[Freeswitch-users] Asterisk to FreeSWITCH migration guide

[Avi Marcus]

Easier to do what? fs_cli -x "sofia status" or "show channels" or whatever
and you can do "as xml" so you can parse it easier.
But better is to just pick up a library for your language to make the ESL
stuff much easier.

-Avi

On Wed, Aug 10, 2011 at 3:54 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> 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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>
>
> _______________________________________________
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> http://www.cluecon.com 877-7-4ACLUE
>
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