[Freeswitch-users] Asterisk to FreeSWITCH migration guide

Thu Aug 11 23:55:34 MSD 2011

in your cdr records, all of the variables are present there and they are
template-able
freeswitch separates the logic from the call handling.

On Wed, Aug 10, 2011 at 11:12 AM, Sam <lakersman2006 at yahoo.com> wrote:

> So have you had to retrieve the dial status from bridging a call in
> freeswitch? For the life of me I cannot properly get the answered_time  when
> looking up the channel variables after the bridge call finishes an answered
> call.
>
> ------------------------------
> *From:* Moe Navid <manavid at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 1:44 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> There is no way by any means to compare Asterisk's AGI with the different
> facilities FreeSWITCH offers you in terms of controlling your call flow.
>
> For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones
> of channel locks and core dumps… Asterisk's dial status might seem
> compelling when you want to do simple things like calling cards etc… but
> when it comes to complex accounting and routing sky is limitless with the
> power of FreeSWITCH.
>
> I found FreeSWITCH's learning curve to be like vim, initially it may seem a
> bit difficult but in long run it pays of very well.
>
> If you know the difference between Dial command in Asterisk and Bridge in
> FreeSWITCH you would never go back to Asterisk. I give you just 3 simple
> examples:
> 1) Bridge command (via the channel variables) gives you the ability to
> control PDD on calls. Asterisk does not have such facility nonetheless it
> does not even bother to give you any useful information about your "Dial
> Status"! To control the PDD I had to tweak my kamailio.
>
> 2) If you want to implement a simple rate engine + fail over routing with
> asterisk + agi for failover you have to have a loop and watch for
> CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can
> just simply define your fail overs in your bridge args.
>
> 3) If you are in a cluster, have multiple gateways acting as proxy and you
> want to define outbound proxy for your carriers/endpoints you either have to
> define bunch of sip peers with outbound proxies or do it in dirty way which
> I did, I used to add a header in my outgoing calls X-Carrier-IP and had my
> kamailio to take care of the rest. In FreeSWITCH you just simply add
> ;fspath= to your bridge args.
>
> List can go on and on and on…
>
> Asterisk's dial status was the most annoying part of asterisk in my
> opinion :)
>
> On Aug 9, 2011, at 5:54 PM, Sam 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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>
> _______________________________________________
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>
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-- 
Anthony Minessale II

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