Topic: freepbx to skype.

[tropicalview]

Hi all,

I searched around a little to connect my freepbx to skype.
the reason is that the local providers closed down the VoIP protocols to get to online carrieres.

I came to these alternatives:

-> sisky (windows based, working now for me, busy with test (working on a virtual machine in the same box))
-> uplink (windows based, connecting to the skype application)
-> http://www.rsdevs.com/psgw_linux.shtml#specs (they have a variaity of products on windows and linux base.
->chanskype (linux based, but needs a GUI)


up to now i only put on sth sisky and it's working.
I think uplink is also easy to setup.

Now i want to post here to ask if other people has good / bad expirence with a setup like that and any tip is more than welkome.
If you want more info about how i have done this, just ask me.

[tropicalview]

Hi All,

i did test with the 2 options:

1 sisky
2 uplink

the sisky installation is really easy to install, and configure,
it works almost directly out of the box. and has great features like call logs with costs, and settings by web interface.
but the quality of the call is poor, the person that's on the line will not hear you good, the frequency of your voice will varry and looooooooooong delays.
I don't know if i had bad luck, but i tried everything to get it working good (as installing directx 9).
afterall i tried a couple of time to reach the support by phone / skype / chat but they where closed / not available all the time.

because this solution did not what we hoped it will do i uninstalled it and installed uplink.
uplink is a littlebit more difficult to get working but works great.
be carefull if you try to install, only install the main package.
the installation program wants to place a lot of not needed programs on your machine.

once setup the uplink has a + by being able to work as service (do not have to log in to the machine after startup)

the uplink doesn't provide the web interface like sisky and can not give you nice caller lists with costs (perhaps skype it self will be able to report that)
but the sound quality is great,delays are bearable and the overall is fine to use as corperate international caller carrier.

I hope these posts will help other people that consider to install the same.
if you have any questions, please post below.

[arne]

I follow your experiments, because I have done a number of them myself for a little time ago.

I did not find any good enough skype gateways (8-9 months ago), but I also experimented with diferent ways of comming trough the blocked internet connections.

One option that I tested out and actually got working was Asterisk and IP telephony trough a OpenVPN encrypted tunnel. It worked but sound quality was not good enough. (If I do not remeber it wrong, I think I had iptelephony trough a TCP 80 and a TCP 443 tunnel.)

What ended up at the two solutions that worked good enough and that I am still using is these:

SIP is a difficult protocol that is easy to block. IAX2, the native Asterisk protocol is a protocol that is possible to get trough, some way, allmost everywhere. I don't remeber what is FreePBX, is it Asterisk based ? (I think so, I'm only running a Asterisk rpm on my SME server.)

Anyhow what I'm doing is to use Asterisk via IAX2 on UDP port 53. This is the standard port/protocol for dns inquires during ordinary web browsing, so this solution works allmost everywhere. (But it depends a bit how your ISP has blocked for your ip telephony, and what you will try to obtain.)

An quite good IAX2 client that can connect to IAX2 standard port or UDP 53 is Zoiper. http://www.zoiper.com/

On most hotspot with some restrictions Zoiper/UDP53/Aterisk works bether and quicker and more easy than the Skype client.

For the mobile telephone part of my project I first bought a SIP/GSM gateway. It worked but it had rather poor technical quality, so I just experimented with some alternatives. The solution I use now, is to program all my international contacts as extensions on my Asterisk server. When I call an incomming number, I can just push the extension number, and I will be transfered via a preprogammed route to different places in the world. This makes it possible to make international calls to approx same price as a local call. (And with the SIP/GSM gateway it could be allmost for free, but with some technical bugs, from time to time.)

There is also one other option that I used/tested for a while. This is the free PBXES Asterisk server. I think it is possible to make it work together with your own local server. (And PBXES is not prevented by a firewall) https://www2.pbxes.com/index_e.php

I used both the SIP/GSM gateway and PBXES for a while, but in the end it ended up with the simple solutions I am using now. (Zero bugs or problems during more than 6 months.)

For me the Skype/Sip gatway alternative was also a solution that I did not use, but I find it quite interesting, if it should come up a solution, where thil will work "in a good enough way".

Please post your "findings" I would like to do some tests as well.

Another option - Gizmo is an alternative to skype, and I believe Gismo has a some kind of "buildt in SIP gateway": http://gizmo5.com/pc/asterisk/

It's a rather long time since I tested/used Gizmo so I'm not updated on that.

One other option I just found (never tested): http://www.chanskype.com/

[Franco]

My voip provider allows me to dial skype users for free, I just can't get the calls from the skype network (unless of course they use skypeout and dial my number).
It's a bit awkward in the way I need to dial 'user#skype' so I set up some quick dial options on my IP Phone, which is registered with sail and I have a trunk setup.
This is what they're using http://www.skip2pbx.com, not free and requires a dedicated server.

[tropicalview]

Hi arne,

many thanks for your great explanation.
my idea was to help others by this post but it seems it does help me more than i could help others.

Anyway i have to go trough your post later to understand it good (english is my second language)

In your explanation i'm interested in the gizmo part.
a question that raised, did you install gizmo on the sme server or on a windows box?

Kind regards

Hendrik

[arne]

Stuntshell ->

Thanks for info. In most cases it is possible to come trough with iptelephony the one way or the other, I think. IAX2 that can use only one port and any UDP port is then a good option.

tropicalview ->

Iptelephony has been one of my interesting hobbies the last two years. The only problem now is that everything works, and it is difficult to find a situation that can not be dealt with. At that point you learn nothing anymore. So it's interesting to see others as well does some other experiences.

When it comes to Gizmo, this is basically, as far as I remember a Windows client. BUt it can also be connected to the SME/Asterisk server via a SIP connection (Between the SME/Asterisk server and the Gizmo server. I tested a lot of alternatives, so I do not remeber it all, and my experiences might not me updated. I beleive the reason that I left the Gizmo alternative and moved over to something else were prices. The cheap and very good alternative that can be connected to SME/Asterisk (or used from a Windows client) is Voipbuster: http://www.voipbuster.com/en/index.html

About english .. it is my second language too, for sure, but I just type away, and hope someone can read it

[tropicalview]

Hi Arne,

unless the skype option is working now very fine (exept missing DTMF & callerid)
i'm also very interested in the Gizmo solution.

I took a look into that and it seems gizmo does support direct SIP connections, but then i'm back to my original problem (ports blocked)

could you please give me some tips how you did the installation?  is it a direct trunk to sip, or do you connect to the windows application (like uplink-> skype)

kind regards,

[arne]

I think there is two alternative ways to connect to Gizmo or Voibuster:

1. You can use a Windows client. (Softphone.)

2. You can make a SIP trunk at your (SME Linux) Asterisk server.

An other option that also exist, and that is more easy to get working trough firewalls etc, is to find a ventor that also accept connetion via the IAX2 protocol. (I used that before, but changed to SIP because I could do it cheaper with SIP, if it should be a problem to obtain logon/connection, I couls change back to IAX2.)

(I now use both for Voipbuster.)

When it comes to Asterisk servers there is two completely different way of setting up these servers.

Alternative 1: Some server with automated configuration, like the Selintra package for the SME Server.

Alternative 2: Configuration via manual editing of configuration files. (Takes a long time to learn, but the most funny alternative if you got the time. It is a some kind of scripting language that makes you able to design any solution, you can think about. Info in this book: http://cachefly.oreilly.com/books/9780596510480.pdf )

Note: The info in this book will mainly fit together with Asterisk servers that has a manual configuration setup. Astlinux is a very good one. I think it can also be downloaded with a Linux emulator, so it can be tested on a Windows workstation: http://www.astlinux.org/ It is very small and can also run from boot medias like a USB stick.

(I have two Asterisk servers. One is running as a part of the SME server. The other one is a Astlinux installation running on a rebuilt HP thin client.)

[arne]

This vendor offer both a sip and a iax2 connection. I used them before. I live in Europe, but it worked without a problem to use an american iptelephony vendor. http://www.vitelity.net/?p=main