Discussion:
[Freeswitch-users] New to FreeSwitch
Ritesh Singh
2008-03-28 13:07:45 UTC
Permalink
Hi All,

I am very new to freeswitch. It will be great if some one can tell me few
things:

1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. Then what
changes i am supposed to do and at what place. I would like to use the
mod_portaudio for this purpose.

2) I started the windows freeswitch and loaded port_audio by using "load
mod_portaudio" , then used "pa call 1234"...the session gets initiated but
the call gets hanged up with the log "portaudio/1234 [CS_RING]
[NO_ROUTE_DESTINATION]".

3) Suppose i have a jabber server x.net. and i have 2 users having account
at that jabber server. Then how can i use mod_dingaling so that i have the
voice chat between those user of the jabber server x.net.
Also, i would like to dump the all the record regarding the call made, like
duration of call , person who initiated the call, call destination , etc.

Any help in this regard is highly appreciable. Please do forgive me if you
think its stupid mail but i am desparate for these answers.

This is my second mail as no one replied my first mail. Please do reply this
mail else i ll be broken ....

Thanks and Regards
Ritesh
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Brian West
2008-03-28 14:57:23 UTC
Permalink
Post by Ritesh Singh
Hi All,
I am very new to freeswitch. It will be great if some one can tell
Welcome to the community.
Post by Ritesh Singh
1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y.
Then what changes i am supposed to do and at what place. I would
like to use the mod_portaudio for this purpose.
2) I started the windows freeswitch and loaded port_audio by using
"load mod_portaudio" , then used "pa call 1234"...the session gets
initiated but the call gets hanged up with the log "portaudio/1234
[CS_RING] [NO_ROUTE_DESTINATION]".
I'm sure this message is rather clear... You do not have a route in
your dialplan for 1234 thus it fails.
Post by Ritesh Singh
3) Suppose i have a jabber server x.net. and i have 2 users having
account at that jabber server. Then how can i use mod_dingaling so
that i have the voice chat between those user of the jabber server
x.net.
Also, i would like to dump the all the record regarding the call
made, like duration of call , person who initiated the call, call
destination , etc.
Feels like your trying to do too much at once before you fully
understand what is going on.
Post by Ritesh Singh
Any help in this regard is highly appreciable. Please do forgive
me if you think its stupid mail but i am desparate for these answers.
You're trying to fly before you can even crawl. You have many things
to learn.

Best things to do are start here http://wiki.freeswitch.org and
#freeswitch on irc.freenode.net

Remember I'll be very glad to answer questions but the requirement is
that you MUST put the info you learn on the wiki and pay it forward to
others.

Thanks,
/b



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Michael Collins
2008-03-28 19:01:15 UTC
Permalink
Ritesh,



Welcome to the FS community! Brian already mentioned a few things, like
the wiki and the IRC channel, so definitely get familiar with those.



One thing I'd like to recommend is that you check out the dialplan xml
files. (I hope you're comfortable with XML!) In the freeswitch conf
directory look under dialplan for "default.xml" - this has the pre-built
dialplan. (You did do "make samples" after the initial install, didn't
you? :-)) Freeswtich uses regular expressions for matching various
values in the dialplan. (I hope you like regular expressions! :-))
Here's an example right from the sample dialplan:



<extension name="echo">

<condition field="destination_number" expression="^9996$">

<action application="answer"/>

<action application="send_display" data="Echo Test"/>

<action application="echo"/>

</condition>

</extension>



This is the sample echo test extension. XML helps make some of the
features obvious, like the extension name. The most interesting thing
here is the "condition" tag. That line essential says, "If the
destination number is exactly 9996 then execute these actions." The
actions are "answer" - that answer the call, then display "Echo Test" on
the display, and lastly launch the echo application. If you want to
hear the echo test then try this:

pa call 9996



If you get the echo test then you know the basic setup of the dial plan
and port audio is working. Please try it out and report back to the
list or to IRC. (BTW, do you have an IRC screen name?) We'll go from
there!



-MC (IRC: mercutioviz)





________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Ritesh Singh
Sent: Friday, March 28, 2008 6:08 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] New to FreeSwitch



Hi All,

I am very new to freeswitch. It will be great if some one can tell me
few things:

1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. Then
what changes i am supposed to do and at what place. I would like to use
the mod_portaudio for this purpose.

2) I started the windows freeswitch and loaded port_audio by using "load
mod_portaudio" , then used "pa call 1234"...the session gets initiated
but the call gets hanged up with the log "portaudio/1234 [CS_RING]
[NO_ROUTE_DESTINATION]".

3) Suppose i have a jabber server x.net. and i have 2 users having
account at that jabber server. Then how can i use mod_dingaling so that
i have the voice chat between those user of the jabber server x.net.
Also, i would like to dump the all the record regarding the call made,
like duration of call , person who initiated the call, call destination
, etc.

Any help in this regard is highly appreciable. Please do forgive me if
you think its stupid mail but i am desparate for these answers.

This is my second mail as no one replied my first mail. Please do reply
this mail else i ll be broken ....

Thanks and Regards
Ritesh




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kokoska rokoska
2008-03-29 12:57:23 UTC
Permalink
This post might be inappropriate. Click to display it.
Brian West
2008-03-29 16:15:00 UTC
Permalink
Post by kokoska rokoska
Hi all!
I'm very new to Freeswitch and thus I'm looking for advices/hints for
painless start :-)
Nothing is painless.
Post by kokoska rokoska
I have a lot of experience with Asterisk and OpenSER, but the
philosophy
od Freeswitch differs...
Thats an understatement. ;)
Post by kokoska rokoska
I like to deploy PBX/Switch for a lot of SIP users wich registers with
it and - also - with larg number of SIP gateways the PBX/Switch should
regester with. Users population/gateways/call routing have to be dynamic
(database-driven, like I'm accustomed form Asterisk and OpenSER) with
quite standard features (conditional/unconditional forwarding,
voice-mail, call-waiting, resource limits etc.) and especially with good
over-all performance.
Like i red in docs, dynamic SIP users could by done with mod_xml_curl
directory but I like to ask: Is it "fast enough"?
Direct DB in my opinion is a very bad idea. With xml_curl you can
interface to just about anything and cluster it up and fail over
rather easily with http gets. And no it's NOT slow, that depends on
how fast your web server and db are... trust me it can scream if you
do it correctly.
Post by kokoska rokoska
Couldn't be better direct DB lookups? If yes, how to accomplish that?
BTW: Is there a way how to share "registered" users between
independant
Freeswitch boxes like I do with OpenSER
(single registrar, many proxies)?
./configure --enable-core-odbc
Post by kokoska rokoska
The second thing I'm thinking about is dynamic call-routing rules (aka
dialplan) with a lots of "destination numbers mangling" I have to do.
What is better way - using mod_xml_curl and try to serve exact
"extension" based on db lookups and followed processing or using an
event socket (in outbound mode I think) and completly control call-
flow
in Freeswitch from remote deamon. Or should I look to another
scenario?
use xml_curl.
Post by kokoska rokoska
Like I wrote, now I'm using Asterisk (together with OpenSER) with
realtime users and whole my dialpan looks
exten=> _X.,1,AGI(routing.bin)
exten=> _X.,2,Hangup
where routing.bin is my simple "ANSI C" application doing all I need and
"from time to time" communicating with "underlying" Asterisk :-)
I know I could study in-depth all source code and experiment with
various deployment scenarios, but it is distressful and long, long way
I'm trying to aviod. That is why I ask you, the ones with much deeper
knowledge of Freeswitch, what is the best point to start.
Better get ready to dive in.

http://wiki.freeswitch.org or #freeswitch on irc.freenode.net
Post by kokoska rokoska
Any suggestions, recommendation or hints are very appreciated! :-)
be like Nike and JUST DO IT! ;)

/b

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kokoska rokoska
2008-03-29 17:51:37 UTC
Permalink
Hi Brian,

thank you very much for your answer!
Post by Brian West
Post by kokoska rokoska
Hi all!
I'm very new to Freeswitch and thus I'm looking for advices/hints for
painless start :-)
Nothing is painless.
I see:-) Few years with SIP servers and signaling "nonstandards" cost me
a lot of gray hairs...
Post by Brian West
Post by kokoska rokoska
I have a lot of experience with Asterisk and OpenSER, but the philosophy
od Freeswitch differs...
Thats an understatement. ;)
Post by kokoska rokoska
I like to deploy PBX/Switch for a lot of SIP users wich registers with
it and - also - with larg number of SIP gateways the PBX/Switch should
regester with. Users population/gateways/call routing have to be dynamic
(database-driven, like I'm accustomed form Asterisk and OpenSER) with
quite standard features (conditional/unconditional forwarding,
voice-mail, call-waiting, resource limits etc.) and especially with good
over-all performance.
Like i red in docs, dynamic SIP users could by done with mod_xml_curl
directory but I like to ask: Is it "fast enough"?
Direct DB in my opinion is a very bad idea. With xml_curl you can
interface to just about anything and cluster it up and fail over rather
easily with http gets.
Yes, I catch the main idea behind it :-)
Post by Brian West
And no it's NOT slow, that depends on how fast
your web server and db are... trust me it can scream if you do it correctly.
Now I'm very satisfied with MySQL cluster and the performance it gives
me throught direct queries (tens of thousands of SELECTs per second). I
haven't much exeperience with webservers, but I'm affraid that number of
responses per second will be much lower (DB lookup + webserver overhead).

But - you wrote its fast engough, so I belive it. Thank you for an advice!
Post by Brian West
Post by kokoska rokoska
Couldn't be better direct DB lookups? If yes, how to accomplish that?
BTW: Is there a way how to share "registered" users between independant
Freeswitch boxes like I do with OpenSER
(single registrar, many proxies)?
./configure --enable-core-odbc
Thanks again, will look at it!
Post by Brian West
Post by kokoska rokoska
The second thing I'm thinking about is dynamic call-routing rules (aka
dialplan) with a lots of "destination numbers mangling" I have to do.
What is better way - using mod_xml_curl and try to serve exact
"extension" based on db lookups and followed processing or using an
event socket (in outbound mode I think) and completly control call-flow
in Freeswitch from remote deamon. Or should I look to another scenario?
use xml_curl.
I will follow your recommendation :-)
Post by Brian West
Post by kokoska rokoska
Like I wrote, now I'm using Asterisk (together with OpenSER) with
realtime users and whole my dialpan looks
exten=> _X.,1,AGI(routing.bin)
exten=> _X.,2,Hangup
where routing.bin is my simple "ANSI C" application doing all I need and
"from time to time" communicating with "underlying" Asterisk :-)
I know I could study in-depth all source code and experiment with
various deployment scenarios, but it is distressful and long, long way
I'm trying to aviod. That is why I ask you, the ones with much deeper
knowledge of Freeswitch, what is the best point to start.
Better get ready to dive in.
I know I need to :-)
But, for the first time, its better to use well-proven ways from
experienced users, if they will be so kind and give me some clue :-)
Many thanks for yours!
Post by Brian West
http://wiki.freeswitch.org or #freeswitch on irc.freenode.net
I'm affraid I red all pages I found of Freeswitch wiki, but a lot of
thinks are still unclear. For example info about Sofia "fine-tuning"
(Path support, headers removing/adding/altering, replies modifications
etc.) I could not find...
And conceptual advice (xml_curl contra event socket) I have never saw in
any wiki :-)
Post by Brian West
Post by kokoska rokoska
Any suggestions, recommendation or hints are very appreciated! :-)
be like Nike and JUST DO IT! ;)
I do :-)


Thank you very much for your help!

Best regards,

kokoska.rokoska
Brian West
2008-03-29 18:19:36 UTC
Permalink
Adding headers are easy via application=set data=sip_h_Header=value

What do you mean path support? It doesn't have the power OpenSER has
on total manipulation of sip headers.

/b
Post by kokoska rokoska
I'm affraid I red all pages I found of Freeswitch wiki, but a lot of
thinks are still unclear. For example info about Sofia "fine-tuning"
(Path support, headers removing/adding/altering, replies modifications
etc.) I could not find...
And conceptual advice (xml_curl contra event socket) I have never saw in
any wiki :-)
kokoska rokoska
2008-03-29 20:09:51 UTC
Permalink
Post by Brian West
Adding headers are easy via application=set data=sip_h_Header=value
Thank you very much for that info! Is this work for INVITE only (like in
Asterisk) or is it possible to add headers to the other messages and
especially to the replies?
BTW: Is there a way to send custom reply based on my decision (480
contra 404, 502 contra 404 etc.)?
Post by Brian West
What do you mean path support?
I like to know, if I add "Path" header to REGISTERs on my loadbalancer,
than if Freeswitch succesfuly registeres NATed user with its real IP
address instead of address of loadbalancer. And if Freeswitch will send
NAT keep-alive packets to the NATed user based on Path header too...
Post by Brian West
It doesn't have the power OpenSER has
on total manipulation of sip headers.
I know it, but its not drawback for me. I need only basic option to
manipulate SIP messages and really appreciate that Freeswitch is B2BUA
and not proxy.
And, like I red somewhere, with much more better SIP stack (Sofia) than
Asterisk has. The main things I'm missing in Asterisk is lack of PRACK
(100 REL), not ever sent replies like 181 and 182, never used/sent
Diversion header and last, but not least missing support for
removing/altering headers...


Thanks again, Brian, for your answers and notable help!

Best regards,

kokoska.rokoska
Brian West
2008-03-30 02:36:27 UTC
Permalink
Post by kokoska rokoska
Post by Brian West
Adding headers are easy via application=set data=sip_h_Header=value
Thank you very much for that info! Is this work for INVITE only (like in
Asterisk) or is it possible to add headers to the other messages and
especially to the replies?
BTW: Is there a way to send custom reply based on my decision (480
contra 404, 502 contra 404 etc.)?
Respond application.
Post by kokoska rokoska
Post by Brian West
What do you mean path support?
I like to know, if I add "Path" header to REGISTERs on my
loadbalancer,
than if Freeswitch succesfuly registeres NATed user with its real IP
address instead of address of loadbalancer. And if Freeswitch will send
NAT keep-alive packets to the NATed user based on Path header too...
No clue.
Post by kokoska rokoska
Post by Brian West
It doesn't have the power OpenSER has
on total manipulation of sip headers.
I know it, but its not drawback for me. I need only basic option to
manipulate SIP messages and really appreciate that Freeswitch is B2BUA
and not proxy.
And, like I red somewhere, with much more better SIP stack (Sofia) than
Asterisk has. The main things I'm missing in Asterisk is lack of PRACK
(100 REL), not ever sent replies like 181 and 182, never used/sent
Diversion header and last, but not least missing support for
removing/altering headers...
Thanks again, Brian, for your answers and notable help!
No problem keep asking.

/b
Post by kokoska rokoska
Best regards,
kokoska.rokoska
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
kokoska rokoska
2008-03-30 09:15:30 UTC
Permalink
Post by Brian West
Post by kokoska rokoska
Hi all!
I'm very new to Freeswitch and thus I'm looking for advices/hints for
painless start :-)
Nothing is painless.
Post by kokoska rokoska
I have a lot of experience with Asterisk and OpenSER, but the philosophy
od Freeswitch differs...
Thats an understatement. ;)
Post by kokoska rokoska
I like to deploy PBX/Switch for a lot of SIP users wich registers with
it and - also - with larg number of SIP gateways the PBX/Switch should
regester with. Users population/gateways/call routing have to be dynamic
(database-driven, like I'm accustomed form Asterisk and OpenSER) with
quite standard features (conditional/unconditional forwarding,
voice-mail, call-waiting, resource limits etc.) and especially with good
over-all performance.
Like i red in docs, dynamic SIP users could by done with mod_xml_curl
directory but I like to ask: Is it "fast enough"?
Direct DB in my opinion is a very bad idea. With xml_curl you can
interface to just about anything and cluster it up and fail over rather
easily with http gets. And no it's NOT slow, that depends on how fast
your web server and db are... trust me it can scream if you do it correctly.
This question is a little bit off-topic, but I try it :-)

I have very limited experiences with webservers, so can you recommend me
fast enough webserver and underlying tehnology?

For idea proof-of-concept I can use Apache + mod_php, but for production
use I'm affraid it will be nearly useless because of PHP is interpreted
and thus very slow.
So what is - by your mean - good technology to start with? Java +
Tomcat? Or is there a way how to utilize my "basic" knowlege of "C"
without writting my own webserver? :-)

Have a nice day, best regards,

kokoska.rokoska
Brian West
2008-03-30 15:39:26 UTC
Permalink
Wasn't this what the whole Zend optimizer was for? So it would cache
and pre-compiled scripts?

Similar to how fastcgi does ... Either one is ok. You're over
thinking... You're not going to put it all on ONE single machine but a
web cluster so you can spread it out.

/b
Post by kokoska rokoska
This question is a little bit off-topic, but I try it :-)
I have very limited experiences with webservers, so can you
recommend me
fast enough webserver and underlying tehnology?
For idea proof-of-concept I can use Apache + mod_php, but for
production
use I'm affraid it will be nearly useless because of PHP is
interpreted
and thus very slow.
So what is - by your mean - good technology to start with? Java +
Tomcat? Or is there a way how to utilize my "basic" knowlege of "C"
without writting my own webserver? :-)
Have a nice day, best regards,
kokoska.rokoska
kokoska rokoska
2008-03-31 09:19:16 UTC
Permalink
Post by Brian West
Wasn't this what the whole Zend optimizer was for? So it would cache
and pre-compiled scripts?
Thanks Brian for your answer!

When I started with Asterisk in the past, I wrote all my AGI scripts in
PHP. And than I saw big latency in call setup and lack of performance.
Thus I try Zend optimizer, eAccelerator and APC too. You are right, all
of them help much, but I still wasn't satisfied.
After rewritting all of my code to ANSI C, everything change rapidly.
The same HW could setup nearly twice as much calls per second than with
PHP and bytecode cache...
Post by Brian West
Similar to how fastcgi does ... Either one is ok.
I have never heard about fastcgi. I will look at it. Thank you!
Post by Brian West
You're over thinking...
May be you are right :-)
I prepare test setup and let we see what happens :-)
Post by Brian West
You're not going to put it all on ONE single machine but a
web cluster so you can spread it out.
Yes, I can. But one of the most important reasons for "switch to
Freeswitch" is better performance (call setup rate and concurrent calls)
on existing HW I believe in :-)


Thanks again for useful answers! I setup something in lab and - I'm
affraid - will ask for other unclear things...

Best regards,

kokoska.rokoska
Arnaldo de Moraes Pereira
2008-03-31 11:35:40 UTC
Permalink
On Mon, Mar 31, 2008 at 6:19 AM, kokoska rokoska
Post by kokoska rokoska
Post by Brian West
Wasn't this what the whole Zend optimizer was for? So it would cache
and pre-compiled scripts?
Thanks Brian for your answer!
When I started with Asterisk in the past, I wrote all my AGI scripts in
PHP. And than I saw big latency in call setup and lack of performance.
Thus I try Zend optimizer, eAccelerator and APC too. You are right, all
of them help much, but I still wasn't satisfied.
After rewritting all of my code to ANSI C, everything change rapidly.
The same HW could setup nearly twice as much calls per second than with
PHP and bytecode cache...
Even if you have to write C code for performance, it's easy to write
an application module to handle all needed tasks. FreeSWITCH's
architecture allows one to worry about accomplish the tasks when
writing code, instead of worrying about workarounds to get your code
to work the way you want, without breaking other things... and without
hacks.

Just wanted to say that, because I ended up in the same situation,
having to write asterisk apps. I had a hard time writing them and was
_very_ impressed when started writing FreeSWITCH's. All weird problems
were just gone, the things just work and the API is consistent.

I've just started using mod_python and my guess is that it'll fit your
needs. When you're done testing, please share the results, so we can
try to understand where are the bottlenecks, if any, and improve it.
Post by kokoska rokoska
Post by Brian West
Similar to how fastcgi does ... Either one is ok.
I have never heard about fastcgi. I will look at it. Thank you!
Post by Brian West
You're over thinking...
May be you are right :-)
I prepare test setup and let we see what happens :-)
Post by Brian West
You're not going to put it all on ONE single machine but a
web cluster so you can spread it out.
Yes, I can. But one of the most important reasons for "switch to
Freeswitch" is better performance (call setup rate and concurrent calls)
on existing HW I believe in :-)
Thanks again for useful answers! I setup something in lab and - I'm
affraid - will ask for other unclear things...
Best regards,
kokoska.rokoska
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
kokoska rokoska
2008-03-31 12:07:41 UTC
Permalink
Post by Arnaldo de Moraes Pereira
On Mon, Mar 31, 2008 at 6:19 AM, kokoska rokoska
Post by kokoska rokoska
Post by Brian West
Wasn't this what the whole Zend optimizer was for? So it would cache
and pre-compiled scripts?
Thanks Brian for your answer!
When I started with Asterisk in the past, I wrote all my AGI scripts in
PHP. And than I saw big latency in call setup and lack of performance.
Thus I try Zend optimizer, eAccelerator and APC too. You are right, all
of them help much, but I still wasn't satisfied.
After rewritting all of my code to ANSI C, everything change rapidly.
The same HW could setup nearly twice as much calls per second than with
PHP and bytecode cache...
Even if you have to write C code for performance, it's easy to write
an application module to handle all needed tasks. FreeSWITCH's
architecture allows one to worry about accomplish the tasks when
writing code, instead of worrying about workarounds to get your code
to work the way you want, without breaking other things... and without
hacks.
It sounds very good!
Post by Arnaldo de Moraes Pereira
Just wanted to say that, because I ended up in the same situation,
having to write asterisk apps. I had a hard time writing them and was
_very_ impressed when started writing FreeSWITCH's. All weird problems
were just gone, the things just work and the API is consistent.
Thank you very much, Arnaldo, for that info!

I try to wrote some Asterisk apps (and also slightly change behaviour of
chan_sip) to siut my needs, but it was the pain. I'm not a real
programmer, better to say script writer from compulsion, and from my
point of view chan_sip, logger and cdr... change between versions so
dramatically that I resign to do it cleanly and ended up with mentioned
workaround - call routing I do thru AGI only without assistance of
Asterisk dialplan and call logging is done thru CDR manager...
Post by Arnaldo de Moraes Pereira
I've just started using mod_python and my guess is that it'll fit your
needs.
I know nerly nothing about python (except some chnges to mediaproxy I
need to do by myself), but I'm open to all new things and learn quickly :-)
I will look at it, thank you!
Post by Arnaldo de Moraes Pereira
When you're done testing, please share the results, so we can
try to understand where are the bottlenecks, if any, and improve it.
I do what I can, but I'm affraid I need to spend some time to understand
Freeswitch before I deploy exectly what I need.
But, after all work will be done, I try to share as much info as I can..

Thanks again, Arnaldo, for notable help!

Best regards,

kokoska.rokoska
Mark Crane
2008-03-31 19:32:35 UTC
Permalink
You want a fast webserver...take a look at:
http://www.lighttpd.net/

"lighttpd powers several popular Web 2.0 sites like
YouTube, wikipedia and meebo. Its high speed
io-infrastructure allows them to scale several times
better with the same hardware than with alternative
web-servers." - quote from website.

Mark
Post by kokoska rokoska
On Mar 29, 2008, at 7:57 AM, kokoska rokoska
Post by kokoska rokoska
Hi all!
I'm very new to Freeswitch and thus I'm looking
for advices/hints for
Post by kokoska rokoska
painless start :-)
Nothing is painless.
Post by kokoska rokoska
I have a lot of experience with Asterisk and
OpenSER, but the philosophy
Post by kokoska rokoska
od Freeswitch differs...
Thats an understatement. ;)
Post by kokoska rokoska
I like to deploy PBX/Switch for a lot of SIP
users wich registers with
Post by kokoska rokoska
it and - also - with larg number of SIP gateways
the PBX/Switch should
Post by kokoska rokoska
regester with. Users population/gateways/call
routing have to be dynamic
Post by kokoska rokoska
(database-driven, like I'm accustomed form
Asterisk and OpenSER) with
Post by kokoska rokoska
quite standard features
(conditional/unconditional forwarding,
Post by kokoska rokoska
voice-mail, call-waiting, resource limits etc.)
and especially with good
Post by kokoska rokoska
over-all performance.
Like i red in docs, dynamic SIP users could by
done with mod_xml_curl
Post by kokoska rokoska
directory but I like to ask: Is it "fast enough"?
Direct DB in my opinion is a very bad idea. With
xml_curl you can
interface to just about anything and cluster it up
and fail over rather
easily with http gets. And no it's NOT slow, that
depends on how fast
your web server and db are... trust me it can
scream if you do it correctly.
This question is a little bit off-topic, but I try
it :-)
I have very limited experiences with webservers, so
can you recommend me
fast enough webserver and underlying tehnology?
For idea proof-of-concept I can use Apache +
mod_php, but for production
use I'm affraid it will be nearly useless because of
PHP is interpreted
and thus very slow.
So what is - by your mean - good technology to start
with? Java +
Tomcat? Or is there a way how to utilize my "basic"
knowlege of "C"
without writting my own webserver? :-)
Have a nice day, best regards,
kokoska.rokoska
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Post by kokoska rokoska
http://www.freeswitch.org
____________________________________________________________________________________
Like movies? Here's a limited-time offer: Blockbuster Total Access for one month at no cost.
http://tc.deals.yahoo.com/tc/blockbuster/text4.com
Matt Klein
2008-03-31 20:10:52 UTC
Permalink
lighttpd

Agree 100%.

Low overhead I've seen over apache etc.

m
Post by Mark Crane
http://www.lighttpd.net/
"lighttpd powers several popular Web 2.0 sites like
YouTube, wikipedia and meebo. Its high speed
io-infrastructure allows them to scale several times
better with the same hardware than with alternative
web-servers." - quote from website.
Mark
Post by kokoska rokoska
On Mar 29, 2008, at 7:57 AM, kokoska rokoska
Post by kokoska rokoska
Hi all!
I'm very new to Freeswitch and thus I'm looking
for advices/hints for
Post by kokoska rokoska
painless start :-)
Nothing is painless.
Post by kokoska rokoska
I have a lot of experience with Asterisk and
OpenSER, but the philosophy
Post by kokoska rokoska
od Freeswitch differs...
Thats an understatement. ;)
Post by kokoska rokoska
I like to deploy PBX/Switch for a lot of SIP
users wich registers with
Post by kokoska rokoska
it and - also - with larg number of SIP gateways
the PBX/Switch should
Post by kokoska rokoska
regester with. Users population/gateways/call
routing have to be dynamic
Post by kokoska rokoska
(database-driven, like I'm accustomed form
Asterisk and OpenSER) with
Post by kokoska rokoska
quite standard features
(conditional/unconditional forwarding,
Post by kokoska rokoska
voice-mail, call-waiting, resource limits etc.)
and especially with good
Post by kokoska rokoska
over-all performance.
Like i red in docs, dynamic SIP users could by
done with mod_xml_curl
Post by kokoska rokoska
directory but I like to ask: Is it "fast enough"?
Direct DB in my opinion is a very bad idea. With
xml_curl you can
interface to just about anything and cluster it up
and fail over rather
easily with http gets. And no it's NOT slow, that
depends on how fast
your web server and db are... trust me it can
scream if you do it correctly.
This question is a little bit off-topic, but I try
it :-)
I have very limited experiences with webservers, so
can you recommend me
fast enough webserver and underlying tehnology?
For idea proof-of-concept I can use Apache +
mod_php, but for production
use I'm affraid it will be nearly useless because of
PHP is interpreted
and thus very slow.
So what is - by your mean - good technology to start
with? Java +
Tomcat? Or is there a way how to utilize my "basic"
knowlege of "C"
without writting my own webserver? :-)
Have a nice day, best regards,
kokoska.rokoska
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
Post by kokoska rokoska
http://www.freeswitch.org
____________________________________________________________________________________
Like movies? Here's a limited-time offer: Blockbuster Total Access for one month at no cost.
http://tc.deals.yahoo.com/tc/blockbuster/text4.com
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Jay Fenton
2008-03-31 20:27:37 UTC
Permalink
Post by Mark Crane
lighttpd
Agree 100%.
Low overhead I've seen over apache etc.
Lighttpd's great, second only to Nginx http://nginx.net/

(Nginx benches a little less CPU heavy under load)

--
Regards,

Jay Fenton <jay.fenton at howlertech.com>
CTO, Howler Technologies Ltd
t: +44 (0)207 099 7095
f: +44 (0)207 099 7098

Registered in England & Wales, Company No. 06285634
kokoska rokoska
2008-04-01 08:11:01 UTC
Permalink
Post by Jay Fenton
Post by Mark Crane
lighttpd
Agree 100%.
Low overhead I've seen over apache etc.
Lighttpd's great, second only to Nginx http://nginx.net/
(Nginx benches a little less CPU heavy under load)
Many thanks to everybody for suggestion!
I will look at lighttpd.

Best regards,

kokoska.rokoska
J. Javier Maestro
2008-04-08 19:02:02 UTC
Permalink
Post by Mark Crane
http://www.lighttpd.net/
If anyone wants to look at some real light webserver with some nice coding
under the hood, have a look at Cherokee

http://www.cherokee-project.com/


My 2 cents >:)
--
J. Javier Maestro <jjmaestro at nosys.es>
Socio Consultor - Nosys AJjV S.L.
Daniel Swarbrick
2008-04-08 22:23:28 UTC
Permalink
Post by J. Javier Maestro
Post by Mark Crane
http://www.lighttpd.net/
If anyone wants to look at some real light webserver with some nice coding
under the hood, have a look at Cherokee
http://www.cherokee-project.com/
My 2 cents >:)
This is getting a little OT now, but I guess we should throw nginx
(http://nginx.net/) into the mix. Supposedly faster and less leaky than
lighty.

Sometimes what is more important is low memory footprint - for those
with cheap/small VPSes or wanting to embed a webserver in some low end
embedded Linux box (along with FS of course...)
kokoska rokoska
2008-04-09 05:15:42 UTC
Permalink
Post by Daniel Swarbrick
Post by J. Javier Maestro
Post by Mark Crane
http://www.lighttpd.net/
If anyone wants to look at some real light webserver with some nice coding
under the hood, have a look at Cherokee
http://www.cherokee-project.com/
My 2 cents >:)
This is getting a little OT now, but I guess we should throw nginx
(http://nginx.net/) into the mix. Supposedly faster and less leaky than
lighty.
Sometimes what is more important is low memory footprint - for those
with cheap/small VPSes or wanting to embed a webserver in some low end
embedded Linux box (along with FS of course...)
Thank you very much, Daniel, for your tip!
I will look at it.

BTW: For me is important number of call setups per second and number of
concurent call on 2 to 16 core computers...

Best regards,

kokoska.rokoska

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