I just want to reply to mr Collins with FS: your post looks very much like advertisement, and I have seen that "fs is so much better than *" all over internet from people connected to fs. That is unethical to say the least. In fact we have exprerienced fs crashes with core dump at least once in 6 months and we process just under 40K calls/month. As to "nat tools" which you mentioned they just do not work. In fact usually * box works much better for natted users. As to xml curl interface - we do use it, and it's a pathetic way to feed a dialplan to a switch, since it's inefficient resource wise, but there was no other way available for real time solution where's * supports real time db out of the box. Trust me we do have development experience with both * socket interface and fs one, and in my opinion * solution is far better and has far less bugs.
-----Original Message----- From: James Mbuthia Sent: 12/08/2010 5:55:42 PM Subject: Re: [OpenSIPS-Users] Freeswitch vs Asterisk >From the comments mentioned it seems FS meets my core requirements which are scalability and stability. I don't have the financial and manpower resources for a large scale implementation so am looking at getting a high end server and a solution that can scale well until I can through in more resources. It seems also FS is more stable than * which is a huge plus for a small operation like mine and since I only need few features from the solutions available then FS makes more sense On Wed, Dec 8, 2010 at 8:29 PM, Michael Collins <[email protected]> wrote: > Dave, > > Thanks for your two cents. :) > > Regarding the PRI stuff, Sangoma is really doing a lot with FreeTDM (the > replacement for OpenZAP) and it will be a full-featured PRI stack. If you're > missing anything in the PRI implementation then Moises Silva would > definitely want to hear about it. > > On the voicemail stuff we have heard similar reports. In fact, we have an > intrepid community member who is building "Jester Mail" as a FS alternative > to Asterisk's Comedian mail. The basic idea is that Jester Mail will be 100% > customizable such that you can drop in FS as a replacement for Asterisk and > your voicemail users would be none the wiser. > > By early next year you will probably have more options if you wish to swap > out your remaining Asterisk servers. > > -MC > > > On Wed, Dec 8, 2010 at 9:53 AM, Dave Singer <[email protected]>wrote: > >> We have both asterisk and Freeswitch in production. The primary place >> where we have * installed is as a pbx for our business customers (where we >> started doing business and didn't know any better). We are still using * for >> them for two reasons: migration time and voicemail app I feel is still >> better in a couple points. They are low volume usage so crashes are very >> rare. >> We also have some boxes where we connect to telecom PRI circuits where the >> API for FS doesn't support some params we need to set. So we are stuck there >> for now. There systems handle moderate volume, 30 - 90 simultaneous calls. >> This call volume has proved to be deadly to asterisk and we have to restart >> asterisk daily or suffer a crash in the middle of peek times. >> We use FreeSwitch as the workhorse with a custom routing module combined >> with Opensips as a class 4 switch (whole sale trunking service). With high >> powered servers (latest dual xeon quad core, 16GB ram, and 10Gbit ethernet) >> it can handle thousands of simultaneous calls. They run for months without >> problem (would be longer but for reboots for upgrades, etc., not FS >> crashes). >> We also have a class 5 system that handles residential users which uses FS >> and opensips for failover. Again no FS crashes. >> FS is also our conference server for all our services. >> >> We started out using * building the business PBXs. Later found FS as we >> were developing the residential system and converted to using it. >> Coming from * to FS has some difficulties because of the different ways of >> doing things like the flow of the dialplan where all conditions are >> evaluated at the time of entry to the dialplan, not as each line is executed >> (executing another extension solved this problem for me). >> I do think FS has a little higher learning curve, I have found it better >> in almost every area, especially stability and flexibility. >> >> Well, those are my 2 cents. :-D >> Dave >> >> On Tue, Dec 7, 2010 at 11:27 AM, Michael Collins <[email protected]>wrote: >> >>> Comments inline. (Full disclosure: I am on the FreeSWITCH team, so if I >>> come off as biased then you know why. ;) >>> >>> On Tue, Dec 7, 2010 at 8:29 AM, [email protected] < >>> [email protected]> wrote: >>> >>>> We use freeswitch in prod alone, no opensips yet. I would say fs is >>>> definetly more scalable than *. >>>> Stability wise seems like fs is on par with *. >>>> >>> YMMV, but a large percentage of FreeSWITCH users have abandoned Asterisk >>> specifically because of stability issues, like random and inexplicable >>> crashes. >>> >>> >>>> * has substantially better interface for control over socket connection >>>> - it's easier to implement and it's more consistent. >>>> >>> This statement is patently false. The FreeSWITCH event socket interface >>> is incredibly powerful and is absolutely more consistent than the AMI. Those >>> wondering about inconsistencies in the AMI should listen to a seasoned AMI >>> developer talk about the challenges: >>> http://www.viddler.com/explore/cluecon/videos/29/ >>> >>> >>>> Configuration wise, I think * is easier, xml- based approach in fs is >>>> cumbersome and has no real advantage over *. >>>> >>> This one really is like Coke vs. Pepsi. Some people hate XML, some people >>> hate INI-style config files. Personally, I've done both and now that I'm >>> accustomed to FreeSWITCH's XML files I find them much easier to read than >>> Asterisk's config files. There is one "real advantage" to using XML for >>> configs and that is that machines and humans can both produce XML, so it's >>> relatively simple to let a machine generate XML-based configs on the fly. >>> (FreeSWITCH uses "mod_xml_curl" as the basis for dynamic configuration - >>> it's very cool and I recommend that you check it out.) >>> >>> >>>> We have endless problems with fs nat handling, lots of no audio issues >>>> with end users behind a nat. That's why we want to try opensips solution >>>> for >>>> that. >>>> >>> Almost all NAT problems stem from phones which don't handle NAT properly >>> or NAT devices that scramble ports and IP addresses when packets pass >>> through. FreeSWITCH has several NAT-busting tools to assist the system >>> admin. Some tools are for when FS is behind NAT, others are for when the >>> phones are behind NAT. Bottom line is this: if the NAT device and the phones >>> are not horribly broken then FS works great with NAT and in many cases "just >>> works." However, when you start mixing crazy scenarios with broken phones >>> then bad things will happen. Example: Polycom phones are wonderful except >>> that they don't support rport - FS has a mechanism to assist with this but >>> if you turn it on to "fix" the Polycom phones then it will break all other >>> phone types. (There is a limit to the amount of pandering that the FS devs >>> will do in order to interop with broken devices. In many cases they simply >>> say "NO" to doing stupid things in order to work with broken devices. If you >>> must work with such a device then perhaps FreeSWITCH isn't for you.) >>> >>> All that being said, the FreeSWITCH developers have a simple mantra that >>> they follow to the letter: Use what works for your situation. If Asterisk >>> works for you then by all means use it! You won't hurt our feelings. (I work >>> daily with the FreeSWITCH dev team.) If you have people knowledgeable in >>> Asterisk or FreeSWITCH then it might be advantageous to go with the project >>> for which you have more resources. In any case, if you are interested in >>> FreeSWITCH we have a great IRC channel (#freeswitch on irc.freenode.net), >>> an actively mailing list, and a small but growing international community of >>> users. You are most welcome to join us to see what we're about. >>> >>> Happy VoIPing! >>> -Michael S Collins >>> IRC:mercutioviz >>> >>> >>> >>>> >>>> >>>> -----Original Message----- >>>> From: James Mbuthia >>>> Sent: 12/07/2010 8:54:51 AM >>>> Subject: [OpenSIPS-Users] Freeswitch vs Asterisk >>>> >>>> Hi guys, >>>> >>>> I want to integrate my Opensips implementation with either Asterisk or >>>> Freeswitch to do the following functions >>>> >>>> - Act as a Media server >>>> - Connect to the PSTN >>>> - Act as a B2BUA >>>> >>>> >>>> There's been alot of hype about Freeswitch and I wanted to know from >>>> people >>>> who've integrated it to OpenSIPS how it compares to Asterisk especially >>>> in >>>> the case of installation and intergration, scalability and ease of >>>> maintenance. Any info would be a huge help >>>> >>>> regards, >>>> james >>>> _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
