Hmm, it's like Ferrari owners talking about which one is better: Volkswagen or Toyota :)
2010/12/10 Aloysius Lloyd <[email protected]> > Paul, > > I do not quite understand what is "find me" doing with NAT > > Thanks > Lloyd > > > On Fri, Dec 10, 2010 at 10:11 AM, Jeff Pyle <[email protected]>wrote: > >> 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 >> > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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