Ok this is a really pointless discussion; Please use Asterisk or
FreeSWITCH forum for these things. This is not a debate forum.
Thanks to everyone for thei wonderful feedback;
On 12/10/10 10:31 AM, Laszlo wrote:
Hmm, it's like Ferrari owners talking about which one is better:
Volkswagen or Toyota :)
2010/12/10 Aloysius Lloyd <[email protected]
<mailto:[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] <mailto:[email protected]>> wrote:
Guys,
Point taken. Personally I prefer Coke over Pepsi.
- Opensips user Jeff
On 12/10/10 10:04 AM, "[email protected]
<mailto:[email protected]>" <[email protected]
<mailto:[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] <mailto:[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]>
>[mailto:[email protected]
<mailto:[email protected]>] Namens
[email protected] <mailto:[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] <mailto:[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]
<mailto:[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] <mailto:[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]
<mailto:[email protected]> <
>>>> [email protected] <mailto:[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
<http://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
>>>>>
>
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