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Skype on Asterisk 1.4.31 --- Help needed

Offline Teviot

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Skype on Asterisk 1.4.31 --- Help needed
« on: June 21, 2010, 06:19:10 AM »
I'm am trying to install "Skype for Asterisk" on my SAIL/SARK pbx.I am trying to work out what to do from README file found at http://downloads.digium.com/pub/telephony/skypeforasterisk/README.

So far I have found an that the step involving the register part isn't working for me. Maybe I doing something wrong but I believe that I have followed the instrautions correctly.

Code: [Select]
[root@saint ~]# wget http://downloads.digium.com/pub/register/linux/register
--14:35:03--  http://downloads.digium.com/pub/register/linux/register
           => `register'
Resolving downloads.digium.com... 76.164.171.232, 2001:470:e0d4::e8
Connecting to downloads.digium.com|76.164.171.232|:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: 1,178,780 (1.1M) [text/plain]

100%[=================================================================================>] 1,178,780    155.00K/s    ETA 00:00

14:35:12 (139.55 KB/s) - `register' saved [1178780/1178780]

[root@saint ~]# chmod 500 ~/register
[root@saint ~]# register
-bash: register: command not found
[root@saint ~]#


Can somebody help me as I diving into to new territory and not really should where I'm going.

I'm using SME v7.5 with a clean install and asterisk installed from source.

Have also found http://docs.digium.com/SFA/skype_for_asterisk_admin_manual.pdf
« Last Edit: June 21, 2010, 06:38:56 AM by teviot »
Regards
M0GLJ
......................................................
I am new to SAIL SME Server v8b6 and have been using SME for many years.
I have already done some research and only ask questions if I still can't work it out.

Offline SARK devs

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Re: Skype on Asterisk 1.4.31 --- Help needed
« Reply #1 on: June 21, 2010, 07:10:22 AM »
its just a bad readme

try

Code: [Select]
./register
Kind Regards

S

Offline Franco

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Re: Skype on Asterisk 1.4.31 --- Help needed
« Reply #2 on: June 22, 2010, 05:14:20 AM »
Please let me know if you succeed on this.
I'm running asterisk 1.4.29 with this and also the sip trunk, mine have been compiled from source.

Best,