Inbound calls - YES / outbound calls - NO, help me please


This is the Nth post about this in the entire forum but all i have red didn`t helped me solving my issue.

I have FreePBX and Asterisk latest versions (downloaded few days ago) + a2billing.
I have online Skype number which i was able to setup for my inbound calls. Everything tested and it`s working (tested with IVR saying the time now).

I have 2 skype channels - 1 for inbound and 1 for outbound connection (1 connected user at a time).

Using SIP phone I am connected to my FreePBX/asterisk machine. But I am unable to make outbound calls. Calling between extensions is ok, tested it.

This is my asterisk log:

Peers & Trunk are connected, working. All tested.

Peers & Trunk are connected, working. All tested.

apparently not quite so. That is a problem with your Skype setup. Unfortuntely due to it’s ever changing nature, and it’s “end of line” status:-

(There are other other methods that generally don’t work either) I suggest that you might need support from Digium or Skype directly and not from FreePBX per se.

If you are very explicit about how you installed FreePBX and from what distro or you did it yourself, versions of everything, OS, all that good stuff you have neglected to share as yet :wink: , then perhaps someone here with a similar setup could help you better (sorry the mind-readers here all went to Mars last week, I wonder what they know that we don’t !! )

Perhaps start with a more conventional VSP and a more basic setup and start over again, this time with baby steps, but yes it does work for almost everyone else :wink: .

(A2billing is another notorious “thing that doesn’t work OOTB”)

FYI latest version of asterisk:-
asterisk-11.0.0-beta1.tar.gz 2012-08-10

latest version of FreePBX

)neither work very well :wink: but I hope you see why “latest version” can confuse . . .

I do know that skype for asterisk is going to its end. I just don`t see any reason why not to use it :wink:

The distro is the latest thing that someone needs to helping me solve my problem. Debian, Cent OS, Fedora, FreeBSD or whatever… it doesnt matter :-) My problem is not with the distro, its with the app :slight_smile:
F.Y.I: Ubuntu

versions - latest stable, i believe i said this
F.Y.I: Asterisk
F.Y.I: 2.10.0

All installed by myself.

A2billing - yes, i know that too. Just mention it if it my give some clues :slight_smile:

Another VSP is something i will do. I just want to know ho to fix the skype problem :wink: Changing the VSP is the easiest way. Im not here to make it easy, Im here to learn something!

Ahr… almost forgot! About those mind readers… don`t worry about them :slight_smile: We were traveling together but i took an earlier flight back, because of the skype issue :slight_smile:

If you think that it doesn’t matter how you installed it and on what OS and you think that asterisk is the latest and you installed it on Ubuntu with a squeeze .deb and you think that the problem is the app, then I suggest you get in touch with whoever packaged that version, or maybe the linux fairies, it really DOES matter, all linii are not the same, and FreeBSD is not even linux ;), believe me!!

FYI the latest version of Asterisk 1.6 is as of today 2012-08-15

2012-04-23 asterisk-

as to, then

2010-06-18 asterisk-


2010-07-23 README-
2010-06-29 README-
2010-07-14 README-
2010-08-10 README-
2010-07-23 README-
2010-07-27 README-
2010-09-15 README-
2010-08-24 README-
2010-09-15 README-
2010-11-11 README-
2010-09-21 README-
2010-12-08 README-
2010-11-23 README-
2011-01-18 README-
2011-01-14 README-
2010-12-15 README-
2011-01-18 README-
2011-02-21 README-
2011-02-28 README-
2011-01-18 README-
2011-01-26 README-
2011-02-16 README-
2011-03-16 README-
2011-03-17 README-
2011-04-21 README-
2011-04-26 README-
2011-02-28 README-
2011-06-23 README-
2011-06-28 README-
2011-06-29 README-
2011-06-24 README-
2011-08-05 README-
2011-12-08 README-
2011-12-19 README-
2012-03-15 README-
2012-04-23 README-

(sorry, I am being an ass here, and having too much fun, 2012-04-23 README- will cover everything)

Even non mindreaders won’t call two years ago “latest”, especially as that was before the Skype channel driver was released :slight_smile:

So you will need Asterisk and asterisk-addons in 1.6 (1.8 and above preclude that addon reqirement)

the digium site is quite helpful there, so go there, download it, read the rules and do it.

In case you are tempted to get FreePBX from a similarly outdated source, I suggest you get the “real” latest FreePBX with


Your trouble is if you install it yourself you have to fix it yourself, few will have the patience to do it for you. Not that I don’t applaud you for going that way, I am also an autodidact, unfortunately that will leave you swinginging in the wind until you pull yourself up by your bootstraps (wow, there is a rather incongruent mixed metaphor!), few will help you there, the rewards are great though.

Gently, I suggest you start with a proven and accepted distro first, (to my knowledge as yet you will be limited to Centos for some strange legacy reason) when you get to wear big boy boots, then absoultely, install on Debian, Ubuntu, FreeBSD or even WinBlows, it is all open source and depends pretty well on other open source projectsm good luck with zend on FreeBSD or CPM though if you want commercial modules.

When you have a relatively current and otherwise working sysytem, then is maybe the time to whine about your Digium code, but as previously stated, the Skype code is presumably from Digium and you presumably paid for it, does it not make sense to bug them about their bugs first ?

To my knowledge there is nothing in FreePBX that has any touch on Skype trunking apart from assuming you got it working first.