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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org