I have 5 Asterisk servers with an average of 6000 calls/day. All work at least eight months without stopping, somene with stops planned for maintenance (upgrade, patch, etc...).
Mounting the operating system properly and taking care of the major faults I see no problem with Asterisk. The two problems I faced with it Asterisk is up to 200 simultaneous calls, where Asterisk behaves in a weird way and AMI don't send some events. This is my experience with Asterisk. I may be correct as I could be wrong. Best regards, Rodrigo Lang. 2010/12/10 Laszlo <[email protected]> > 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 >> >> > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Rodrigo Lang Opening your mind - Just another Open Source site<http://openingyourmind.wordpress.com/>
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