[OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

[ALICOMPUTECH]

Hello
     bundle of thanks for the reply,
i am sorry for not explaining the problem in a proper way, actually OpenBTS does not support handoff of calls and i want to control it via OpenSIPS
OpenBTS does not offer handoff between base stations during a call. Handoff between calls can be done using SIP registrations to a central Asterisk.
So i want to replace the asterisk for scalability and service isolation

""Probably you're looking for: http://www.opensips.org/Resources/DocsTutLoadbalancing""
yes i need to implement loadbalancer  for Asterisk Cluster but i need to explore hanoffs of calls

""BTW, do you have OpenBTS running in a production environment""
and finally its a sort of research project and i need to implement it under the control of policies

thanks in advance

Best Regards

Bye 

----- Original Message -----
From: "Erik Dekkers" <erik.dekkers at wvds.nl>
To: "ALICOMPUTECH" <alicomputech at yahoo.com>, "OpenSIPS users mailling list" <users at lists.opensips.org>
Sent: Tuesday, March 29, 2011 2:45:28 PM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna
Subject: RE: [OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

Probably you're looking for: http://www.opensips.org/Resources/DocsTutLoadbalancing
BTW, do you have OpenBTS running in a production environment?

Regards,

Erik

-----Oorspronkelijk bericht-----
Van: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] Namens ALICOMPUTECH
Verzonden: dinsdag 29 maart 2011 13:35
Aan: users at lists.opensips.org
Onderwerp: [OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

Hello
      Everyone
               I want to replace the Asterisk (being used as a SIP Server for registration, authentication and call routing) with OpenSIPS in OpenBTS project, as i am planning to have an Asterisk cluster for dedicated services and OpenSIPS will be forwarding the SIP calls to the cluster.

OpenBTS implements GSM Um air interface and emulate the Mobile handsets as the SIP endpoint and these handsets can be used as SIP extensions in a SIP-capable server.

I need to know the handoff and/or handover support in OpenSIPS as i am a newbie to this wonderful open source solution.

If there is any pointer and/or previously handoff/handover work done please share, it will then ease my work

thanks in advance

Best Regards

Bye





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