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Ok this is a really pointless discussion; Please use Asterisk or
FreeSWITCH forum for these things. This is not a debate forum.<br>
<br>
Thanks to everyone for thei wonderful feedback;<br>
<br>
<br>
<br>
On 12/10/10 10:31 AM, Laszlo wrote:
<blockquote
cite="mid:AANLkTi=vzyBMuM3zO_fpKz7WMcu7OA294UMVKRXx2d6Z@mail.gmail.com"
type="cite">Hmm, it's like Ferrari owners talking about which one
is better: Volkswagen or Toyota :)<br>
<br>
<div class="gmail_quote">2010/12/10 Aloysius Lloyd <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:lloyd.aloysius@gmail.com">lloyd.aloysius@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div><font face="verdana, sans-serif">Paul,</font></div>
<div><font face="verdana, sans-serif"><br>
</font></div>
<div><font face="verdana, sans-serif">I do not quite
understand what is "find me" doing with NAT</font></div>
<div><font face="verdana, sans-serif"><br>
</font></div>
<div><font face="verdana, sans-serif">Thanks</font></div>
<div><font face="verdana, sans-serif">Lloyd<font
color="#888888"><br>
</font></font>
<div>
<div class="h5">
<br>
<br>
<div class="gmail_quote">On Fri, Dec 10, 2010 at 10:11
AM, Jeff Pyle <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:jpyle@fidelityvoice.com"
target="_blank">jpyle@fidelityvoice.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt
0pt 0.8ex; border-left: 1px solid rgb(204, 204,
204); 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 moz-do-not-send="true"
href="mailto:paul.gore.j@gmail.com"
target="_blank">paul.gore.j@gmail.com</a>" <<a
moz-do-not-send="true"
href="mailto:paul.gore.j@gmail.com"
target="_blank">paul.gore.j@gmail.com</a>><br>
<div>
<div>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 moz-do-not-send="true"
href="mailto:paul.gore.j@gmail.com"
target="_blank">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 moz-do-not-send="true"
href="mailto:users-bounces@lists.opensips.org"
target="_blank">users-bounces@lists.opensips.org</a><br>
>[mailto:<a moz-do-not-send="true"
href="mailto:users-bounces@lists.opensips.org"
target="_blank">users-bounces@lists.opensips.org</a>]
Namens <a moz-do-not-send="true"
href="mailto:paul.gore.j@gmail.com"
target="_blank">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 moz-do-not-send="true"
href="mailto:msc@freeswitch.org"
target="_blank">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 moz-do-not-send="true"
href="mailto:dave.singer@wideideas.com"
target="_blank">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 moz-do-not-send="true"
href="mailto:msc@freeswitch.org"
target="_blank">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 moz-do-not-send="true"
href="mailto:paul.gore.j@gmail.com"
target="_blank">paul.gore.j@gmail.com</a> <<br>
>>>> <a moz-do-not-send="true"
href="mailto:paul.gore.j@gmail.com"
target="_blank">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 moz-do-not-send="true"
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 moz-do-not-send="true"
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>
>>>> <a class="moz-txt-link-freetext" href="IRC:mercutioviz">IRC:mercutioviz</a><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>
>_______________________________________________<br>
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><br>
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<br>
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