Re: [Freeswitch-users] Suggestion for xml_curl performance

[shehzad p]

Hi Brian,

My setup is to use FS as basic calls routing.
1. Calls are coming to FS from more than one customer Gateways, and I need
to authenticate them and check for enough balance based on database,
   [Caller Gateways] ===> [FreeSWITCH] ===>
[Provider Gateways]
2. After knowing that Caller Gateways is valid, then based on dialed number
it search in database for Provider Gateway and bridge the call there.
3. After call finish CDR is inserted back into database.

My old setup was using Javascript which works fine in traffic of 10 to 20
calls, but then increase of traffic causes many problems.

Now I eliminate use of any of the script (javascript or any other) for call
routing, and route calls directly from dialplan, 
So I have setup test system using xml-curl to generate dynamic dialplan, 
I used below xml_curl PHP example as reference:
http://wiki.freeswitch.org/wiki/Mod_xml_curl_PHP_example
For CDR processing I used xml_cdr, with help of the example in FS source
:scripts/contrib/trixter/xml-cdr.


Waiting for any better suggestions, any comments...

thanks
msp.

Brian West-3 wrote:
> 
> it all depends on what you're doing.. can you elaborate?
> 
> /b
> 
> On Feb 20, 2009, at 4:18 AM, shehzad p wrote:
> 
>> Recently I faced some performance bottleneck by using Javascript.
> 
> 
> ___
> 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
> 
> 

-- 
View this message in context: 
http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22133185.html
Sent from the Freeswitch-users mailing list archive at Nabble.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

Re: [Freeswitch-users] mod_erlang_event compile problem

[Leon de Rooij]

Hi Andrew,

Thanks! it compiles fine now.. Also thanks for the tip about empd, got  
it running without errors now :-)

regards,

Leon


On Feb 20, 2009, at 8:08 PM, Andrew Thompson wrote:

> On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote:
>> Hi,
>>
>> I wanted to try out the mod_erlang_event module. I have Erlang R12B5
>> compiled and it's in the same location as the Makefile specifies (/ 
>> usr/
>> local/lib/erlang/...), but running make in the src/mod/ 
>> event_handlers/
>> mod_erlang_event goes wrong:
>>
>
> Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so  
> it
> didn't show up for me. Thanks to MikeJ for the fix suggestion.
>
>> Also, after this, FS goes haywire after loading the module
>> and spews out these messages continuously:
>>
>
> You don't have the erlang port mapper daemon running (epmd).
> mod_erlang_event needs it to be running in order to be able to  
> register
> itself as an erlang node. On your system; epmd isn't in $PATH so my
> system() call that tries to start it fails. I've made the module init
> system fail properly instead of looping indefinitely as well as  
> print a
> slightly more helpful error message now. Let me know if you have any
> better luck :) The fix is in-tree as of r12192.
>
> Thanks again for the bug report.
>
> Andrew
>
> ___
> 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


___
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

Re: [Freeswitch-users] Passing Caller_ID to MOD_LCR

[Rupa Schomaker (lists)]

> "default" is a reserved profile name -- I should probably prevent that
> from loading.

correction, default is not reserved...

> Regarding passing the callerid to the custom sql, let me see what I can
> come up with...

You can now specify channel variables in your custom sql.  So, you
should be able to pass CID or a subset of CID to your custom sql.  I've
updated the wiki with info on this.

beware: channel vars only work when called in the context of a session.
 Using a profile that uses a custom sql with channel variables from the
commandline will result in an error.

-Rupa

___
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

Re: [Freeswitch-users] [ANN] Spice Telephony - an open source FreeSWITCH/Erlang callcenter platform

[Andrew Thompson]

On Sat, Feb 14, 2009 at 03:04:01PM -0800, JCATS wrote:
> 
> Have you planned any predictive dialer features ( like VICIDIAL )?
>

As Ken Rice mentioned, this isn't really the focus of the project - it's
more for inbound and directed outbound (calling campaigns to specific
people/businesses - not everyone in the phonebook). Primary focus is
inbound (multi brand, skill based routing, dynamic wrapup times, etc).

Expect a new release sometime soonish that actually does something
useful (accepts and routes inbound calls from FreeSWITCH to an agent).
Also; public source control. There's just some additional corporate
nonsense that I have to sort out (again) before that can go live.

Andrew

___
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

Re: [Freeswitch-users] mod_erlang_event compile problem

[Andrew Thompson]

On Fri, Feb 20, 2009 at 05:19:25PM +0100, Leon de Rooij wrote:
> Hi,
> 
> I wanted to try out the mod_erlang_event module. I have Erlang R12B5  
> compiled and it's in the same location as the Makefile specifies (/usr/ 
> local/lib/erlang/...), but running make in the src/mod/event_handlers/ 
> mod_erlang_event goes wrong:
> 

Yeah, this was a gcc4 thing, I've done most of my testing on gcc3 so it
didn't show up for me. Thanks to MikeJ for the fix suggestion.

> Also, after this, FS goes haywire after loading the module  
> and spews out these messages continuously:
>

You don't have the erlang port mapper daemon running (epmd).
mod_erlang_event needs it to be running in order to be able to register
itself as an erlang node. On your system; epmd isn't in $PATH so my
system() call that tries to start it fails. I've made the module init
system fail properly instead of looping indefinitely as well as print a
slightly more helpful error message now. Let me know if you have any
better luck :) The fix is in-tree as of r12192.

Thanks again for the bug report.

Andrew

___
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

Re: [Freeswitch-users] js session.streamFile() interrupt slowness

[Michael Collins]

On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby  wrote:
> I have a few scripts that use the javascript
> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function
> that returns false to interrupt the streaming file. There is a short
> delay between the time when I press a key and the time the file stops
> playing. Is there anything I can adjust that would affect that? It's
> only maybe 2-3 seconds, but it "feels" too long to me.
>
> --Stephen

Could you pastebin your entire script plus the relevant dialplan
entry? Also, could you tell us which operating system and FS revision?
-MC

___
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

[Freeswitch-users] mod_erlang_event compile problem

[Leon de Rooij]

Hi,

I wanted to try out the mod_erlang_event module. I have Erlang R12B5  
compiled and it's in the same location as the Makefile specifies (/usr/ 
local/lib/erlang/...), but running make in the src/mod/event_handlers/ 
mod_erlang_event goes wrong:

Compiling handle_msg.c...
cc1: warnings being treated as errors
handle_msg.c: In function 'handle_msg_sendmsg':
handle_msg.c:429: warning: the address of 'uuid' will always evaluate  
as 'true'
handle_msg.c: In function 'handle_msg_handlecall':
handle_msg.c:541: warning: the address of 'uuid_str' will always  
evaluate as 'true'
make[1]: *** [handle_msg.o] Error 1
make: *** [all] Error 1

At line 429 in handle_msg.c it says:

if (!switch_strlen_zero(uuid) && (session =  
switch_core_session_locate(uuid))) {

(at line 541 is the same problem)

Is this a bug ? I tried removing the first part "! 
switch_strlen_zero(uuid) &&" after which it compiles fine, but since I  
don't fully understand what's going on, I'm sure this is not the  
solution.. Also, after this, FS goes haywire after loading the module  
and spews out these messages continuously:

2009-02-20 14:15:48 [ERR] mod_erlang_event.c:1417  
mod_erlang_event_runtime() Failed to start empd manually
2009-02-20 14:15:48 [DEBUG] mod_erlang_event.c:1401  
mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031
2009-02-20 14:15:48 [WARNING] mod_erlang_event.c:1415  
mod_erlang_event_runtime() Failed to publish port to empd, trying to  
start empd manually
2009-02-20 14:15:48 [ERR] mod_erlang_event.c:1417  
mod_erlang_event_runtime() Failed to start empd manually
2009-02-20 14:15:48 [DEBUG] mod_erlang_event.c:1401  
mod_erlang_event_runtime() Socket up listening on 127.0.0.1:8031
2009-02-20 14:15:48 [WARNING] mod_erlang_event.c:1415  
mod_erlang_event_runtime() Failed to publish port to empd, trying to  
start empd manually
etc..

Can someone help me and point out what's wrong ?

thanks & kind regards,

Leon de Rooij


___
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

Re: [Freeswitch-users] SIP dump to DB

[kokoska rokoska]

jonathan augenstine napsal(a):
> You can tcpdump and then use wireshark to graph the calls.  When the
> dump is displayed in wireshark, select 'Statistics' -> VoIP Calls.  You
> will see a display of all VoIP calls.  Select the one you want graphed,
> or select them all and you will see REINVITE and REFER interaction as
> well as RTP streams.
> 

Thank you very much, jonathan, for your interest!

I use ngrep+wireshark many times a day, but I'm affraid it is not
suitable for that amount of data.

Even with few hundreds MiBs of pcap file wireshark becoms very slow and
I can't imagine how to load 50-100 GiB file with milions of calls and
try to search for one of them :-)

And, even worse, I should "rotate" the file and, don't end with call
divided to multiple files...


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

Re: [Freeswitch-users] SIP dump to DB

[kokoska rokoska]

Shelby Ramsey napsal(a):
> Sorry .. I didn't give enough detail.  My point was to dump it via NGREP
> ... parse it using something else to get it into a database where it
> would be usable.

This is good point! Thank you very much, Shelby!

> Then you can match calls from the CDR (using the UUID)
> to the database.  

This is exactly what I try to accomplish :-)

> The benefit is that you don't have to put the burden
> on your FS boxes to do it ... Just monitor from another device and then
> dump it into the database. 

I think of 2 possibilities:
1. On router replicate whole traffic to dedicated machine and process
dump there (but it probably kills that machine, because it gets all RTP
traffic)
2. Modify FreeSWITCH/Sofia (I have no idea how hard it will be, or if it
is even possible) to duplicate all SIP messages to given URI - main
benefit of this scenario is that I have only SIP messages on logging
machine and that I can use SERlike proxy to parse messages and store
them to DB.

> Of course you better have a beast of
> database if you want to do 10,000 writes per second :) or be running
> something like NDB that scales well.
> 

NDB is overkill for logging :-)
What I think of is some kind of "caching" on FreeSWITCH side. I.e. store
mesasges to DB in bigger chukns (whole call etc.) It will significatnly
reduce DB utilization...

> There are other tools like scapy as well that can be quite useful in fact.
> 
I will look at it, thank you Shelby!

Best regards,

kokoska.roksoka


___
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

Re: [Freeswitch-users] SIP dump to DB

[kokoska rokoska]

Anthony Minessale napsal(a):
> you could try sippcapdump but i hear it needs work but it snoops the
> wire and tries to make individual files out of each call.
> 
> 

Thank you very much, Anthony, for the suggestion!

I will look at it.

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

Re: [Freeswitch-users] SIP dump to DB

[jonathan augenstine]

You can tcpdump and then use wireshark to graph the calls.  When the dump is
displayed in wireshark, select 'Statistics' -> VoIP Calls.  You will see a
display of all VoIP calls.  Select the one you want graphed, or select them
all and you will see REINVITE and REFER interaction as well as RTP streams.

Jonathan


On Fri, Feb 20, 2009 at 7:33 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> you could try sippcapdump but i hear it needs work but it snoops the wire
> and tries to make individual files out of each call.
>
>
>
> On Fri, Feb 20, 2009 at 9:27 AM, kokoska rokoska 
> wrote:
>
>>
>>
>>
>> Shelby Ramsey napsal(a):
>> > Why not just use NGREP and then dump the packets at a more reasonable
>> > pace?
>>
>> Thank you very much, Shelby, for your interest!
>>
>> I can't use ngrep because the dump is not searchable IMO :-)
>> See below, please.
>>
>>  You aren't going to be able to analysis in real time anyway.
>> >
>>
>> Some basic diagnosis should by doable realtime. For example:
>> 1. find whole SIP trace for selected call (both a-leg, b-leg)
>> 2. find all succeful user REGISTERs for given period
>> ...
>>
>> And I have no idea how to do it on 50-100 GiB plain text file.
>>
>> 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
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org 
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
>
> ___
> 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
>
>
___
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

Re: [Freeswitch-users] SIP dump to DB

[Shelby Ramsey]

Sorry .. I didn't give enough detail.  My point was to dump it via NGREP ...
parse it using something else to get it into a database where it would be
usable.  Then you can match calls from the CDR (using the UUID) to the
database.  The benefit is that you don't have to put the burden on your FS
boxes to do it ... Just monitor from another device and then dump it into
the database.  Of course you better have a beast of database if you want to
do 10,000 writes per second :) or be running something like NDB that scales
well.

There are other tools like scapy as well that can be quite useful in fact.

Shelby
___
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

Re: [Freeswitch-users] SIP dump to DB

[Anthony Minessale]

you could try sippcapdump but i hear it needs work but it snoops the wire
and tries to make individual files out of each call.


On Fri, Feb 20, 2009 at 9:27 AM, kokoska rokoska wrote:

>
>
>
> Shelby Ramsey napsal(a):
> > Why not just use NGREP and then dump the packets at a more reasonable
> > pace?
>
> Thank you very much, Shelby, for your interest!
>
> I can't use ngrep because the dump is not searchable IMO :-)
> See below, please.
>
>  You aren't going to be able to analysis in real time anyway.
> >
>
> Some basic diagnosis should by doable realtime. For example:
> 1. find whole SIP trace for selected call (both a-leg, b-leg)
> 2. find all succeful user REGISTERs for given period
> ...
>
> And I have no idea how to do it on 50-100 GiB plain text file.
>
> 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
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
___
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

Re: [Freeswitch-users] SIP dump to DB

[kokoska rokoska]

Shelby Ramsey napsal(a):
> Why not just use NGREP and then dump the packets at a more reasonable
> pace? 

Thank you very much, Shelby, for your interest!

I can't use ngrep because the dump is not searchable IMO :-)
See below, please.

 You aren't going to be able to analysis in real time anyway.
> 

Some basic diagnosis should by doable realtime. For example:
1. find whole SIP trace for selected call (both a-leg, b-leg)
2. find all succeful user REGISTERs for given period
...

And I have no idea how to do it on 50-100 GiB plain text file.

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

Re: [Freeswitch-users] SIP dump to DB

[Shelby Ramsey]

Why not just use NGREP and then dump the packets at a more reasonable pace?
You aren't going to be able to analysis in real time anyway.
___
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

Re: [Freeswitch-users] SIP dump to DB

[kokoska rokoska]

Adam Long napsal(a):
> MySQL MEMORY/HEAP table might be ideal for this. This data is prob not
> critical and is probably
> being used for diagnosing peer connectivity issues anyway.
> If it is critical well then... there are always trade offs right :)
> 

Thank you very much, Adam, for interest!

You are right - it is for diagnosis. And while the data is not critical,
I can't leave them in MySQL MEMORY table. Based on my counts I have to
store and maintain about 70-90 GiB of SIP messages and I simply don't
have enough RAM :-)

> I think in general what he is speaking of is just some sort of temporary SIP
> trace setup that can log
> that can be controlled or filtered. So its not just all or nothing.
> 

Exactly! :-)

> I wonder is it possible to enable the current sip trace functionality via a
> variable. 
> For example something like this.
> 
>  data="{sip_trace=on}sofia/public/xxx...@10.10.10.1" />
> This of course only helps for bridged B2BUA calls.
> 

It will be very helpful...

> But if inbound sip tracing is required a user param like sip_trace could
> address that, yes?
> 

Yes, I need all SIP packets except nat-keep-alives...
BTW: How to recognize them? I know how could I do it on proxy (SER
like), but what about on FreeSWITCH?
Anyway - for me it shouldn't be an issue, because I filter then on
loadbalancer.

> That could be a good starting point, then perhaps I could help roll a
> mod_xml_siptrace module
> based on the mod_xml_cdr design/concept that could somehow link into the
> existing sip trace logging.
> 

It will be very powerfull, but I'm affraid it can't scale to thousands
request per second.

> Just some thoughts, no idea how much of this exists today.
> 

Thanks once more, Adam!

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

Re: [Freeswitch-users] SIP dump to DB

[kokoska rokoska]

Anthony Minessale napsal(a):
> what exact info do you need? That's likely to be a challenge with any
> database to store at that speed.
> 

Thank you very much, Anthony, for your reply!

I should say: Personally I don't need it (I see preformance penalty),
but few people around me need to store somewhere ALL sip messages going
through the server. And ALL means "really all" (well, it will be very
helpful if I can skip OPTIONS and other nat-keep-alive related messages).
So I need something like "sipgrep dump" but I should be able to simply
corelate messages to user (in case of MESSAGE, REGISTER etc.) and to
user+call (in case of INVITE, BYE, CANCEL etc.).

-
BTW:
I'm sure it will be challenge for DB - and thus I made some tests with
mentioned siptrace.
One SIP call is about 18-21 SIP messages, so for 100 cps I should make a
little bit over 2.000 INSERTs per second. 900 REGISTERs per second will
generate about 5.500 INSERTs per second => I need to fire cca 8.000
INSERTs per second...
-

But, may be, better solution exists - without DB.
Any hint is very appreciated :-)

Best regards,

kokoska.rokoska


> On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska
> mailto:kokoska.roko...@post.cz>> wrote:
> 
> 
> Hi all,
> 
> I'm facing the problem I need all SIP messages "going thru" FreeSWITCH
> (I know FS i B2BUA - so, better to say just "all SIP messages") logged
> somewhere and this log have to be "searchable" (by call-id etc) and I
> should be able to simply delete "old" messages...
> And more over - it should be done on not trivial SIP messages amount -
> say hundreds messages per second.
> 
> My questions are:
> 1. Do you have any suggestion how to do it with FreeSWITCH?
> 2. Or it is not possible now, and "bounty" is necessary? :-)
> 3. How hard it will be to implement?
> 
> --
> FYI:
> I think something like SER/Kamailio/OpenSIPS siptrace is what I'am
> (probably) looking for:
> http://www.kamailio.org/docs/modules/1.4.x/siptrace.html
> It could go (based on my tests) up to 7-10.000 messages per second till
> MySQL "dies"...
> --
> 
> Thanks for your time :-)
> 
> 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
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> 
> IRC: irc.freenode.net  #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> 
> iax:gu...@conference.freeswitch.org/888
> 
> googletalk:conf+...@conference.freeswitch.org
> 
> pstn:213-799-1400
> 
> 
> 
> 
> ___
> 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


___
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

Re: [Freeswitch-users] SIP dump to DB

[Adam Long]

MySQL MEMORY/HEAP table might be ideal for this. This data is prob not
critical and is probably
being used for diagnosing peer connectivity issues anyway.
If it is critical well then... there are always trade offs right :)

I think in general what he is speaking of is just some sort of temporary SIP
trace setup that can log
that can be controlled or filtered. So its not just all or nothing.

I wonder is it possible to enable the current sip trace functionality via a
variable. 
For example something like this.


This of course only helps for bridged B2BUA calls.

But if inbound sip tracing is required a user param like sip_trace could
address that, yes?

That could be a good starting point, then perhaps I could help roll a
mod_xml_siptrace module
based on the mod_xml_cdr design/concept that could somehow link into the
existing sip trace logging.

Just some thoughts, no idea how much of this exists today.

Regards,
-Adam



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, February 20, 2009 8:55 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SIP dump to DB

what exact info do you need? That's likely to be a challenge with any
database to store at that speed.
On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska 
wrote:

Hi all,

I'm facing the problem I need all SIP messages "going thru" FreeSWITCH
(I know FS i B2BUA - so, better to say just "all SIP messages") logged
somewhere and this log have to be "searchable" (by call-id etc) and I
should be able to simply delete "old" messages...
And more over - it should be done on not trivial SIP messages amount -
say hundreds messages per second.

My questions are:
1. Do you have any suggestion how to do it with FreeSWITCH?
2. Or it is not possible now, and "bounty" is necessary? :-)
3. How hard it will be to implement?

--
FYI:
I think something like SER/Kamailio/OpenSIPS siptrace is what I'am
(probably) looking for:
http://www.kamailio.org/docs/modules/1.4.x/siptrace.html
It could go (based on my tests) up to 7-10.000 messages per second till
MySQL "dies"...
--

Thanks for your time :-)

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



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400


___
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

Re: [Freeswitch-users] Suggestion for xml_curl performance

[Brian West]

it all depends on what you're doing.. can you elaborate?

/b

On Feb 20, 2009, at 4:18 AM, shehzad p wrote:

> Recently I faced some performance bottleneck by using Javascript.


___
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

Re: [Freeswitch-users] Calling an IPv6 host with a SIP URI from portAudio

[Anthony Minessale]

another way would be to make the original condition have no actions then
use another condition under that with an ip6 specific regex and use action
and anti-action to differentiate



On Fri, Feb 20, 2009 at 12:22 AM, Michael S Collins wrote:

>
> On Feb 19, 2009, at 7:53 PM, Jason White  wrote:
>
> > I have it working now. The relevant changes were as follows.
> >
> >  
> >  
> >  
> >  
> >
> >
>
> Jason,
>
> I like this approach. Good use of the many Dialplan tools.
>
> -MC
>
> ___
> 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
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
___
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

Re: [Freeswitch-users] SIP dump to DB

[Anthony Minessale]

what exact info do you need? That's likely to be a challenge with any
database to store at that speed.

On Fri, Feb 20, 2009 at 5:24 AM, kokoska rokoska wrote:

>
> Hi all,
>
> I'm facing the problem I need all SIP messages "going thru" FreeSWITCH
> (I know FS i B2BUA - so, better to say just "all SIP messages") logged
> somewhere and this log have to be "searchable" (by call-id etc) and I
> should be able to simply delete "old" messages...
> And more over - it should be done on not trivial SIP messages amount -
> say hundreds messages per second.
>
> My questions are:
> 1. Do you have any suggestion how to do it with FreeSWITCH?
> 2. Or it is not possible now, and "bounty" is necessary? :-)
> 3. How hard it will be to implement?
>
> --
> FYI:
> I think something like SER/Kamailio/OpenSIPS siptrace is what I'am
> (probably) looking for:
> http://www.kamailio.org/docs/modules/1.4.x/siptrace.html
> It could go (based on my tests) up to 7-10.000 messages per second till
> MySQL "dies"...
> --
>
> Thanks for your time :-)
>
> 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
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org 
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
___
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

[Freeswitch-users] SIP dump to DB

[kokoska rokoska]

Hi all,

I'm facing the problem I need all SIP messages "going thru" FreeSWITCH
(I know FS i B2BUA - so, better to say just "all SIP messages") logged
somewhere and this log have to be "searchable" (by call-id etc) and I
should be able to simply delete "old" messages...
And more over - it should be done on not trivial SIP messages amount -
say hundreds messages per second.

My questions are:
1. Do you have any suggestion how to do it with FreeSWITCH?
2. Or it is not possible now, and "bounty" is necessary? :-)
3. How hard it will be to implement?

--
FYI:
I think something like SER/Kamailio/OpenSIPS siptrace is what I'am
(probably) looking for:
http://www.kamailio.org/docs/modules/1.4.x/siptrace.html
It could go (based on my tests) up to 7-10.000 messages per second till
MySQL "dies"...
--

Thanks for your time :-)

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

Re: [Freeswitch-users] Suggestion for xml_curl performance

[shehzad p]

My setup of the system is like:
When calls come from Originator gateway I route the call to Terminator
Gateway based on database lookup.
FS works as switching platform.

In previous setup using JavaScript, JavaScript caused the performance
bottleneck when call traffic increases.
Now I am testing xml_curl so asking for any suggestion, if some one has
experienced...



shehzad p wrote:
> 
> Hi all,
> 
> Recently I faced some performance bottleneck by using Javascript.
> 
> Now I am testing xml_curl for next setup, 
> In terms of performance and stability will any body give me some
> information, what are the pros & cons of using xml_curl, what is
> precaution for using it, or any other recommendations...
> 
> Thanks,
> msp
> 

-- 
View this message in context: 
http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22118614.html
Sent from the Freeswitch-users mailing list archive at Nabble.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

[Freeswitch-users] Suggestion for xml_curl performance

[shehzad p]

Hi all,

Recently I faced some performance bottleneck by using Javascript.

Now I am testing xml_curl for next setup, 
In terms of performance and stability will any body give me some
information, what are the pros & cons of using xml_curl, what is precaution
for using it, or any other recommendations...

Thanks,
msp
-- 
View this message in context: 
http://www.nabble.com/Suggestion-for-xml_curl-performance-tp22118122p22118122.html
Sent from the Freeswitch-users mailing list archive at Nabble.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