No subject
Thu Dec 2 10:23:01 CET 2010
e scalability and stability. I don't have the financial and manpower re=
sources for a large scale implementation so am looking at getting a high en=
d server and a solution that can scale well until I can through in more res=
ources. It seems also FS is more stable than * which is a huge plus for a s=
mall operation like mine and since I only need few features from the soluti=
ons available then FS makes more sense<br>
<br><div class=3D"gmail_quote">On Wed, Dec 8, 2010 at 8:29 PM, Michael Coll=
ins <span dir=3D"ltr"><<a href=3D"mailto:msc at freeswitch.org">msc at freeswi=
tch.org</a>></span> wrote:<br><blockquote class=3D"gmail_quote" style=3D=
"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Dave,<div><br></div><div>Thanks for your two cents. :)</div><div><br></div>=
<div>Regarding the PRI stuff, Sangoma is really doing a lot with FreeTDM (t=
he replacement for OpenZAP) and it will be a full-featured PRI stack. If yo=
u're missing anything in the PRI implementation then Moises Silva would=
definitely want to hear about it.</div>
<div><br></div><div>On the voicemail stuff we have heard similar reports. I=
n fact, we have an intrepid community member who is building "Jester M=
ail" as a FS alternative to Asterisk's Comedian mail. The basic id=
ea 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.=A0</div>
<div><br></div><div>By early next year you will probably have more options =
if you wish to swap out your remaining Asterisk servers.</div><div><br></di=
v><div><font color=3D"#888888">-MC</font><div><div></div><div class=3D"h5">
<br><br><div class=3D"gmail_quote">On Wed, Dec 8, 2010 at 9:53 AM, Dave Sin=
ger <span dir=3D"ltr"><<a href=3D"mailto:dave.singer at wideideas.com" targ=
et=3D"_blank">dave.singer at wideideas.com</a>></span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">We have both asterisk and Freeswitch in prod=
uction. The primary place where we have * installed is as a pbx for our bus=
iness customers (where we started doing business and didn't know any be=
tter). We are still using * for them for two reasons: migration time and vo=
icemail app I feel is still better in a couple points. They are low volume =
usage so crashes are very rare.<div>
We also have some boxes where we connect to telecom PRI=A0circuits=A0where =
the API for FS doesn't support some params we need to set. So we are st=
uck there for now. There systems handle moderate volume, 30 - 90=A0simultan=
eous=A0calls. This call volume has proved to be deadly to asterisk and we h=
ave to restart asterisk daily or suffer a crash in the middle of peek times=
.</div>
<div>We use FreeSwitch as the workhorse with a custom routing module combin=
ed with Opensips as a class 4 switch (whole sale trunking service). With hi=
gh powered servers (latest dual xeon quad core, 16GB ram, and 10Gbit ethern=
et) it can handle thousands of=A0simultaneous=A0calls. They run for months =
without problem (would be longer but for reboots for upgrades, etc., not FS=
crashes).<br>
We also have a class 5 system that handles residential users which uses FS =
and opensips for failover. Again no FS crashes.</div><div>FS is also our co=
nference server for all our services.</div><div><br></div><div>We started o=
ut using * building the business PBXs. Later found FS as we were developing=
the residential system and converted to using it.</div>
<div>Coming from * to FS has some=A0difficulties because of the different w=
ays 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 execute=
d (executing another extension solved this problem for me).</div>
<div>I do think FS has a little higher learning curve, I have found it bett=
er in almost every area, especially stability and=A0flexibility.</div><div>=
<br></div><div>Well, those are my 2 cents. :-D</div><div>Dave</div><div>
<div></div><div><div>
<br>
<div class=3D"gmail_quote">On Tue, Dec 7, 2010 at 11:27 AM, Michael Collins=
<span dir=3D"ltr"><<a href=3D"mailto:msc at freeswitch.org" target=3D"_bla=
nk">msc at freeswitch.org</a>></span> wrote:<br><blockquote class=3D"gmail_=
quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1=
ex">
Comments inline. (Full disclosure: I am on the FreeSWITCH team, so if I com=
e off as biased then you know why. ;)<br><br><div class=3D"gmail_quote"><di=
v>On Tue, Dec 7, 2010 at 8:29 AM, <a href=3D"mailto:paul.gore.j at gmail.com" =
target=3D"_blank">paul.gore.j at gmail.com</a> <span dir=3D"ltr"><<a href=
=3D"mailto:paul.gore.j at gmail.com" target=3D"_blank">paul.gore.j at gmail.com</=
a>></span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">We use freeswitch in prod alone, no opensips=
yet. I would say fs is definetly more scalable than *.<br>
Stability wise seems like fs is on par with *.<br></blockquote></div><div>Y=
MMV, but a large percentage of FreeSWITCH users have abandoned Asterisk spe=
cifically because of stability issues, like random and inexplicable crashes=
.</div>
<div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
* has substantially better interface for control over socket connection - i=
t's easier to implement and it's more consistent.<br></blockquote><=
/div><div>This statement is patently false. The FreeSWITCH event socket int=
erface is incredibly powerful and is absolutely more consistent than the AM=
I. Those wondering about inconsistencies in the AMI should listen to a seas=
oned AMI developer talk about the challenges:</div>
<div><a href=3D"http://www.viddler.com/explore/cluecon/videos/29/" target=
=3D"_blank">http://www.viddler.com/explore/cluecon/videos/29/</a></div><div=
><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex">
Configuration wise, I think * is easier, xml- based approach in fs is cumbe=
rsome and has no real advantage over *.<br></blockquote></div><div>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 acc=
ustomed to FreeSWITCH's XML files I find them much easier to read than =
Asterisk's config files. There is one "real advantage" to usi=
ng XML for configs and that is that machines and humans can both produce XM=
L, so it's relatively simple to let a machine generate XML-based config=
s on the fly. (FreeSWITCH uses "mod_xml_curl" as the basis for dy=
namic configuration - it's very cool and I recommend that you check it =
out.)</div>
<div>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;=
border-left:1px #ccc solid;padding-left:1ex">
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 fo=
r that.<br></blockquote></div><div>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 th=
e NAT device and the phones are not horribly broken then FS works great wit=
h NAT and in many cases "just works." However, when you start mix=
ing crazy scenarios with broken phones then bad things will happen. Example=
: Polycom phones are wonderful except that they don't support rport - F=
S has a mechanism to assist with this but if you turn it on to "fix&qu=
ot; 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 int=
erop with broken devices. In many cases they simply say "NO" to d=
oing stupid things in order to work with broken devices. If you must work w=
ith such a device then perhaps FreeSWITCH isn't for you.)</div>
<div><br></div><div>All that being said, the FreeSWITCH developers have a s=
imple mantra that they follow to the letter: Use what works for your situat=
ion. 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 peo=
ple knowledgeable in Asterisk or FreeSWITCH then it might be advantageous t=
o go with the project for which you have more resources. In any case, if yo=
u are interested in FreeSWITCH we have a great IRC channel (#freeswitch on =
<a href=3D"http://irc.freenode.net" target=3D"_blank">irc.freenode.net</a>)=
, 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.</di=
v>
<div><br></div><div>Happy VoIPing!</div><div>-Michael S Collins</div><div>I=
RC:mercutioviz</div><div><div><br></div><div>=A0</div><blockquote class=3D"=
gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-=
left:1ex">
<div><div></div><div><br>
<br>
-----Original Message-----<br>
From: James Mbuthia<br>
Sent: =A012/07/2010 8:54:51 AM<br>
Subject: =A0[OpenSIPS-Users] Freeswitch vs Asterisk<br>
<br>
Hi guys,<br>
<br>
I want to integrate my Opensips implementation with either Asterisk or<br>
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 from pe=
ople<br>
who've integrated it to OpenSIPS how it compares to Asterisk especially=
in<br>
the case of installation and intergration, scalability and ease of<br>
maintenance. =A0Any info would be a huge help<br>
<br>
regards,<br>
james<br>
<br>
</div></div>:::0:a0e8dc7ff9acb0ae85abefba43f14c73:-1:x:::<br>
<br>
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