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 _______________________________________________ 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