is handled the same after that point. I think I've had enough messing with Asterisk for a little while. Though my ATA should show up in just a few days...