[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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