=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@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@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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