[Freeswitch-users] Asterisk to FreeSWITCH migration guide

[Anthony Minessale]

=D

ok, sure.  If that's your only complaint.... see
commit 9d98d49f0556fb79656c8403f285ae0a615439d3

Some caveats

1) There is actually less specific, more generalized data in this DIALSTATUS
variable than what we already report, when you're ready to move on see the
originate_disposition variable:  It's kind of like going from reporting the
precise geo-location of a cafe in Paris to generalizing it to "EUROPE"

We follow the Q.850 standard for call cause codes and follow the SIP RFC to
map sip response codes to/from the Q.850 equivalent.  Also each module has
its own version "sip_hangup_disposition" for sip so you have both the real
sip response code AND the official Q.850 equiv variables set on each call.


2) We don't have a torture feature so we never return that code.


3) Since our originate can return before a call is answered I added "EARLY"
which means the originate succeeded but its still not answered.

4) For any others that do not map directly to FreeSWITCH, I just installed a
copy of originate_disposition for good measure.

P.S

This email took longer to compose than the patch did while sitting in the
middle of a crowded room so you probably could have simply parsed the
originate originate_disposition yourself but if it helps people get over
being stuck in a paradigm, it's worth it for me to write........


On Tue, Aug 9, 2011 at 7:54 PM, 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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> http://www.cluecon.com 877-7-4ACLUE
>
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>
>
> _______________________________________________
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> http://www.cluecon.com 877-7-4ACLUE
>
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>
>


-- 
Anthony Minessale II

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