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

Flavio Goncalves flavio at voffice.com.br
Mon Oct 25 23:44:43 CEST 2010


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 successfully implemented most of them including Call Pickup, Call
Forward, Attended and Unattended Transfers. Asterisk is an excellent gateway
when you want to implement all these features. A good phone is also
important. Don't try this with X-Lite.

Best Regards,

Flavio E. Goncalves
CEO - V.Office
Fone: +554830258590/+554884085000
OpenSIPS Bootcamp (New Jersey, NY  Nov. 15-19)



2010/10/24 Mike O'Connor <mike at oeg.com.au>

> 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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