?>Comments on: Asterisk and FreeSWITCH
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/
Open Source VoIP by Aaron RosenthalFri, 03 Feb 2012 12:54:19 +0000http://wordpress.org/?v=2.9hourly1By: QueemiChumout
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-840
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By: Bart
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-755
BartThu, 14 Jan 2010 13:30:44 +0000http://www.os-voip.com/?p=101#comment-755I agree with gitguy, don't use asterisk if your deployment is sip based. FS does much better on SIP transparency and RTP handling. I had a lot of issues with asterisk which are now resolved now on FS. Congrats to the FS development team.I agree with gitguy, don’t use asterisk if your deployment is sip based. FS does much better on SIP transparency and RTP handling. I had a lot of issues with asterisk which are now resolved now on FS. Congrats to the FS development team.
]]>By: Mark J Crane
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-754
Mark J CraneThu, 19 Nov 2009 07:13:03 +0000http://www.os-voip.com/?p=101#comment-754Like many others I started with Asterisk then learned FreeSWITCH. My favorite things about FreeSWITCH are that it can run natively on multiple platforms, its scallable, modular and very customizable. Its like a voip tool box with many pieces that can be put together to build whatever voip application you want.
I like FreeSWITCH so much that I built an open source graphical interface for FreeSWITCH called FusionPBX.Like many others I started with Asterisk then learned FreeSWITCH. My favorite things about FreeSWITCH are that it can run natively on multiple platforms, its scallable, modular and very customizable. Its like a voip tool box with many pieces that can be put together to build whatever voip application you want.
I like FreeSWITCH so much that I built an open source graphical interface for FreeSWITCH called FusionPBX.
]]>By: TimCox
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-723
TimCoxSun, 04 Oct 2009 12:48:54 +0000http://www.os-voip.com/?p=101#comment-723Also, FreeSwitch is the basis of CudaTel by Barracuda, I have Trixbox now w/Polycoms and am looking forward to the switchover/Integration at my office (Asterisk deadlocks SUCK).Also, FreeSwitch is the basis of CudaTel by Barracuda, I have Trixbox now w/Polycoms and am looking forward to the switchover/Integration at my office (Asterisk deadlocks SUCK).
]]>By: Open Source VoIP in the carrier space : A look at Bandwidth.com | OS-VoIP
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-507
Open Source VoIP in the carrier space : A look at Bandwidth.com | OS-VoIPFri, 31 Oct 2008 21:58:55 +0000http://www.os-voip.com/?p=101#comment-507[...] have nearly the same breadth of capabilities and support found in Asterisk. Take a gander at a comparison I wrote about the two. FreeSWITCH is what sits behind Bandwidth.com’s new PhoneBooth product, [...][...] have nearly the same breadth of capabilities and support found in Asterisk. Take a gander at a comparison I wrote about the two. FreeSWITCH is what sits behind Bandwidth.com’s new PhoneBooth product, [...]
]]>By: elibiordefibia
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-419
elibiordefibiaSun, 28 Sep 2008 11:32:54 +0000http://www.os-voip.com/?p=101#comment-419favorited this one, dudefavorited this one, dude
]]>By: coloquic
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-400
coloquicMon, 22 Sep 2008 18:32:18 +0000http://www.os-voip.com/?p=101#comment-400favorited this one, brotherfavorited this one, brother
]]>By: H323 Ip Phone
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-382
H323 Ip PhoneSat, 20 Sep 2008 09:15:44 +0000http://www.os-voip.com/?p=101#comment-382<strong>H323 Ip Phone...</strong>
Very nice, I like your post....H323 Ip Phone…
Very nice, I like your post….
]]>By: Anthony Minessale
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-52
Anthony MinessaleMon, 11 Aug 2008 20:09:58 +0000http://www.os-voip.com/?p=101#comment-52Being the Author of both FreeSWITCH and a long list of Asterisk 1.2 features I can offer up my opinion on the matter for what it's worth.
I was asked so many times why I bothered with FreeSWITCH that I wrote the following article to explain it:
http://www.freeswitch.org/node/117
Being an Asterisk developer myself who still hosts a bunch of my old asterisk apps, ( http://www.freeswitch.org/node/50 )
my only concern with all this is the perception of some sort of Open Source "turf war" about what software is "better" etc. When I decide what open source software I use I judge it purely on if it's the *right tool for the job* (tm). I decided Asterisk was not the right tool only after 3 years of using it an learning it inside and out hence why I started FreeSWITCH. Many may find Asterisk more than adequate or maybe they use Asterisk, FreeSWITCH and OpenSER all together in an enterprise system. Why should we care? We are all on the same team..The Open Source VoIP team.Being the Author of both FreeSWITCH and a long list of Asterisk 1.2 features I can offer up my opinion on the matter for what it’s worth.
I was asked so many times why I bothered with FreeSWITCH that I wrote the following article to explain it: http://www.freeswitch.org/node/117
Being an Asterisk developer myself who still hosts a bunch of my old asterisk apps, ( http://www.freeswitch.org/node/50 )
my only concern with all this is the perception of some sort of Open Source “turf war” about what software is “better” etc. When I decide what open source software I use I judge it purely on if it’s the *right tool for the job* ™. I decided Asterisk was not the right tool only after 3 years of using it an learning it inside and out hence why I started FreeSWITCH. Many may find Asterisk more than adequate or maybe they use Asterisk, FreeSWITCH and OpenSER all together in an enterprise system. Why should we care? We are all on the same team..The Open Source VoIP team.
]]>By: asdx
http://www.os-voip.com/2008/08/asterisk-and-freeswitch/comment-page-1/#comment-50
asdxSun, 10 Aug 2008 22:16:42 +0000http://www.os-voip.com/?p=101#comment-50Some things I would like to have in FS:
1- an easy way to dial out from conferences, from the phone-side.
2- the ability to set per-bridge/leg timeout, along with global timeout.
3- complete IAX/H.323 stack that supports registration, trunking and everything.
4- "sip/iax2 show peer " to see user settings from the cli.Some things I would like to have in FS:
1- an easy way to dial out from conferences, from the phone-side.
2- the ability to set per-bridge/leg timeout, along with global timeout.
3- complete IAX/H.323 stack that supports registration, trunking and everything.
4- “sip/iax2 show peer ” to see user settings from the cli.