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 <m...@oeg.com.au<mailto:m...@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 _______________________________________________ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users@lists.opensips.org<mailto:Users@lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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