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@tiendalinux.com>
To: freeswitch-users@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@tiendalinux.com
Email/MSN: nestor@tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia
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