[Freeswitch-users] Asterisk to FreeSWITCH migration guide

Thu Aug 11 06:40:31 MSD 2011

hi sam,

i found this post
http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477

modify the script to suit your need.

hope it helps.  just dig on w/ FS :-)
-nandy

On Thu, Aug 11, 2011 at 12:28 AM, Sam <lakersman2006 at yahoo.com> wrote:

> Thanks for being so accommodating. I was a bit frustrated in trying to port
> over an asterisk agi script to freeswitch. I have spent many hours trying to
> learn how to configure freeswitch, I was about to give up, but I will play
> with the new changes you made and see if that works for me.
>
> One other question, when the bridged call hangs up I do not see any value
> for the hangup time when using getVariable("hangup_time"), so how can I get
> it?
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =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
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/e6395121/attachment-0001.html 