Hi Mike,
It is possible to implement all PBX functions using OpenSIPS. It is not
easy, it depends on your phones and gateways. They have to support several
RFCs such as RFC3515, RFC3891, RFC3892. Check RFC5359 (
http://tools.ietf.org/html/rfc5359) for more details about call flows. I
have
As a FreeSWITCH user for about 1yr for conferencing systems, I can
assure it works very well as a virtualized PBX on KVM, Xen or OpenVZ
virtualization platforms.
Fernando Gregianin Testa
Voice Technology Ltda
ddr +55 11 21752166
cel +55 11 88225531
On 24-10-2010 16:25, Jeff Pyle wrote:
Mike,
So are you able to integrate FreeSwitch with OpenSIPS like Asterisk is
integrated(Usernames and Passwords link up)?
On Mon, Oct 25, 2010 at 8:44 AM, Fernando Gregianin Testa
te...@voicetechnology.com.br wrote:
As a FreeSWITCH user for about 1yr for conferencing systems, I can
assure it works
Mike,
We've been asking much the same questions. We have decided to take a serious
look at Freeswitch for the Asterisk-style functions, while leaving the core
routing functions to Opensips.
- Jeff
On Oct 24, 2010, at 1:16 AM, Mark Sayer wrote:
Those virtual PBX functions, like your
Hi Guys
I've been using OpenSIPS now for about 9 month (after upgrading from
OpenSER 1.2 used that for about 2 years) for my core SIP routing and
billing.
I'm now getting questions from customers about Virtual PBX functionality
and I would like the opinion of the group about how well this could
Those virtual PBX functions, like your present voicemail, cannot be
provided by OpenSIPS. They are Asterisk-style functions.
Mark
On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor m...@oeg.com.au wrote:
Hi Guys
I've been using OpenSIPS now for about 9 month (after upgrading from
OpenSER 1.2
