replace Asterisk with OpenSIPS in OpenBTS
Project
Hi,
First of all OpenSIPS is a sip server so it works only with SIP.
Secondly, by default opensips is SIP proxy, so it cannot do handover.
But using the Back2Back User agent module, you may be able to play with
the ongoing calls and move them between
On 3/29/11 1:34 PM, ALICOMPUTECH wrote:
I need to know the handoff and/or handover support in OpenSIPS as i am a
newbie to this wonderful open source solution.
What I've seen DECT based networks (which from a SIP point of view work
more or less the same as GSM with handsets moving between
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
Hi,
First of all OpenSIPS is a sip server so it works only with SIP.
Secondly, by default opensips is SIP proxy, so it cannot do handover.
But using the Back2Back User agent module, you may be able to play with
the ongoing calls and move them between different termination points.
I can help
Verzonden: dinsdag 29 maart 2011 13:35
Aan: users@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
...@yahoo.com, OpenSIPS users mailling list
users@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
@lists.opensips.org
Sent: Tuesday, March 29, 2011 2:25:50 PM GMT +01:00 Amsterdam / Berlin / Bern /
Rome / Stockholm / Vienna
Subject: Re: [OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS
Project
Hi,
First of all OpenSIPS is a sip server so it works only with SIP.
Secondly, by default
