SIP in FreeSWITCH vs Asterisk

Joined: Wed 06 of Jul, 2005

Re: SIP in FreeSWITCH vs Asterisk

Posted:Wed 20 of Oct, 2010 (20:17 UTC)
Joined: Fri 20 of Apr, 2007

Re: SIP in FreeSWITCH vs Asterisk

Posted:Mon 15 of Sep, 2008 (13:43 UTC)
This is great... asterisk guys answering on the FreeSWITCH forums ;-) Thanks John!

That said... It's been my very personal experience, that everything i've wanted to do with asterisk i could do with FreeSWITCH. And there have been many times i've wanted to do things with asterisk and couldn't... but FreeSWITCH has native support for them. TLS is a big one for me... and, at this point in time, i could wait until 1.6 is considered "stable".. but it was months ago that i was making this decision. To the best of my knowledge, TLS (and even TCP) support in asterisk is an afterthought that wasn't addressed until recently.

Anyway, back to the topic at hand... FreeSWITCH makes an excellent PBX without the need of a proxy (and i'm sure, i don't "suspect") and will let you have complete control over the media and what happens with it.

A couple of articles for your viewing pleasure :-)

(this last one isn't really pertinent to your current investigation, just something i like to share.... FOSS working together for the "greater good")
Joined: Wed 07 of Jan, 2004

Re: SIP in FreeSWITCH vs Asterisk

Posted:Sun 14 of Sep, 2008 (23:27 UTC)
Asterisk will do this fine, and I suspect you probably don't need a SIP proxy for what you want. Short summary here: http://www.voip-info.org/boards/index.php?t=16415

Joined: Fri 14 of Sep, 2007

SIP in FreeSWITCH vs Asterisk

Posted:Sun 14 of Sep, 2008 (00:58 UTC)
I posted the below in the Asterisk forum on here, and have no experience with FreeSWITCH (although am currently installing it in a Centos VMware machine to experiment) and wasn't sure if perhaps FreeSWICH may be better in order to achieve what I'm trying to do - think of it as a small (but expanding) hosted VoIP service...

I've got an asterisk box shared by 4 offices, each has about 5 or 6 extensions in it although this may expand in the near future.

I'm looking to make sure all internal calls (within the same building) go direct from phone to phone rather than Asterisk remaining in the media path, in order to save as much bandwidth as possible, I believe Asterisk has issues with with this and that the general consensus is to use a SIP proxy in front of it and keep Asterisk for providing services such as IVR and Voicemail, but I'd love to hear from anyone with experience in doing something similar ?