[Freeswitch-users] Asterisk to FreeSWITCH migration guide
Steven Ayre
steveayre at gmail.com
Wed Aug 10 13:51:44 MSD 2011
I see Avi Marcus has already directed you to the Wiki page.
As with Asterisk there's no official GUI, but there are several open source
projects providing one. Or you can build your own. A few include blue.box
(2800hz project) and FusionPBX.
http://wiki.freeswitch.org/wiki/Freeswitch_Gui
There are 2 ACD modules: mod_fifo and mod_callcentre. mod_fifo is the older
one. That doesn't mean it's not as good, they just approach the problem in
different ways.
As for G729 which you mentioned... do NOT use the Intel IPP codec. It is
ILLEGAL unless you have purchased a valid licence for it, which is extremely
unlikely. You can support it using a hardware transcoding card (Sangoma),
mod_com_g729 (version licensed by FreeSWITCH) and mod_g729 (which is
passthrough only, no transcoding but fine for bridging calls).
chan_mobile's closest match is probably mod_gsmopen. I believe it uses a
cable rather than bluetooth though, and is faily new so probably
'experimental'.
IAX2 is supported by mod_opal. As you noted though, it isn't as stable as
say mod_sofia.
For fax handling check the T38 functionality provided by mod_spandsp.
-Steve
On 9 August 2011 17:48, Nestor A Diaz <nestor at tiendalinux.com> wrote:
> **
> 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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