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
On Mar 28, 2008, at 8:07 AM, Ritesh Singh wrote:
Hi All,
I am very new to freeswitch. It will be great if some one can tell me few things:
Welcome to the community.
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.
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.
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
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ritesh Singh Sent: Friday, March 28, 2008 6:08 AM To: freeswitch-users@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
Hi all! I'm very new to Freeswitch and thus I'm looking for advices/hints for painless start :-) I have a lot of experience with Asterisk and OpenSER, but the philosophy od Freeswitch differs... What is going on: 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"? 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)? 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? Like I wrote, now I'm using Asterisk (together with OpenSER) with realtime users and whole my dialpan looks like: 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. Any suggestions, recommendation or hints are very appreciated! :-) Best regards, kokoska.rokoska
On Mar 29, 2008, at 7:57 AM, kokoska rokoska wrote:
Hi all!
I'm very new to Freeswitch and thus I'm looking for advices/hints for painless start :-)
Nothing is painless.
I have a lot of experience with Asterisk and OpenSER, but the philosophy od Freeswitch differs...
Thats an understatement. ;)
What is going on: 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.
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
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.
Like I wrote, now I'm using Asterisk (together with OpenSER) with realtime users and whole my dialpan looks like: 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
Any suggestions, recommendation or hints are very appreciated! :-)
be like Nike and JUST DO IT! ;) /b
Hi Brian, thank you very much for your answer! Brian West napsal(a):
On Mar 29, 2008, at 7:57 AM, kokoska rokoska wrote:
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...
I have a lot of experience with Asterisk and OpenSER, but the philosophy od Freeswitch differs...
Thats an understatement. ;)
What is going on: 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 :-)
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!
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!
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 :-)
Like I wrote, now I'm using Asterisk (together with OpenSER) with realtime users and whole my dialpan looks like: 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!
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 :-)
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
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 On Mar 29, 2008, at 12:51 PM, kokoska rokoska wrote:
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 :-)
Brian West napsal(a):
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.)?
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...
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
On Mar 29, 2008, at 3:09 PM, kokoska rokoska wrote:
Brian West napsal(a):
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.
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.
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
Best regards,
kokoska.rokoska
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West napsal(a):
On Mar 29, 2008, at 7:57 AM, kokoska rokoska wrote:
Hi all!
I'm very new to Freeswitch and thus I'm looking for advices/hints for painless start :-)
Nothing is painless.
I have a lot of experience with Asterisk and OpenSER, but the philosophy od Freeswitch differs...
Thats an understatement. ;)
What is going on: 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
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 On Mar 30, 2008, at 4:15 AM, kokoska rokoska wrote:
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 napsal(a):
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...
Similar to how fastcgi does ... Either one is ok. I have never heard about fastcgi. I will look at it. Thank you!
You're over thinking...
May be you are right :-) I prepare test setup and let we see what happens :-)
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
On Mon, Mar 31, 2008 at 6:19 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Brian West napsal(a):
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.
Similar to how fastcgi does ... Either one is ok. I have never heard about fastcgi. I will look at it. Thank you!
You're over thinking...
May be you are right :-) I prepare test setup and let we see what happens :-)
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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Arnaldo de Moraes Pereira napsal(a):
On Mon, Mar 31, 2008 at 6:19 AM, kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Brian West napsal(a):
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!
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...
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!
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
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 --- kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Brian West napsal(a):
On Mar 29, 2008, at 7:57 AM, kokoska rokoska
wrote:
Hi all!
I'm very new to Freeswitch and thus I'm looking
for advices/hints for
painless start :-)
Nothing is painless.
I have a lot of experience with Asterisk and
OpenSER, but the philosophy
od Freeswitch differs...
Thats an understatement. ;)
What is going on: 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
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
____________________________________________________________________________________ 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
lighttpd Agree 100%. Low overhead I've seen over apache etc. m On Mon, 31 Mar 2008, Mark Crane wrote:
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
--- kokoska rokoska <kokoska.rokoska@post.cz> wrote:
Brian West napsal(a):
On Mar 29, 2008, at 7:57 AM, kokoska rokoska
wrote:
Hi all!
I'm very new to Freeswitch and thus I'm looking
for advices/hints for
painless start :-)
Nothing is painless.
I have a lot of experience with Asterisk and
OpenSER, but the philosophy
od Freeswitch differs...
Thats an understatement. ;)
What is going on: 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
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
____________________________________________________________________________________ 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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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
Jay Fenton napsal(a):
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
On Mar Mon 31 2008 12:32, Mark Crane wrote:
You want a fast webserver...take a look at: 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@nosys.es> Socio Consultor - Nosys AJjV S.L.
On Tue, 2008-04-08 at 21:02 +0200, J. Javier Maestro wrote:
On Mar Mon 31 2008 12:32, Mark Crane wrote:
You want a fast webserver...take a look at: 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...)
Daniel Swarbrick napsal(a):
On Tue, 2008-04-08 at 21:02 +0200, J. Javier Maestro wrote:
On Mar Mon 31 2008 12:32, Mark Crane wrote:
You want a fast webserver...take a look at: 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
participants (10)
-
Arnaldo de Moraes Pereira
-
Brian West
-
Daniel Swarbrick
-
J. Javier Maestro
-
Jay Fenton
-
kokoska rokoska
-
Mark Crane
-
Matt Klein
-
Michael Collins
-
Ritesh Singh