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 Post subject:
PostPosted: Sun Sep 30, 2007 5:09 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4064
Hi

Quote:
"SER and a2billing how to use the same database(mysql)?"



Therein lies the problem to solve!

Joe


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 Post subject:
PostPosted: Fri Jan 25, 2008 1:50 pm 
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Joined: Wed Jan 02, 2008 3:33 pm
Posts: 26
This is a very interesting thread. Thanks to all for your efforts.

Any update? I was also thinking about creating a VIEW for OpenSER from A2B table, but have not tried yet. Are you sure that it doesnt work, Howard?

I will try it now and come back to you guys soon.

Keep it up! :-)


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 Post subject:
PostPosted: Sat Jan 26, 2008 5:29 pm 
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Joined: Wed Jan 02, 2008 3:33 pm
Posts: 26
Howard, you must be wrong. I have created a view for Openser from cc_card, and IT WORKS!!!

:P :P :P :P :P


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 Post subject:
PostPosted: Sun Jan 27, 2008 9:32 pm 
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Joined: Mon Oct 01, 2007 10:44 pm
Posts: 230
Location: Bovey, Devon, UK
whatis wrote:
Howard, you must be wrong. I have created a view for Openser from cc_card, and IT WORKS!!!

:P :P :P :P :P


Care to share the view :) I would have thought sip buddies would be better though as we don't want iax peers loaded as sip clients

regards


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 Post subject:
PostPosted: Sun May 11, 2008 2:19 pm 
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Joined: Tue May 23, 2006 8:30 pm
Posts: 67
HI
I also have created a view from cc_card to use asterisk as a sip server but now I need to use openser because my sip clients network is growing, please share your View for openser, this is my view for asterisk, hope it helps to someone:

-----
PARA CREAR REALTIME SIP USERS FROM CC_CARD DE A2BILLING CON DIFERENTES CONTEXTOS SEGUN PREFIJOS:

CREATE VIEW sipuserRT02 AS
SELECT id AS id,
useralias AS name,
useralias AS username,
'friend' AS type,
username AS accountcode,
useralias AS regexten,
CONCAT(firstname, ' ','<',useralias,'>') AS callerid,
'billing' AS amaflags,
userpass AS secret,
'dynamic' AS host,
'RFC2833' AS dtmfmode,
CONCAT('rt-clientes-',substring(cc_card.useralias,1,2)) AS context,
useralias AS mailbox,
'yes' AS nat,
'no' AS qualify,
'no' AS canreinvite,
'all' AS disallow,
'g729,ulaw,alaw,gsm' AS allow,
'0' AS regseconds
FROM mya2billing.cc_card WHERE tariff=2;

------

slds.-
rafael



whatis wrote:
Howard, you must be wrong. I have created a view for Openser from cc_card, and IT WORKS!!!

:P :P :P :P :P


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 Post subject:
PostPosted: Mon May 12, 2008 3:25 pm 
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Joined: Mon Apr 30, 2007 6:43 am
Posts: 1060
Location: Canada
Is there a reason why you are not using any information from the SIP friend table? Without doing so, I fail to see how you view will allow anyone to make a successful call or even to register.

Cheers


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 Post subject:
PostPosted: Mon May 12, 2008 10:11 pm 
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Joined: Tue May 23, 2006 8:30 pm
Posts: 67
Hi
It works for me, I am using this view to avoid the generation of more than 70000 sip clients that I already have as a customers in a2billing, I use this configuration just to simplify provisioning process, now I want to integrate openser as a sip registrar and proxy server for better performance.


asiby wrote:
Is there a reason why you are not using any information from the SIP friend table? Without doing so, I fail to see how you view will allow anyone to make a successful call or even to register.

Cheers


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 Post subject:
PostPosted: Tue Sep 23, 2008 6:32 pm 
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Joined: Wed May 09, 2007 6:05 pm
Posts: 3
This is an interesting subject in which I am quite interested although I am not a programmer by any means but an integrator and sys admin.

Has anyone of you explored other options such as; AstBill - http://astbill.com/whatis or better yet Open Settlement Protocol (OSP) ( http://en.wikipedia.org/wiki/Open_settlement_protocol ) - used by service providers for called number translation and Call Detail Record collection via HTTP, XML, etc. Various opensource implementations include;

OpenOSP - http://sourceforge.net/projects/openosp ; developed by Cisco
RAMS - http://sourceforge.net/projects/rams , and
OSP Toolkit - http://sourceforge.net/projects/osp-toolkit

With OSP, billing, CDR, can be done directly between Asterisk and OpenSER without a2billing in between. Asterisk already has /etc/asterisk/osp.conf inbuilt and OpenSER also has an OSP module; http://www.kamailio.org/docs/modules/1.3.x/osp.html. It may just be a matter of adding any of the above listed OSP servers into OpenSER so it can accept OSP authentication and CDR info from Asterisk for billing. This is an open standard protocol that service providers use for service/reporting/billing. Adding a UI to it should be easy.

There is also RADIUS-to-OSP - http://sourceforge.net/projects/radius-to-osp/ , for those interested in RADIUS as well.

To further confirm the reliability of OpenSER and other open source projects, it may also surprise you all which big brands out there uses OpenSER as well as other open source projects; http://voip-hype.com/cisco-uses-openser ... nksys-one/

More info here; http://www.transnexus.com/OSP%20Toolkit ... uments.htm


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 Post subject:
PostPosted: Wed Nov 19, 2008 1:23 am 
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Joined: Wed Jul 25, 2007 4:42 am
Posts: 21
Hi All,

After over 1 year, I finally got some spare time to put my hand back on this project.

I am currently use A2Billing 1.3.4 + Asterisk 1.4.22 + Opensips 1.4.2.

Now, i can confirm that what i was trying to do can be done and now i have done what i want to achieve.

Assuming people who have read the previous posts. the problem was in the mysql. Finally, i know how i can fix the "VIEW" between opensips and a2billing.

Here is the command for VIEW for subscriber table in opensips.

Create view subscriber AS
SELECT id as id, useralias as username,
'your.domain' as domain,
uipass as password,
email as email_address,
md5(concat(useralias,':',uipass)) as ha1, md5(concat(useralias,'@yourdomain:',uipass)) as ha1b,
useralias as rpid,
useralias as uuid
FROM mya2billing.cc_card;

Then next step is to update the table version on the version table.
update version set table_version='6' where table_name=subscriber;

version number need to be bigger then the original one. This is what i got sucked before.

Now you should able to use opensips as your sip register and send call to a2billing for prepaid voip billing!


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 Post subject:
PostPosted: Wed Dec 10, 2008 11:37 pm 
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Joined: Wed Jan 02, 2008 3:33 pm
Posts: 26
Good job, howard!

I used your old VIEW and it works without any problem. I can send calls from Openser / Opensips to asterisk and then to PSTN.

Now I am trying to call a SIP buddy from PSTN but can not get it to work. Has anyone tried to send calls from Asterisk to Openser ?

PSTN => Asterisk => Openser => SIP phone

Openser + Asterisk + a2b would be perfect if we can solve the problem above. :roll:


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 Post subject:
PostPosted: Wed Dec 10, 2008 11:42 pm 
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Joined: Mon Apr 30, 2007 6:43 am
Posts: 1060
Location: Canada
I am wondering why would you do that? If the number is coming in via a DID used as access number, then use IP authentication to accept all calls coming in on the DID and dispatch them to the right place in your asterisk servers. I am not entirely sure if I am making any sense. Let's says 95% sure.

Cheers


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 Post subject:
PostPosted: Wed Dec 10, 2008 11:55 pm 
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Joined: Wed Jan 02, 2008 3:33 pm
Posts: 26
hmm, but the SIP buddies registered with Openser, not with Asterisk! Or am I missing something ? :-/


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 Post subject:
PostPosted: Mon Dec 22, 2008 11:05 am 
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Joined: Mon Dec 22, 2008 10:45 am
Posts: 25
I didn't read all...

but, I have 1openser, and 3 A2billing

all work with the same database.
it work!

registrar is done by openser, and dispatch to the A2Billing(s).
I had just a problem with the asterisk. I had to comment the secret line in the SIP.conf ...

Added after 2 minutes:

asterisk see in the frame the username!
if we comment the 'secret' in the sip.conf. asterisk accept the call, because the user is auth in the openser!


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 Post subject:
PostPosted: Wed Dec 24, 2008 9:32 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4064
Quote:
registrar is done by openser, and dispatch to the A2Billing(s).
I had just a problem with the asterisk. I had to comment the secret line in the SIP.conf ...


Hi

I may have not understood this correctly, but what is to stop someone registering a SIP connection directly to your asterisk server, as there is no secret to authenticate them?

Joe


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 Post subject:
PostPosted: Wed Dec 24, 2008 9:37 am 
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Joined: Mon Dec 22, 2008 10:45 am
Posts: 25
I have one openser and some asterisk(s)
I just need to auth in the openser not in the asterisk.


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