<div><font class="Apple-style-span" face="verdana, sans-serif">Paul,</font></div><div><font class="Apple-style-span" face="verdana, sans-serif"><br></font></div><div><font class="Apple-style-span" face="verdana, sans-serif">I do not quite understand what is "find me" doing with NAT</font></div>
<div><font class="Apple-style-span" face="verdana, sans-serif"><br></font></div><div><font class="Apple-style-span" face="verdana, sans-serif">Thanks</font></div><div><font class="Apple-style-span" face="verdana, sans-serif">Lloyd<br>
</font>
<br><br><div class="gmail_quote">On Fri, Dec 10, 2010 at 10:11 AM, Jeff Pyle <span dir="ltr"><<a href="mailto:jpyle@fidelityvoice.com">jpyle@fidelityvoice.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Guys,<br>
<br>
Point taken. Personally I prefer Coke over Pepsi.<br>
<br>
<br>
- Opensips user Jeff<br>
<br>
<br>
On 12/10/10 10:04 AM, "<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>" <<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>><br>
<div><div></div><div class="h5">wrote:<br>
<br>
>I haven't seen many posts from frustrated peole, majority of them come<br>
>from people either selling fs based services or part of fs development<br>
>team.<br>
>From my experience with fs 1.0.4 it was crashing every 2 months, 1.0.6 is<br>
>better, I already posted crashing rate for our use case.<br>
>I haven't experienced any stabilty issues with * 1.6 yet, but it only<br>
>sees light traffic.<br>
>FS is a great piece of software but it does have issues, sometimes even<br>
>simplest things like "find me" function work flawlessly in * and pain in<br>
>the ass to impelement in fs due to either bad nat handling or some other<br>
>bugs.<br>
><br>
><br>
>-----Original Message-----<br>
>From: Erik Dekkers<br>
>Sent: 12/10/2010 3:28:11 AM<br>
>To: '<a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>'; 'OpenSIPS users<br>
> mailling list'<br>
>Subject: RE: [OpenSIPS-Users] Freeswitch vs Asterisk<br>
><br>
>The reason people are yelling on the internet "Freeswitch is much better<br>
>than asterisk" is pure frustration.<br>
>They have used asterisk for years, were faced with crashes and since they<br>
>are using freeswitch they don't see those crashes anymore (apart from the<br>
>reason of those crashes).<br>
>No wonder they tell everyone freeswitch is better than asterisk. From<br>
>their point of view asterisk is bad.<br>
><br>
>It's not Mr. Collins opinion that asterisk is worse than freeswitch. It<br>
>are the ex-asterisk people who are saying that, think about that.<br>
><br>
>-----Oorspronkelijk bericht-----<br>
>Van: <a href="mailto:users-bounces@lists.opensips.org">users-bounces@lists.opensips.org</a><br>
>[mailto:<a href="mailto:users-bounces@lists.opensips.org">users-bounces@lists.opensips.org</a>] Namens <a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a><br>
>Verzonden: donderdag 9 december 2010 16:27<br>
>Aan: OpenSIPS users mailling list<br>
>Onderwerp: Re: [OpenSIPS-Users] Freeswitch vs Asterisk<br>
><br>
>I just want to reply to mr Collins with FS: your post looks very much<br>
>like advertisement, and I have seen that "fs is so much better than *"<br>
>all over internet from people connected to fs. That is unethical to say<br>
>the least.<br>
>In fact we have exprerienced fs crashes with core dump at least once in<br>
>6 months and we process just under 40K calls/month.<br>
>As to "nat tools" which you mentioned they just do not work. In fact<br>
>usually * box works much better for natted users.<br>
>As to xml curl interface - we do use it, and it's a pathetic way to feed<br>
>a dialplan to a switch, since it's inefficient resource wise, but there<br>
>was no other way available for real time solution where's * supports real<br>
>time db out of the box.<br>
>Trust me we do have development experience with both * socket interface<br>
>and fs one, and in my opinion * solution is far better and has far less<br>
>bugs.<br>
><br>
>-----Original Message-----<br>
>From: James Mbuthia<br>
>Sent: 12/08/2010 5:55:42 PM<br>
>Subject: Re: [OpenSIPS-Users] Freeswitch vs Asterisk<br>
><br>
>From the comments mentioned it seems FS meets my core requirements which<br>
>are scalability and stability. I don't have the financial and manpower<br>
>resources for a large scale implementation so am looking at getting a<br>
>high end server and a solution that can scale well until I can through in<br>
>more resources. It seems also FS is more stable than * which is a huge<br>
>plus for a small operation like mine and since I only need few features<br>
>from the solutions available then FS makes more sense<br>
><br>
>On Wed, Dec 8, 2010 at 8:29 PM, Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>><br>
>wrote:<br>
><br>
>> Dave,<br>
>><br>
>> Thanks for your two cents. :)<br>
>><br>
>> Regarding the PRI stuff, Sangoma is really doing a lot with FreeTDM<br>
>> (the replacement for OpenZAP) and it will be a full-featured PRI<br>
>> stack. If you're missing anything in the PRI implementation then<br>
>> Moises Silva would definitely want to hear about it.<br>
>><br>
>> On the voicemail stuff we have heard similar reports. In fact, we have<br>
>> an intrepid community member who is building "Jester Mail" as a FS<br>
>> alternative to Asterisk's Comedian mail. The basic idea is that Jester<br>
>> Mail will be 100% customizable such that you can drop in FS as a<br>
>> replacement for Asterisk and your voicemail users would be none the<br>
>>wiser.<br>
>><br>
>> By early next year you will probably have more options if you wish to<br>
>> swap out your remaining Asterisk servers.<br>
>><br>
>> -MC<br>
>><br>
>><br>
>> On Wed, Dec 8, 2010 at 9:53 AM, Dave Singer<br>
>><<a href="mailto:dave.singer@wideideas.com">dave.singer@wideideas.com</a>>wrote:<br>
>><br>
>>> We have both asterisk and Freeswitch in production. The primary place<br>
>>> where we have * installed is as a pbx for our business customers<br>
>>> (where we started doing business and didn't know any better). We are<br>
>>> still using * for them for two reasons: migration time and voicemail<br>
>>> app I feel is still better in a couple points. They are low volume<br>
>>> usage so crashes are very rare.<br>
>>> We also have some boxes where we connect to telecom PRI circuits<br>
>>> where the API for FS doesn't support some params we need to set. So<br>
>>> we are stuck there for now. There systems handle moderate volume, 30 -<br>
>>>90 simultaneous calls.<br>
>>> This call volume has proved to be deadly to asterisk and we have to<br>
>>> restart asterisk daily or suffer a crash in the middle of peek times.<br>
>>> We use FreeSwitch as the workhorse with a custom routing module<br>
>>> combined with Opensips as a class 4 switch (whole sale trunking<br>
>>> service). With high powered servers (latest dual xeon quad core, 16GB<br>
>>> ram, and 10Gbit ethernet) it can handle thousands of simultaneous<br>
>>> calls. They run for months without problem (would be longer but for<br>
>>> reboots for upgrades, etc., not FS crashes).<br>
>>> We also have a class 5 system that handles residential users which<br>
>>> uses FS and opensips for failover. Again no FS crashes.<br>
>>> FS is also our conference server for all our services.<br>
>>><br>
>>> We started out using * building the business PBXs. Later found FS as<br>
>>> we were developing the residential system and converted to using it.<br>
>>> Coming from * to FS has some difficulties because of the different<br>
>>> ways of doing things like the flow of the dialplan where all<br>
>>> conditions are evaluated at the time of entry to the dialplan, not as<br>
>>> each line is executed (executing another extension solved this problem<br>
>>>for me).<br>
>>> I do think FS has a little higher learning curve, I have found it<br>
>>> better in almost every area, especially stability and flexibility.<br>
>>><br>
>>> Well, those are my 2 cents. :-D<br>
>>> Dave<br>
>>><br>
>>> On Tue, Dec 7, 2010 at 11:27 AM, Michael Collins<br>
>>><<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>>wrote:<br>
>>><br>
>>>> Comments inline. (Full disclosure: I am on the FreeSWITCH team, so<br>
>>>> if I come off as biased then you know why. ;)<br>
>>>><br>
>>>> On Tue, Dec 7, 2010 at 8:29 AM, <a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a> <<br>
>>>> <a href="mailto:paul.gore.j@gmail.com">paul.gore.j@gmail.com</a>> wrote:<br>
>>>><br>
>>>>> We use freeswitch in prod alone, no opensips yet. I would say fs is<br>
>>>>> definetly more scalable than *.<br>
>>>>> Stability wise seems like fs is on par with *.<br>
>>>>><br>
>>>> YMMV, but a large percentage of FreeSWITCH users have abandoned<br>
>>>> Asterisk specifically because of stability issues, like random and<br>
>>>> inexplicable crashes.<br>
>>>><br>
>>>><br>
>>>>> * has substantially better interface for control over socket<br>
>>>>> connection<br>
>>>>> - it's easier to implement and it's more consistent.<br>
>>>>><br>
>>>> This statement is patently false. The FreeSWITCH event socket<br>
>>>> interface is incredibly powerful and is absolutely more consistent<br>
>>>> than the AMI. Those wondering about inconsistencies in the AMI<br>
>>>> should listen to a seasoned AMI developer talk about the challenges:<br>
>>>> <a href="http://www.viddler.com/explore/cluecon/videos/29/" target="_blank">http://www.viddler.com/explore/cluecon/videos/29/</a><br>
>>>><br>
>>>><br>
>>>>> Configuration wise, I think * is easier, xml- based approach in fs<br>
>>>>> is cumbersome and has no real advantage over *.<br>
>>>>><br>
>>>> This one really is like Coke vs. Pepsi. Some people hate XML, some<br>
>>>> people hate INI-style config files. Personally, I've done both and<br>
>>>> now that I'm accustomed to FreeSWITCH's XML files I find them much<br>
>>>> easier to read than Asterisk's config files. There is one "real<br>
>>>> advantage" to using XML for configs and that is that machines and<br>
>>>> humans can both produce XML, so it's relatively simple to let a<br>
>>>>machine generate XML-based configs on the fly.<br>
>>>> (FreeSWITCH uses "mod_xml_curl" as the basis for dynamic<br>
>>>> configuration - it's very cool and I recommend that you check it<br>
>>>> out.)<br>
>>>><br>
>>>><br>
>>>>> We have endless problems with fs nat handling, lots of no audio<br>
>>>>> issues with end users behind a nat. That's why we want to try<br>
>>>>> opensips solution for that.<br>
>>>>><br>
>>>> Almost all NAT problems stem from phones which don't handle NAT<br>
>>>> properly or NAT devices that scramble ports and IP addresses when<br>
>>>> packets pass through. FreeSWITCH has several NAT-busting tools to<br>
>>>> assist the system admin. Some tools are for when FS is behind NAT,<br>
>>>> others are for when the phones are behind NAT. Bottom line is this:<br>
>>>> if the NAT device and the phones are not horribly broken then FS<br>
>>>> works great with NAT and in many cases "just works." However, when<br>
>>>> you start mixing crazy scenarios with broken phones then bad things<br>
>>>> will happen. Example: Polycom phones are wonderful except that they<br>
>>>> don't support rport - FS has a mechanism to assist with this but if<br>
>>>> you turn it on to "fix" the Polycom phones then it will break all<br>
>>>> other phone types. (There is a limit to the amount of pandering that<br>
>>>> the FS devs will do in order to interop with broken devices. In many<br>
>>>> cases they simply say "NO" to doing stupid things in order to work<br>
>>>> with broken devices. If you must work with such a device then<br>
>>>> perhaps FreeSWITCH isn't for you.)<br>
>>>><br>
>>>> All that being said, the FreeSWITCH developers have a simple mantra<br>
>>>> that they follow to the letter: Use what works for your situation.<br>
>>>> If Asterisk works for you then by all means use it! You won't hurt<br>
>>>> our feelings. (I work daily with the FreeSWITCH dev team.) If you<br>
>>>> have people knowledgeable in Asterisk or FreeSWITCH then it might be<br>
>>>> advantageous to go with the project for which you have more<br>
>>>> resources. In any case, if you are interested in FreeSWITCH we have<br>
>>>> a great IRC channel (#freeswitch on <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a>), an actively<br>
>>>> mailing list, and a small but growing international community of<br>
>>>>users. You are most welcome to join us to see what we're about.<br>
>>>><br>
>>>> Happy VoIPing!<br>
>>>> -Michael S Collins<br>
>>>> IRC:mercutioviz<br>
>>>><br>
>>>><br>
>>>><br>
>>>>><br>
>>>>><br>
>>>>> -----Original Message-----<br>
>>>>> From: James Mbuthia<br>
>>>>> Sent: 12/07/2010 8:54:51 AM<br>
>>>>> Subject: [OpenSIPS-Users] Freeswitch vs Asterisk<br>
>>>>><br>
>>>>> Hi guys,<br>
>>>>><br>
>>>>> I want to integrate my Opensips implementation with either Asterisk<br>
>>>>> or Freeswitch to do the following functions<br>
>>>>><br>
>>>>> - Act as a Media server<br>
>>>>> - Connect to the PSTN<br>
>>>>> - Act as a B2BUA<br>
>>>>><br>
>>>>><br>
>>>>> There's been alot of hype about Freeswitch and I wanted to know<br>
>>>>> from people who've integrated it to OpenSIPS how it compares to<br>
>>>>> Asterisk especially in the case of installation and intergration,<br>
>>>>> scalability and ease of maintenance. Any info would be a huge help<br>
>>>>><br>
>>>>> regards,<br>
>>>>> james<br>
>>>>><br>
><br>
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</div></div></blockquote></div><br></div>