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
