Support A2Billing :

provided by Star2Billing S.L.

Support A2Billing :
It is currently Wed Apr 24, 2024 7:18 pm
Voice Broadcast System


All times are UTC




Post new topic Reply to topic  [ 12 posts ] 
Author Message
 Post subject: Skype for Asterisk and A2billing Integration
PostPosted: Sun Mar 14, 2010 11:15 am 
Offline

Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Hi,

If I buy skype for asterisk, can I use skype clients going through a2billing system so that it will ask card number and pin number?

Thanks


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Sun Mar 14, 2010 11:35 am 
Offline

Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Yes


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Sun Mar 14, 2010 11:59 am 
Offline

Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
jroper wrote:
Yes


How can I do it? Any document or resource to read?

Thanks again


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Sun Mar 14, 2010 6:05 pm 
Offline

Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
http://pbxinaflash.com/forum/showthread.php?t=6752 shows how to do it with FreePBX.

The Digium documentation is clear and easy to read.

Once working, then configure Skype as an access number for calling card operations.

Joe


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Tue Mar 16, 2010 11:52 pm 
Offline

Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thank you.

I have successfully installed Skype for Asterisk. From FreeBBX I set the destination to music on hold and it works fine. But when I use Skype as a Inbound access number, skype says "No answer",


-- Executing [maxtalk-dialin@from-pstn:1] Set("Skype/maxtalk-dialin-09b10fe8", "__FROM_DID=maxtalk-dialin") in new stack
-- Executing [maxtalk-dialin@from-pstn:2] Gosub("Skype/maxtalk-dialin-09b10fe8", "app-blacklist-check|s|1") in new stack
-- Executing [s@app-blacklist-check:1] LookupBlacklist("Skype/maxtalk-dialin-09b10fe8", "") in new stack
-- Executing [s@app-blacklist-check:2] GotoIf("Skype/maxtalk-dialin-09b10fe8", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:3] Return("Skype/maxtalk-dialin-09b10fe8", "") in new stack
-- Executing [maxtalk-dialin@from-pstn:3] ExecIf("Skype/maxtalk-dialin-09b10fe8", "0 |Set|CALLERID(name)=stest caller") in new stack
-- Executing [maxtalk-dialin@from-pstn:4] Set("Skype/maxtalk-dialin-09b10fe8", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [maxtalk-dialin@from-pstn:5] SetCallerPres("Skype/maxtalk-dialin-09b10fe8", "allowed_not_screened") in new stack
-- Executing [maxtalk-dialin@from-pstn:6] Set("Skype/maxtalk-dialin-09b10fe8", "_RGPREFIX=Skype:") in new stack
-- Executing [maxtalk-dialin@from-pstn:7] Set("Skype/maxtalk-dialin-09b10fe8", "CALLERID(name)=Skype:Test account") in new stack
-- Executing [maxtalk-dialin@from-pstn:8] Goto("Skype/maxtalk-dialin-09b10fe8", "custom-a2billing|maxtalk-dialin|1 ") in new stack
-- Goto (custom-a2billing,maxtalk-dialin,1)
rm-1005-15*CLI>

This is my extensions_a2billing.conf

exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup


[custom-a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|2)
exten => _X.,n,Hangup

I am using [agi-conf2] entry.

According to log, call is not forwarding to [agi-conf2]? Can I get more elaborated log?

Please help :(


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Wed Mar 17, 2010 12:27 am 
Offline

Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

This - Goto (custom-a2billing,maxtalk-dialin,1) does not match an extension in custom-a2billing.

You have _X. e.g. match any number between 0 and 9 followed by anything.

maxtalk-dialin does not start with a number between 0 and 9

Joe


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Wed Mar 17, 2010 12:56 am 
Offline

Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thank you for the prompt reply. Now its working well. :)

By the way it ask for account number and not the pin number. Can edit a2billing.conf so that it asks for pin number?

Thanks


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Wed Mar 17, 2010 7:26 am 
Offline

Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
The account number is number A2Billing uses to authenticate customers.

Joe


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Wed Mar 17, 2010 11:30 am 
Offline

Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Thanks.

One more clarification please. Which file should I edit to force to use g729 for sip trunks of a2billing?


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Thu Mar 18, 2010 4:49 pm 
Offline

Joined: Sun Jan 17, 2010 9:22 pm
Posts: 64
Location: Canada
You can force a trunk to use g729 by the settings in the trunk
ie.
disallow=all
allow=g729

also can be set globially in sip.conf

bare in mind that the g729 codec is usually not installed by default on most systems and will have to be added as it requires a license in most uses.


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Fri Mar 26, 2010 6:51 am 
Offline

Joined: Wed Dec 30, 2009 10:10 am
Posts: 17
Hi,

I have successfully set Skype for Asterisk as a DID and now it is working fine. In CDR I found that skype ID of the caller is coming as the source of the call. Can we use this ID to authenticate the call so that it system will not prompt for PIN number.

Thanks :)


Top
 Profile  
 
 Post subject: Re: Skype for Asterisk and A2billing Integration
PostPosted: Sun Sep 04, 2011 8:18 pm 
Offline

Joined: Wed Sep 15, 2010 9:09 am
Posts: 96
sky1975 wrote:
Hi,

I have successfully set Skype for Asterisk as a DID and now it is working fine. In CDR I found that skype ID of the caller is coming as the source of the call. Can we use this ID to authenticate the call so that it system will not prompt for PIN number.

Thanks :)


Really ? How to add caller ID = Skype ID (character - not number ), so that the system will not prompt PIN number (pinless) ?

Because in my extension, I use the code
Code:
exten => XXXXX,1,Answer
exten => XXXXX,2,Noop(${CALLERID(name)} , ${CALLERID(num)})
exten => XXXXX,3,Set(CALLERID(num) = ${CALLERID(name))
exten => XXXXX,4,Wait(2)
exten => XXXXX,5,DeadAGI(a2billing.php|3)
exten => XXXXX,6,Wait(2)
exten => XXXXX,7,Hangup


I make a call via my Skypename = SKYPE_NAME to DID skype. So but the system always requires authentication via PIN number.
Result
Code:
  -- Executing [XXXXX@from-sip-external:1] NoOp("SIP/sip.skype.com", "Received incoming SIP connection from unknown peer to XXXXX") in new stack
    -- Executing [XXXXX@from-sip-external:2] Set("SIP/sip.skype.com", "DID=XXXXX") in new stack
    -- Executing [XXXXX@from-sip-external:3] Goto("SIP/sip.skype.com", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/sip.skype.com", "1?from-trunk|XXXXX|1") in new stack
    -- Goto (from-trunk,XXXXX,1)
    -- Executing [XXXXX@from-trunk:1] Set("SIP/sip.skype.com", "__FROM_DID=XXXXX") in new stack
    -- Executing [XXXXX@from-trunk:2] ExecIf("SIP/sip.skype.com", "0 |Set|CALLERID(name)=Anonymous") in new stack
    -- Executing [XXXXX@from-trunk:3] Set("SIP/sip.skype.com", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [XXXXX@from-trunk:4] SetCallerPres("SIP/sip.skype.com", "allowed_not_screened") in new stack
    -- Executing [XXXXX@from-trunk:5] Goto("SIP/sip.skype.com", "a2billing|XXXXX|1") in new stack
    -- Goto (a2billing,XXXXX,1)
    -- Executing [XXXXX@a2billing:1] Answer("SIP/sip.skype.com", "") in new stack
    -- Executing [XXXXX@a2billing:2] NoOp("SIP/sip.skype.com", "SKYPE_NAME | Anonymous") in new stack
    -- Executing [XXXXX@a2billing:3] Set("SIP/sip.skype.com", "CALLERID(num) = SKYPE_NAME") in new stack
    -- Executing [XXXXX@a2billing-did:4] Wait("SIP/sip.skype.com", "2") in new stack
    -- Executing [XXXXX@a2billing-did:5] DeadAGI("SIP/sip.skype.com", "a2billing.php|3") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
    -- <SIP/sip.skype.com> Playing 'prepaid-enter-pin-number' (language 'en')



How to make a call pinless from Skype to a2billing ?
Thank in advance for all helps!
B.R
Ryan


Top
 Profile  
 
Display posts from previous:  Sort by  
Post new topic Reply to topic  [ 12 posts ] 
Hosted Voice Broadcast


All times are UTC


Who is online

Users browsing this forum: No registered users and 20 guests


You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot post attachments in this forum

Search for:
Jump to:  
Powered by phpBB © 2000, 2002, 2005, 2007 phpBB Group