Guys, Point taken. Personally I prefer Coke over Pepsi.
- Opensips user Jeff On 12/10/10 10:04 AM, "[email protected]" <[email protected]> wrote: >I haven't seen many posts from frustrated peole, majority of them come >from people either selling fs based services or part of fs development >team. >From my experience with fs 1.0.4 it was crashing every 2 months, 1.0.6 is >better, I already posted crashing rate for our use case. >I haven't experienced any stabilty issues with * 1.6 yet, but it only >sees light traffic. >FS is a great piece of software but it does have issues, sometimes even >simplest things like "find me" function work flawlessly in * and pain in >the ass to impelement in fs due to either bad nat handling or some other >bugs. > > >-----Original Message----- >From: Erik Dekkers >Sent: 12/10/2010 3:28:11 AM >To: '[email protected]'; 'OpenSIPS users > mailling list' >Subject: RE: [OpenSIPS-Users] Freeswitch vs Asterisk > >The reason people are yelling on the internet "Freeswitch is much better >than asterisk" is pure frustration. >They have used asterisk for years, were faced with crashes and since they >are using freeswitch they don't see those crashes anymore (apart from the >reason of those crashes). >No wonder they tell everyone freeswitch is better than asterisk. From >their point of view asterisk is bad. > >It's not Mr. Collins opinion that asterisk is worse than freeswitch. It >are the ex-asterisk people who are saying that, think about that. > >-----Oorspronkelijk bericht----- >Van: [email protected] >[mailto:[email protected]] Namens [email protected] >Verzonden: donderdag 9 december 2010 16:27 >Aan: OpenSIPS users mailling list >Onderwerp: Re: [OpenSIPS-Users] Freeswitch vs Asterisk > >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 > > >_______________________________________________ >Users mailing list >[email protected] >http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
