So are you able to integrate FreeSwitch with OpenSIPS like Asterisk is integrated(Usernames and Passwords link up)?<br><br>
<div class="gmail_quote">On Mon, Oct 25, 2010 at 8:44 AM, Fernando Gregianin Testa <span dir="ltr"><<a href="mailto:testa@voicetechnology.com.br">testa@voicetechnology.com.br</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">As a FreeSWITCH user for about 1yr for conferencing systems, I can<br>assure it works very well as a virtualized PBX on KVM, Xen or OpenVZ<br>
virtualization platforms.<br><br>Fernando Gregianin Testa<br>Voice Technology Ltda<br>ddr +55 11 21752166<br>cel +55 11 88225531<br>
<div class="im"><br>On 24-10-2010 16:25, Jeff Pyle wrote:<br>> Mike,<br>><br>> We've been asking much the same questions. We have decided to take a<br>> serious look at Freeswitch for the "Asterisk-style" functions, while<br>
> leaving the core routing functions to Opensips.<br>><br>><br>> - Jeff<br>><br>><br>> On Oct 24, 2010, at 1:16 AM, Mark Sayer wrote:<br>><br>>> Those "virtual PBX" functions, like your present voicemail, cannot be<br>
>> provided by OpenSIPS. They are Asterisk-style functions.<br>>><br>>> Mark<br>>><br>>> On Sun, Oct 24, 2010 at 2:04 PM, Mike O'Connor <<a href="mailto:mike@oeg.com.au">mike@oeg.com.au</a><br>
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<div class="im">>> <mailto:<a href="mailto:mike@oeg.com.au">mike@oeg.com.au</a>>> wrote:<br>>><br>>> 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<br>>> 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<br>
>> 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>>> _______________________________________________<br>
>> Users mailing list<br></div>>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a> <mailto:<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>><br>
<div class="im">>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>>><br>>><br>>> _______________________________________________<br>
>> Users mailing list<br></div>>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a> <mailto:<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>><br>
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<div class="h5">>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>><br>><br>><br>> _______________________________________________<br>
> Users mailing list<br>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a><br>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>