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" <[email protected]>
To: "ALICOMPUTECH" <[email protected]>, "OpenSIPS users mailling list"
<[email protected]>
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: [email protected] [mailto:[email protected]]
Namens ALICOMPUTECH
Verzonden: dinsdag 29 maart 2011 13:35
Aan: [email protected]
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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