Hello,
a bit late because of time constraints, and not yet fully complete -- I
tried to punctuate the idea behind the talks and presentations at the
summit, along with some personal conclusions. It should play along with
the slides for those which were not able to attend. We have some short
videos, but need to rework the audio.
http://openser.blogspot.com/
The posting order is the same as events happened. The afternoon session
of the second day will be reviewed soon.
Best regards,
Daniel
Can someone please give me an example configuration of how to set up ser to
pass calls to pstn gateways? I want to set up SER to be able to pass calls
based on certain dial plans to certain gateways I have set up. Please send
me links or sample configuration files. It would be greatly appreciated.
Thanks
Hi Samuel,
thank you for your effort and great beginning. I see too this as
a laborous effort but in case you decide to go on, you are most
welcome to use the Ottendorf announcement as input. It is not
complete, but the major changes are covered (hopefuly :-))
-jiri
At 11:34 28/11/2006, samuel wrote:
>Hi folks,
>I've tried to stay out of this thread but I think it has gone far
>beyond technical arguments and I don't think it will help neither SER
>or openSER projects. I would just ask to stop this thread before it
>get worst or stack to technical discussions (which I'm learning a lot)
>and leave personal appreciations aside.
>
>I still see advantages on each project and depending on the
>requirements, one is more suited than the other. I had the hope that
>both would merge with the best of them but I see this almost
>impossible right now...
>
>I'll try to summarize differences and I would apreciate *technical*
>corrections to it (it's almost impossible to have an update
>list....fortunately, i've tried to enumerate diferences in the dev
>versions which will soon become stable):
>*openSER:
>o1)detailled documentation, including howtos
>o2)fast reaction in the mailing list from the core developers
>o3)incorporate user comments into new features
>o4)decentralised developers
>o5)improved usrloc (loading on mem, cacheless)
>o6)gaining user acceptance->biggest testing comunity
>
>*SER
>s1)improved TCP code
>s2)improved timers
>s3)easier integration with SEMS/SERWEB
>
>I have tried to be as opbjective as possible (but everyone knows it's
>almost impossible) and it would be great to create a common place to
>share experiences with both projects as someone has proposed but I
>guess everyone is really really busy right now...
>
>Samuel.
>_______________________________________________
>Serusers mailing list
>Serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
hi ,
i have integrated SER with NAT Helper +Media proxy + Mysql +PSTN modules. but i am unable to include B2BUA and Radius modules. can some one please guide me in this issue. moreover when i am calling a PSTN number from my SIP UA, after the call is connected, when the call is disconnected on the PSTN end, the BYE packet is not reaching my SIP UA end.similarly, when the call is cancelled at the PSTN end, it shows still RINGING in the SIP UA end. but when i call from SIP UA-1 to SIP UA-2 (online call) , when the call is terminated on either side, call gets disconnected on both sides. so, can some one throw some light on this ? moreover how r B2B UA and Radius related , i mean how do they communicate with each other?
Thank you,
Rupesh.
We use this billing http://www.netup.biz/
It works good with OpenSER Auth, Start, Stop without writing additional
scripts.
Lyubimkov Dmitriy
---------------------------------------------------------------------
Date: Fri, 24 Nov 2006 12:06:48 +0200
From: Daniel-Constantin Mierla <daniel(a)voice-system.ro>
Subject: Re: [Users] Billing system with good integration with OpenSER
/ Documentations
To: Nick De Cristofaro <nickdc(a)link2exchange.com>
Cc: users(a)openser.org
Message-ID: <4566C438.5020102(a)voice-system.ro>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
OpenSER is very good at generating accounting events (START, STOP).
Starting from there you can get CDRs in any format suitable for some
free billing systems out there. The CDRs can be generated via simple
stored procedure or perl/php/... script.
I haven't used any open source billing systems, but there are quite a
lot.
http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems
Cheers,
Daniel
On 11/09/06 13:28, Nick De Cristofaro wrote:
> Hey everyone,
>
> Good sessions yesterday at the OpenSER summit. I found the
> presentations very insightful and I learned a lot of new things. I am
> hoping to get some feedback as a good solution for a billing system
> that we will need to implement over here in our company that can be
> easily interacted through OpenSER. As I am pretty new to this I am
> wondering what component pieces should be setup to do this right.
> OpenSER as our main SIP router with database access and creating CDRs
> or do we need a B2BUA with this? Can anyone recommend some good
> starting point to look into or some publishings that would be a good
> read to get more insight on building this right.
>
> Some of the presentations had some good insights on scaling,
> redundencies and so forth and we would like to do this right from the
> beginning and look at our cost platforms.
>
> Anyone ever used the Hiper biling system? I saw their exhibit
> yesterday and it was pretty nice.
>
> Thanks
> Nick
>
------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
------------------------------
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End of Users Digest, Vol 18, Issue 63
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Resend to the list
> Sip - Specific Event Notification
> (http://www.ietf.org/rfc/rfc3265.txt) defines the NOTIFY Method.
> Within the above RFC the Content-Length header is designated with a
> "t" under the NOT (Notify) column (Section 7.1).
>
> The "t" designation is defined in the SIP
> (http://www.ietf.org/rfc/rfc3261.txt) definition as follows:
>
> t: The header field SHOULD be sent, but clients/servers need to be
> prepared to receive messages without that header field.
>
> If a stream-based protocol (such as TCP) is used as a
> transport, then the header field MUST be sent.
>
>
> Hope this helps
>
> Regards,
> Norm
>
> Chris Robson wrote:
>>
>> Is the field "Content-length" a required field in a NOTIFY message?
>> I see that it is part of almost every message from OpenSER except the
>> NOTIFY message. This is causing some issues with Gaim SIMPLe
>> interface which processes the header based on the header string
>> "Content-Length". Note Gaim does not care about the value associated
>> with the header string just the fact that the string "Content-Length"
>> exists. Therefore, since OpenSER isnt sending the string, Gaim is
>> core dumping. FYI on the core dump. Yes, Gaim still has a bug on
>> how it is processing what it sees as unknown SIP messages which
>> actually is causing the core dump, the real problem is the fact that
>> Gaim is processing the header, expecting to see the field
>> "Content-length". So before I report this to Gaim as a bug, I wanted
>> to get insight into whether the field is a required field.
>>
>> Thanks......
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)openser.org
>> http://openser.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
Hi all,
How to make the sip phone offline(no dial tone) when I stop the ser by command ser.init stop?
Because now my sip phone will appear in the Online Table and dial tone on it although i stop my ser, until it reach the expire time.
Thanks.
Regards,
jorain
