[OpenSIPS-Users] How much of the normal PBX Functions can be implemented using OpenSIPS, Mediaproxy and maybe SEMS

[Duane Larson]

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 <
testa at voicetechnology.com.br> wrote:

> 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,
> >
> > 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 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 <mike at oeg.com.au
> >> <mailto:mike at oeg.com.au>> wrote:
> >>
> >>     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
> be
> >>     done using OpenSIPS, Mediaproxy and maybe SEMS.
> >>
> >>     My current core system has voicemail, call forwarding and T38 fax
> >>     using
> >>     sip forwards to asterisk, but as normal with Asterisk I do get
> >>     occasional calls issues, mostly related to codec negotiation.
> >>
> >>     I want to be able to have all the normal PBX functions like Auto
> >>     attendant, Call forwarding on busy or absence, Call Park, Call
> pickup,
> >>     Call transfer, Call waiting, Conference Call, Custom Greeting, Voice
> >>     Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC
> >>
> >>     So your comments requested.
> >>
> >>     Thanks
> >>     Mike
> >>
> >>
> >>
> >>
> >>     _______________________________________________
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> >>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> >>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >>
> >> _______________________________________________
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> >
> >
> >
> > _______________________________________________
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>
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>



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Duane
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