Incredible PBX: Skype Gateway to Asterisk

[wardmundy]

The Incredible PBX: Adding a Free Skype Gateway to Asterisk

 

[phinphan]

Ward - looks great. I have been trying to get skype working for some time. Have you had any luck getting it running on proxmox?

 

[wardmundy]

Darn near impossible. Skype requires X-Windows.

 

[blanchae]

If I understand correctly, you need X-windows just to setup the Skype gateway then you don't need it again?

 

[phinphan]

I have done a little more searching and have come up with this page which indicates that it may be possible to add the sound and x-windows inside the openvz container:

http://www.myatus.co.uk/2009/08/24/x-server-with-sound-inside-an-openvz-proxmox-container/

I have not tried anything yet.

 

[wardmundy]

If I understand correctly, you need X-windows just to setup the Skype gateway then you don't need it again?

I think it has to be running all the time. Skype won't load without it. We just run Skype minimized in a hidden window, but it's there running under X-Windows all the time.

 

[phinphan]

So far I have been able to install everything on proxmox vm. Skype is showing that it is working and xwindows works just fine. I did install the dummy sound drivers from:

http://www.myatus.co.uk/2009/08/24/x...mox-container/

and made them start automatically. I will see how it goes.

 

[wardmundy]

Dummy sound card might solve it for the Dell servers as well. As I recall, it was the sound card that was a problem on Dell machines.

 

[phinphan]

My proxmox server is a Dell 1435 1u. It installs just fine but I have not had a chance to install the rest of the system and test it.

 

[Speedy2k]

Can someone confirm that that i can install that on a server without a screen at all ?? I would like to install it on a proxmox server, but i have got some problem already with vncserver on PBX, can someone point me or does the script already do that all ?

 

[wardmundy]

If you can't see a screen, you'd never get Skype set up properly. Sorry.

 

[jroper]

Hi

You should be able to install x-server, and remotely connect to it, if you are a Linux user, yo may already be familiar with the principle that the User Interface does not have to be on the same machine as the one it is controlling.

If you are a windows user, then you could look at VNC, which is remote control software, or http://x.cygwin.com/ to run the Linux gui in windows.

Joe

 

[Speedy2k]

I already have used VNC, but it seems to have problem starting on a server that never have a screen, if i setup VNC server with the script provided by PIAF team, and i install it on a server with a screen, it seems to be ok, but as soon as i try to install it remotly, i only got a blank screen when i connect to it ? is there any special way of doing it ?

 

[vcallaway]

Try setting up everything (including the connector) on another machine. Copy the .Skype directory into /root on your server.

You have to run the connector so that it saves the access permission in the skype settings.

Make sure you erase the .Skype directory on your test machine. If you log in to Skype from a different machine it will kick your server off.

I've not tried it, but it should work.

 

[phinphan]

Speedy2k - I got the xwindows running just fine in a proxmox vm by using the proxmox vnc terminal screen. Give that a try.

 

[MisterQ]

I have installed everything, but with a Microphone/speakers plugged into my Server, I can hear the test message dialogue, but what I am saying isn''t getting recorded. I have tried two different Microphones. Any ideas where to look/debug, and/or more importantly - do I need that bit, or is all of the sound manipulation internal, anyways.

 

[jrglass]

Can not call out using Skype addon

I can receive calls from Skype on my PBXIAF phone but when I call out the call call does not go out. I dial *echo123 and I get "all circuts are busy". Here is the log.
Thanks in advance,

Jeff

Is there a way to have the incoming skype call go to an call group rather than all phones?

Connected to Asterisk 1.4.21.2 currently running on pbx (pid = 3247)
Verbosity is at least 3
-- Executing [*3246123@from-internal:1] Macro("SIP/701-b7703540", "user-callerid|SKIPTTL|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/701-b7703540", "AMPUSER=701") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/701-b7703540", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/701-b7703540", "1|Set|REALCALLERIDNUM=701") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/701-b7703540", "AMPUSER=701") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/701-b7703540", "AMPUSERCIDNAME=701") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/701-b7703540", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/701-b7703540", "AMPUSERCID=701") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/701-b7703540", "CALLERID(all)="701" <701>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/701-b7703540", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/701-b7703540", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/701-b7703540", "Using CallerID "701" <701>") in new stack
-- Executing [*3246123@from-internal:2] Set("SIP/701-b7703540", "_NODEST=") in new stack
-- Executing [*3246123@from-internal:3] Macro("SIP/701-b7703540", "record-enable|701|OUT|") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/701-b7703540", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/701-b7703540", "recordingcheck|20100510-112737|1273505257.32") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20100510-112737|1273505257.32: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/701-b7703540", "") in new stack
-- Executing [*3246123@from-internal:4] Macro("SIP/701-b7703540", "dialout-trunk|7|3246123||") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/701-b7703540", "DIAL_TRUNK=7") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/701-b7703540", "0?sub-pincheck|s|1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/701-b7703540", "0?disabletrunk|1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/701-b7703540", "DIAL_NUMBER=3246123") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/701-b7703540", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/701-b7703540", "OUTBOUND_GROUP=OUT_7") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/701-b7703540", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/701-b7703540", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/701-b7703540", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/701-b7703540", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/701-b7703540", "outbound-callerid|7") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/701-b7703540", "0|SetCallerPres|") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/701-b7703540", "0|Set|REALCALLERIDNUM=701") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/701-b7703540", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/701-b7703540", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/701-b7703540", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/701-b7703540", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/701-b7703540", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/701-b7703540", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/701-b7703540", "0|Set|CALLERID(all)=") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/701-b7703540", "0|SetCallerPres|prohib_passed_screen") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/701-b7703540", "1|AGI|fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/701-b7703540", "OUTNUM=3246123") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/701-b7703540", "custom=AMP") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/701-b7703540", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/701-b7703540", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/701-b7703540", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/701-b7703540", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/701-b7703540", "1?customtrunk") in new stack
-- Goto (macro-dialout-trunk,s,21)
-- Executing [s@macro-dialout-trunk:21] Set("SIP/701-b7703540", "pre_num=AMP:SIP/") in new stack
-- Executing [s@macro-dialout-trunk:22] Set("SIP/701-b7703540", "the_num=OUTNUM") in new stack
-- Executing [s@macro-dialout-trunk:23] Set("SIP/701-b7703540", "[email protected]:5070") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/701-b7703540", "1?outnum:skipoutnum") in new stack
-- Goto (macro-dialout-trunk,s,25)
-- Executing [s@macro-dialout-trunk:25] Set("SIP/701-b7703540", "the_num=3246123") in new stack
-- Executing [s@macro-dialout-trunk:26] Dial("SIP/701-b7703540", "SIP/[email protected]:5070|300|") in new stack
-- Called [email protected]:5070
-- SIP/127.0.0.1:5070-0a1e9600 is ringing
-- SIP/127.0.0.1:5070-0a1e9600 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:27] Goto("SIP/701-b7703540", "s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/701-b7703540", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/701-b7703540", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
-- Executing [*3246123@from-internal:5] Macro("SIP/701-b7703540", "outisbusy|") in new stack
-- Executing [s@macro-outisbusy:1] GotoIf("SIP/701-b7703540", "0?emergency|1") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/701-b7703540", "0?intracompany|1") in new stack
-- Executing [s@macro-outisbusy:3] Playback("SIP/701-b7703540", "all-circuits-busy-now&pls-try-call-later| noanswer") in new stack
-- <SIP/701-b7703540> Playing 'all-circuits-busy-now' (language 'en')
-- <SIP/701-b7703540> Playing 'pls-try-call-later' (language 'en')
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/701-b7703540' in macro 'outisbusy'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/701-b7703540'

 

[wardmundy]

Not sure what you're doing to place calls, but this line from your log suggests it's not quite right...

Called [email protected]:5070

You need to dial 10-digit numbers with the way Skype is preconfigured unless you're entering a Skype account name.

 

[jrglass]

This was the the echo test call to echo123.

Thanks,

jeff

 

[wardmundy]

You actually have to dial E-C-H-O, not the corresponding numbers. You need a softphone that can dial alphanumeric characters.