Number 0Nxxxxxxx is currently OFFLINE

This is something that's been bugging me for a little while with my iax2 trunk to 2talk. Fundamentally I'm just a bit sow to get how it all hangs together. Generally it works out of the box so I don't 'need' to understand it to use it ... except it's good to know how it hangs together when things get weird.

That said some kind of recent change (possibly with 2talk plus account set-up) my registration died and 2talk showed me as 'OFFLINE' .. hmm.

*CLI> iax2 show registry
Host                  dnsmgr  Username    Perceived             Refresh  State
202.180.76.166:4569   N       0Nxxxxxx               60  Timeout
202.180.76.166:4569   N       0Nxxxxxx               60  Timeout

Ok so I'm not registered - but, I can make outbound calls ... odd? No - this is a clear description:

The only purpose of registration is to allow a device to identify its location on the network, so that Asterisk knows where to send calls intended for that device.

Thank you Mr O'Reilly http://ofps.oreilly.com/titles/9780596517342/asterisk-DeviceConfig.html

OK so I didn't change anything so perhaps I can force a registry reset / refresh, etc 

*CLI> iax2 unregister
Usage: iax2 unregister
       Unregister (force expiration) an IAX2 peer from the registry.

Hmm .. not helpful - I can't figure out what the peername is .. a few attempts from iax2 show peers to get the name don't seem to help. So I default to a fallback option - change the registry string to something wrong, reload asterisk, then change in back .. maybe. 

So I choose (rather fortuitously)  to replace:

0Nxxxxxx:@iax.2talk.co.nz 

with

0Nxxxxxx:@iax.plus.2talk.co.nz 

and  guess what, it works .. well maybe a change to the 2talk+ accounts? I can't find a reference say here http://blog.2talk.co.nz/iax2.html or  here http://www.2talk.co.nz/2talkplus/ . I've created a ticket so will see what they say.

*UPDATE* apparently if you're on 2talk+ you should always use .plus .. so looks like the last related change got pushed through and they've stopped supporting registration on the old domain. OK.

Cubox [Part 3]: Revisited .. Asterisk 1.8 issues

Turns out there is no sound on out-bound calls so my claims of success were a little premature.

I've been trying to diagnose - asterisk debugging is not super easy to follow and nothing cries PANIC. A good way to test was to use call-recording feature on 2talk and then test outbound call to mobile (using a friend to talk on one and you on the other is easiest ;). If both voices are recorded then obviously 2talk is receiving both and having issues thereafter (simple and obvious but seems to confirm that 2talk are passing on what they receive).

Searching around I see some notes on similar (no/extremely choppy audio on outbound call issues) such as https://issues.asterisk.org/jira/browse/ASTERISK-16258

Some things to try to resolve this.

Also review iax show netstat maybe the lack of sound is due to issues with load encoding on the fly, etc.

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.

Creating sounds files for Asterisk with Audacity

Have set-up a IVR service on my home asterisk set-up so that people ringing our home phone number can choose who to leave a message for (i.e select 1 for my VM extension, etc).

For the link in title and the reference to Audacity was the best solution (sweep wasn't playing nicely with pulseaudio and I was a bit over figuring out why, and sox conversion from basic Sound Recorder files wasn't working as obviously as examples suggested it might). I have played a bit with Audacity as a multi-track recorder for guitar session fooling and was pleased to find all paths seemed to suggest it was the answer.

I was trying to be too clever and set all the options myself - then found this link and presto - one last trick however was to leave it sample format at 32-bit float, mono and 8000Hz rate as mentioned (if you set the project rate to 8000 Hz I found I needed to change it to playback) .. then export as 'WAV (Microsoft) singed 16 bit PCM' sorted.

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.