Cubox [Part 3]: + Asterisk 1.8 .. fully operational

As mentioned I rebuilt the Cubox with Debian 'squeeze' and got mpd running (see previous post). The bigger hurdle for me was to get Asterisk running and complete migration of an old Centos / Trixbox set-up .. Centos was just too awkward for me.

Failing

I began fooling around trying to migrate the config from the Trixbox server to the default Asterisk config files that came installed via apt-get on squeeze. Whilst both were 1.6 I realised the Trixbox ones were quite modified in set-up by use of freepbx gui to configure them. I got to the point where I was coping over the php agi scripts to find they weren't php5 compliant ... OK this isn't going to work cleanly .. back it up.

Thinking about it:
I'm actually pretty happy with freepbx I just don't want the Centos that Trixbox comes with
My Asterisk config is actually pretty simple

1. 1 x IAX2 trunk 
2. 1 x SIP trunk 
3. 1 x out-/in-bound routes 
4. 1 x Ring Group (ring all extensions .. in NO ANSWER hand over to VM IVR) 
5. 1 x VM IVR (leave message for whom?) 
6. 3 x Extensions (2000 - Dect phone, 2001 - Me, 2002 - Mrs and more to follow I'm sure) 

.. so decided to change tack.

Email voicemail from asterisk

Nothing too flash - simple stuff and existing capability in TrixBox. Only problem I had was not being able to listen to the wav format recording on my cell (Android) or some desktops - the problem turned out to be common and resolving was simply a matter of changing a line of config, e.g. http://www.voip-info.org/wiki/view/Asterisk+Voicemail+as+email+using+sendmail

voicemail.conf

[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
;format=g723sf|wav49|wav
format = wav49|gsm|wav

The only other setting I changed (from default as part of this) was to not store the recording on the server (just email) - basically it is good to pick up the message remotely but having listened to it there is really no need to keep it (IMHO messages age so either listen or delete).

So in review there is now some basic IVR to ask who the message is for and once recorded the recording is emailed to said user. Happy campers.

Android 2.3.4 SIP stack disconnects?

Now that I've moved to NDSL and VOIP for calling I can (via Trixbox) route all inbound calls to a calling group including everyone connected (i.e. everyones Android mobile sip client + the analogue phone via a Linksys / Cisco SPA2102).

That said some I found the Android SIP stack very limiting in that calls drop sounds and then disconnect after a short period of time (maybe 30s - 1 minute). Hunting the Internet I find mention of similar problems with links to resolution [1]. Something to try in the next few days (will update if it resolves my problem).

[1] http://forum.sipsorcery.com/viewtopic.php?t=3029&p=18029#p18098

Migrating to Naked DSL + VOIP

Been a bit quiet lately but some major changes at home to tackle the phone / Internet bills (going down to 1 income has it's issues). Phone line ported to 2talk.co.nz which worked out very sweet. Loving the ability to control calls as and when needed, etc.

So along with 2talk 10 ($10 /month package) we're taking up snap.net.nz's offer for $55 5GB base plan with $2 /month all you eat Google+Youtube.com. Worth a crack. Also by dropping my cell package down a notch I've almosty halved the comms bill. Will update with progress.