Hi, The application is a IP-PBX with incoming calls through T1 and VOIP carriers. Asterisk will be used for Voicemail and IVR. I have written some AGI script to route calls based on user preference which is stored in the database. Users may connect through softphone or call in through PSTN. For example, a call comes in and depends on time/callerid/date, Asterisk will determine whether the call should be forwarded out, going to voicemail, etc. For the purpose of SIP registering, I am thinking about using Openser/Freeswitch, whichever is more appropriate. If Freeswitch is more like a switch, should I be using it to replace Openser or part of the routing that is done by Asterisk? Thanks alot for your inputs. Pete On Fri, Apr 11, 2008 at 6:56 AM, David Knell <dave@3c.co.uk> wrote:
Hi Pete,
Tell us a bit more about your application, and we'll be better able to help.
Cheers --
Dave
Hi all,
I am currently evaluating either using Freeswitch or Openser for my SIP server. I am trying to look for information that can tell me the difference between the two in terms of ease-of-maintain, ease-of-implement, and scalability. If you have such kind of information, would you please share with me?
Thank you very much for your input.
Best Regards, Pete
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