Can't dial out using "Skype for Asterisk" plugin...

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Can't dial out using "Skype for Asterisk" plugin...

Postby butlerm1977 » Tue Mar 16, 2010 12:10 pm

Hi there, i'm trying to create a very basic setup. Experimenting with Asterisk 1.6 and the skype for asterisk plugin. Still new to this, so please bare with me.

I've created the skype business control panel and created a fresh skype account (skype-username01) using the BCP. I've loaded the skype for asterisk modules and the g729 codec. Down below, i'll give you a copy of all my config files.

My problem is that I cannot make outgoing phone calls in my test environment. I get the following feedback from the CLI at verbosity 5 when trying to dial 18324276969 from X-lite.

Asterisk Ready.
== Parsing '/etc/asterisk/cli.conf': == Found
*CLI> == Using SIP RTP CoS mark 5
-- Executing [18324276969@default:1] Dial("SIP/2000-00000000", "Skype/skype-username01@18324276969") in new stack
[Mar 14 09:14:07] WARNING[2984]: chan_skype.c:611 call: Unable to ring user '18324276969'
-- Couldn't call skype-username01@18324276969
[Mar 14 09:14:07] NOTICE[2984]: core.cpp:2138 sfa_call_hangup: ending call
== Everyone is busy/congested at this time (0:0/0/0)
-- Auto fallthrough, channel 'SIP/2000-00000000' status is 'CHANUNAVAIL'

I did notice that I CAN call into the asterisk box from the PSTN. Here's a the CLI output while calling in...

*CLI> -- Executing [3000@default:1] Dial("Skype/skype-username01-09d07968", "SIP/2000,20") in new stack
== Using SIP RTP CoS mark 5
-- Called 2000
-- SIP/2000-00000001 is ringing
-- SIP/2000-00000001 answered Skype/skype-username01-09d07968
-- Packet2Packet bridging Skype/skype-username01-09d07968 and SIP/2000-00000001
== Spawn extension (default, 3000, 1) exited non-zero on 'Skype/skype-username01-09d07968'
[Mar 14 09:17:48] NOTICE[3019]: core.cpp:2138 sfa_call_hangup: ending call

So my problem is that I can dial in, but not out. I've tried in Xlite to enter the phone number in a few different ways...ten digit, +1, 1, 001, +001, etc.

Here's my config files...

[root@vm-aster asterisk]# cat /etc/asterisk/extensions.conf
[default]
exten => 1001,1,Answer()
exten => 1001,2,Playback(hello-world)
exten => 1001,3,Hangup()

exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,n,Voicemail(2000,u)

exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)

exten => 3000,1,Dial(SIP/2000,20)
exten => 3000,n,Voicemail(2000,u)

exten => _X.,1,Dial(Skype/skype-username01@${EXTEN})

[root@vm-aster asterisk]# cat /etc/asterisk/sip.conf
[general]
port=5060
bindaddr=0.0.0.0
disallow=all
allow=g729
allow=g722
allow=g711
allow=ulaw
allow=alaw

[2000]
type=friend
secret=xxxxxxxxx
host=dynamic

[root@vm-aster asterisk]# cat /etc/asterisk/chan_skype.conf
[skype-username01]
secret=xxxxxxxxxxx
context=default
exten=3000
disallow=all
allow=ulaw,alaw,g729,g711
direction=both
auth_policy=ignore
buddy_autoadd=no
buddy_presence=no
mohinterpret=default
mohsuggest=none
butlerm1977
Newsterisk
 
Posts: 2
Joined: Tue Mar 16, 2010 9:12 am

Re: Can't dial out using "Skype for Asterisk" plugin...

Postby butlerm1977 » Wed Mar 17, 2010 7:10 am

Well after reviewing other posts on the forum I fixed the problem... I added one dollar credit to my skype account. I was under the impression that skype subscriptions would work to fund the account. Evidently I was wrong. The wording on Digium's website states: "Please note: Skype for Business subscription prices do not apply." Before I purchased my license, I contacted Digium to clarify the meaning of that. I was told to talk to Skype, they could not help me with pricing clarification. A simple no would have been sufficient.

Contacting anyone live at Skype is extremely difficult. I've yet to find a US support number to ask a simple question. Instead i'm having to deal with a chat session.

So my review of Skype for Asterisk:
The good: easy to setup.

The bad: unlimited subscriptions do not work within the Asterisk arena. Looks like USA phone calls will cost $.029 per minute.

For any others out there who are thinking of trying out this plugin, have a close read of the Skype Fair Use Policy, which can be found here...
http://www.skype.com/legal/terms/fair_usage/
Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means).

...limit of 10,000 minutes per user per month

...maximum of 6 hours per day

...no more than 50 different numbers in total can be called per day


Just thought I'd pass this info along...
butlerm1977
Newsterisk
 
Posts: 2
Joined: Tue Mar 16, 2010 9:12 am


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