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

[Jeff Pyle]

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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