[Freeswitch-users] Asterisk to FreeSWITCH migration guide

Nestor A Diaz nestor at tiendalinux.com
Tue Aug 9 20:48:00 MSD 2011


Hi Guys.

I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7 
release appears as a package on the debian distribution, at the 
beginning i was amazed by the fact i can build a PBX for my own business 
and i did, later i began to install this system for my customers and 
sooner i meet the problems, however being the software open source i 
always find a way to fix things using patchs from others, sometimes i 
felt how my life was at risk when the system stops working and that 
usually happens when i have to use queues and dealing with digium hardware.

Fixing those problems either by applying patches or by changing the 
hardware where the digium cards were supposed to be installed helps me, 
but that was to much stress for me and seeking for a balance that will 
let me invest more time on services, configuration and hoping to have 
better hardware options brings me to freeswitch.

I agree with freeswitch philosophy that instead of having thousands of 
modules that don't work fine i prefer just a few that works the way it 
should be, a rock solid system for a corporate pbx and a call center is 
what i want.

So here i am trying to begin the conversion, and i hope the information 
we can transcript in this list will help others that want to try another 
alternative to asterisk.

First of all i think the saner for a migration is to have the two 
systems running either on the same machine or different and use the 
stable features of each one.

So could you please freeswitch users help me with this rosetta stone 
migration guide in order to post it to voip-info.org or freeswitch wiki 
(i list only the ones i currently use ):


*Technology* 	*Asterisk* 	*Freeswitch*
PSTN Connectivity (Digium / Sangoma) 	dahdi 	freetdm
IAX2 	mod_iax 	?? none stable yet.
Use Asterisk to forward traffic via SIP.
Enable Hardware HPET for IAX2 trunk if card not available for Asterisk
Bluetooth Channel 	chan_mobile 	??
Use asterisk via SIP
Skype 	Skypeforasterisk (no longer for sale) 	mod_skypeopen
CDR Stadistics 	Arternic cdr-stats 	??
Queue Statistics 	Asteriskguru queue-stats 	??
Web Management 	Freepbx 	??
IVR 	AGI / AMI 	Event Socket
Codec G.729 	Transcodind Cards
G.729 licenses
Free G.729 (Intel IPP) 	Transcodind Cards
G.729 licenses
fsg729 Intel IPP(any experience with it ? )
Fax Handling 	Iaxmodem with Hylafax 	??
Iaxmodem via asterisk to FS via SIP ?
SIP 	chan_sip 	sofia
ACD 	app_queue 	mod_callcenter



Thank you all


-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia

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