<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Mike,<div><br></div><div>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.</div><div><br></div><div><br></div><div>- Jeff</div><div><br></div><div><br><div><div>On Oct 24, 2010, at 1:16 AM, Mark Sayer wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">Those "virtual PBX" functions, like your present voicemail, cannot be provided by OpenSIPS. They are Asterisk-style functions.<br><br>Mark<br><br><div class="gmail_quote">On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor <span dir="ltr"><<a href="mailto:mike@oeg.com.au">mike@oeg.com.au</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Guys<br>
<br>
I've been using OpenSIPS now for about 9 month (after upgrading from<br>
OpenSER 1.2 used that for about 2 years) for my core SIP routing and<br>
billing.<br>
<br>
I'm now getting questions from customers about Virtual PBX functionality<br>
and I would like the opinion of the group about how well this could be<br>
done using OpenSIPS, Mediaproxy and maybe SEMS.<br>
<br>
My current core system has voicemail, call forwarding and T38 fax using<br>
sip forwards to asterisk, but as normal with Asterisk I do get<br>
occasional calls issues, mostly related to codec negotiation.<br>
<br>
I want to be able to have all the normal PBX functions like Auto<br>
attendant, Call forwarding on busy or absence, Call Park, Call pickup,<br>
Call transfer, Call waiting, Conference Call, Custom Greeting, Voice<br>
Mall, Public Addressing, DND, Direct Inward Dial, Busy Lamp. ETC<br>
<br>
So your comments requested.<br>
<br>
Thanks<br>
Mike<br>
<br>
<br>
<br>
<br>
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