from:"Giovanni Maruzzelli"

On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:

 can any know how to implement fail over with freeswitch, please help me


This issue has been debated many many times in the mailing lists.
(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
 Very nice :)

 On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com wrote:

 Folks;
 Here's something that I did playing around w/ learning Apple Motion.

Me too: very nice!

-gmaruzz

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Fail over

On Mon, Oct 5, 2009 at 8:59 AM, Giovanni Maruzzelli gmar...@celliax.org wrote:
 On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
 srinivas.ksvre...@gmail.com wrote:

 can any know how to implement fail over with freeswitch, please help me


 This issue has been debated many many times in the mailing lists.
 (hint: no live call failover, HA with OpenSERet similia as load-balancers).

 Please have a look at the archives:

 http://lists.freeswitch.org/pipermail/freeswitch-dev/

 http://lists.freeswitch.org/pipermail/freeswitch-users/


On Mon, Oct 5, 2009 at 9:15 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
 Hi Giovanni Maruzzelli

 Thanks for your reply,
 i am new to is there any way to do live call failover.

Srinivas, are you joking ? Please take the time to read the answer,
when you ask a question.

In my previous mail, I have replied to you:

This issue has been debated many many times in the mailing lists.

(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

The Revenge of the Sip

On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks jaybi...@gmail.com wrote:
 Haha classic !!!
 Can't wait for the next installment in the series !!

 J


 On 06/10/2009, at 1:02, Anthony Minessale anthony.miness...@gmail.com
 wrote:

 neat,

 Here's some suggestions for your next ones. =p

 Have them standing around the hologram trying to destroy the Death
 Star(tm) that happens to look a lot like a giant 3d unix '*' character.
 Then have one rebel say, wait!, why are we wasting our time... watch
 this... and dial a number on his cellphone as the whole thing explodes in
 the background.

 Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you have
 fashioned your own TDM card vroom..
 Join me and together we can make linked lists and monolithic processes,
 NEVER!... vroom vroom Master Coda has taught you well.You are no
 match for me...JOIN THE ORANGE SIDE OF THE FORCE





 On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
  Very nice :)
 
  On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com
  wrote:
 
  Folks;
  Here's something that I did playing around w/ learning Apple Motion.

 Me too: very nice!

 -gmaruzz

 --
 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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 --
 Anthony Minessale II

 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

 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

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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!

2009-09-25 Thread Giovanni Maruzzelli

as always, you can call skype the skypeuser skypiax5, then press 1


On Fri, Sep 25, 2009 at 6:15 PM, Michael Collins m...@freeswitch.org wrote:
 Come on in!
 sip:8...@conference.freeswitch.org or via the good old PSTN at
 +1-213-799-1400

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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects

2009-09-17 Thread Giovanni Maruzzelli

I'm gmaruzz on IRC, for GSM, Skype, Italian language, audio stuff, etc...

-giovanni


On Thu, Sep 17, 2009 at 10:21 AM, Tristan Mahé t.m...@telemaque.fr wrote:
 Hi Michael,

 I'm gled on IRC, always connected so ping me if you wanna talk ;)

 Michael Collins a écrit :

 On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé t.m...@telemaque.fr wrote:

 Hi,

 Count on me for answering questions on IRC when I'm in, and for
 subprojects I'm in too as you know ;)

 Merci!

 Okay, what's your IRC nick and when are you generally on line? Also, I'm
 pretty sure that you're fluent in French which is good because we need more
 multilingual people out there. Last question: what are your areas of
 expertise? I'd like to keep a list of people and what they're good at so we
 know whom to ask first when questions come up.

 Thanks again!
 -MC

 
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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

2009-09-17 Thread Giovanni Maruzzelli

On Fri, Sep 18, 2009 at 12:08 AM, Michael Giagnocavo m...@giagnocavo.net 
wrote:
 Dispose is a .NET only thing. But I think you are right – with anthm’s
 changes, any way you kill your session, if you’re on the right thread, it
 should really hangup.


Problem is, we are trying to *not answer* the incoming call, get the
callid from the ring, destroy the session, create another session (on
the same, monoline interface), and make an outbound call.

Javascript (last svn) give us a

2009-09-18 01:18:49.291721 [ERR] inline:1  Session is not answered!

if we try to session.hangup() a session that was not answered (by the
way, it makes sense).

-giovanni




-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Friday Meeting at 11AM CST

2009-09-11 Thread Giovanni Maruzzelli

On Fri, Sep 11, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote:
 http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11

 Here is the agenda please review and add to it anything you think we
 should cover.

This time too, you all can follow the conference calling Skype the
skypeuser skypiax5, then press 1 on the Skype dialpad (max 20
concurrent users).


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In!

2009-09-11 Thread Giovanni Maruzzelli

On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins m...@freeswitch.org wrote:
 FYI, the conference is starting. Please join us!
 sip:8...@conference.freeswitch.org
 213-799-1400

This time too, you all can follow the conference calling Skype the
skypeuser skypiax5, then press 1 on the Skype dialpad (max 20
concurrent users).

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Skypiax false DTMF event

2009-09-10 Thread Giovanni Maruzzelli

On Wed, Sep 9, 2009 at 10:04 PM, Dmitry Bely dmitry.b...@gmail.com wrote:
 I have a problem. After 10-20 minutes of Skype talk via cordless phone
 connected to ATA the latter erroneously generated DTMF 'D'  event.
 Then skypiax looses connection while the call remain active in Skype
 client. The only way to terminate it is to ask another party to hang
 up:

Ciao Dmitry,

could you please fill a Jira with the same infos?

http://wiki.freeswitch.org/wiki/Skypiax#How_To_Report_BUGS_and_Feature_Requests

That is the standard and correct procedure for bugs, so the devs can
follow up on it.

-giovanni



 (...)

 2009-09-09 22:20:07.474051 [DEBUG] skypiax_protocol.c:104 rev
 14707[(nil)|37     ][DEBUG_SKYPE  104 ][interface1][-1, 5,21] READING:
 |||CALL 307 DURATION 500|||
 2009-09-09 22:20:08.473755 [DEBUG] skypiax_protocol.c:104 rev
 14707[(nil)|37     ][DEBUG_SKYPE  104 ][interface1][-1, 5,21] READING:
 |||CALL 307 DURATION 501|||
 2009-09-09 22:20:09.474247 [DEBUG] skypiax_protocol.c:104 rev
 14707[(nil)|37     ][DEBUG_SKYPE  104 ][interface1][-1, 5,21] READING:
 |||CALL 307 DURATION 502|||
 2009-09-09 22:20:10.474611 [DEBUG] skypiax_protocol.c:104 rev
 14707[(nil)|37     ][DEBUG_SKYPE  104 ][interface1][-1, 5,21] READING:
 |||CALL 307 DURATION 503|||
 2009-09-09 22:20:11.474456 [DEBUG] skypiax_protocol.c:104 rev
 14707[(nil)|37     ][DEBUG_SKYPE  104 ][interface1][-1, 5,21] READING:
 |||CALL 307 DURATION 504|||
 2009-09-09 22:20:12.411664 [DEBUG] switch_rtp.c:2239 RTP RECV DTMF D:2000
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:633 rev
 14771[(nil)|37     ][DEBUG_SKYPE  633  ][interface1][-1, 5,21]
 interface1 CHANNEL SEND_DTMF
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:634 rev
 14771[(nil)|37     ][DEBUG_SKYPE  634  ][interface1][-1, 5,21] DTMF: D
 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:882 rev
 14707[(nil)|37     ][DEBUG_SKYPE  882  ][interface1][-1, 5,21] DIGIT
 received: D
 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1352 rev
 14707[(nil)|37     ][DEBUG_SKYPE  1352 ][interface1][-1, 5,21]
 SENDING: |||SET CALL 307 DTMF D
 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1530 rev
 14707[(nil)|37     ][DEBUG_SKYPE  1530 ][interface1][-1, 5,21] Got a
 'continue' XAtom without a previous 'begin'. It's value (between
 vertical bars) is=|||allowed call prop|||
 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:104 rev
 14707[(nil)|37     ][DEBUG_SKYPE  104  ][interface1][-1, 5,21]
 READING: |||ERROR 21 Unknown/dis|||
 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:144 rev
 14707[(nil)|37     ][ERRORA  144  ][interface1][-1, 5,21] Skype got
 ERROR: |||ERROR 21 Unknown/dis|||
 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:146 rev
 14707[(nil)|37     ][ERRORA  146  ][interface1][-1, 5,16] skype_call
 now is DOWN
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:1011 rev
 14771[(nil)|37     ][DEBUG_SKYPE  1011 ][interface1][-1, 1,16] skype
 call ended
 2009-09-09 22:20:12.411664 [NOTICE] mod_skypiax.c:1022 Hangup
 skypiax/interface1/user2 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
 2009-09-09 22:20:12.411664 [DEBUG] switch_channel.c:1715 Send signal
 skypiax/interface1/user2 [KILL]
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev
 14771[(nil)|37     ][DEBUG_SKYPE  566  ][interface1][-1, 1,16]
 interface1 CHANNEL KILL_CHANNEL
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:569 rev
 14771[(nil)|37     ][DEBUG_SKYPE  569  ][interface1][-1, 1,16]
 skypiax/interface1/user2 CHANNEL got SWITCH_SIG_KILL
 2009-09-09 22:20:12.411664 [DEBUG] switch_core_session.c:932 Send
 signal skypiax/interface1/user2 [BREAK]
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev
 14771[(nil)|37     ][DEBUG_SKYPE  566  ][interface1][-1, 1,16]
 interface1 CHANNEL KILL_CHANNEL
 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:589 rev
 14771[(nil)|37     ][DEBUG_SKYPE  589  ][interface1][-1, 1,16]
 skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK
 2009-09-09 22:20:12.428590 [DEBUG] skypiax_protocol.c:670 rev
 14707[(nil)|37     ][DEBUG_SKYPE  670  ][interface1][-1, 1,16] Skype
 incoming audio GONE
 2009-09-09 22:20:12.428590 [DEBUG] mod_skypiax.c:702 rev
 14771[(nil)|37     ][DEBUG_SKYPE  702  ][interface1][-1, 1,16] CHANNEL
 READ FALSE
 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:377
 skypiax/interface1/user2 ending bridge by request from read function
 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
 THREAD DONE [skypiax/interface1/user2]
 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:454 Send signal
 sofia/internal/1...@192.168.121.66        [BREAK]
 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:497
 (skypiax/interface1/user2) State EXCHANGE_MEDIA going to sleep
 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:398
 (skypiax/interface1/user2) Running State Change CS_HANGUP
 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434
 (skypiax/interface1/user2) State HANGUP
 2009-09-09

Re: [Freeswitch-users] Skypiax working but laggy

2009-09-06 Thread Giovanni Maruzzelli

Ubuntu 9.04 is explicitly discouraged, for heavy duty, if you like
Ubuntu, use 8.04.

That said, how is your call flow?

I mean:

Skypeclient-FS-SIP is laggy? How much? (1 sec, 10 sec, ...)
SIP-FS-Skypeclient is not laggy?

Or you mean that one side hear the other in real time, while the other
side hear the other with a lag?

Can you describe the problem with full informations?

(what kind of protocols, clients, how the calls are originated, how
are answered, etc etc etc etc etc :-) )

-giovanni


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Sun, Sep 6, 2009 at 3:24 PM, Gonzalo Servatgser...@gmail.com wrote:
 Hi All,
 I'm just testing out mod_skypiax (great work Giovanni  co!) and while it's
 working and all, I find that when I call in from a Skype contact, it's
 /very/ laggy. I would say something on the skype end and I would hear it on
 the FS end quite a bit later. Funny thing is the audio going the other way
 has much faster response.
 I'm running FreeSWITCH Version 1.0.trunk (14772) on Ubuntu 9.04 Jaunty. Do
 you need more info?
 Thanks,
 Gonzalo
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[Freeswitch-users] mod_skypiax for OSX?????

2009-09-05 Thread Giovanni Maruzzelli

Seeeven!

I saw the modification you made on the wiki page...

You made it, mod_skypiax runs on OSX

Let's merge your mods on the mainline, plese ;-)))

-giovanni




Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] mod_skypiax for OSX?????

2009-09-05 Thread Giovanni Maruzzelli

Seven,

thanks a lot for your efforts.

I will merge it in the next days, and I will take care that it will
not breaks Windows or Linux.

If I find problems I will wait for you having more time in the future.

I send you my super best wishes for your personal things to go well
and solves in the best of the possible ways.

ciao for now,

-giovanni



Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Sat, Sep 5, 2009 at 1:13 PM, Seven Dudujinf...@gmail.com wrote:
 gm,

 Thanks a lot you can merge into the mainline. I check into my branch
 because it's currently not as useful as on Linux and Windows and the
 solution is not good. But it works and it is a good start that
 mod_skypiax runs on OSX. Sure it would be easier for people want to
 test and improve it if it been merged into trunk. I think you can make
 a diff by

 svn diff -r 14472:14772 
 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax

 FYI for personal reason I won't have much time put on this in the
 coming month. Actually the code was done a few weeks ago but i only
 got a chance to commit it yesterday. Sure that is not to say I cannot
 do but fixes. But can you please make sure it won't break Linux/
 windows build when you merge the code? I haven't have a chance to test
 all of them yet.

 -7-

 On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote:
 Seeeven!

 I saw the modification you made on the wiki page...

 You made it, mod_skypiax runs on OSX

 Let's merge your mods on the mainline, plese ;-)))

 -giovanni




 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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Re: [Freeswitch-users] skypiax error

Jingwei,

those are normal warnings made by the Skype client (not by
mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment
out hdmi lines. Is a problem with a lazy implementation of that file,
that supposes you got an hdmi.

The other warning is because there are some files missing from the
Xvfb installation made by centos, but are completely harmless. In the
future I will make the script to redirect them to /dev/null :-)

Bottom line: all is OK.

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
Hi Folks,

I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
After having followed the big help doc from the wiki page
(http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
I hit an error when running multi.sh (under
freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username).

Couldn't open RGB_DB '/usr/share/X11/rgb'
error opening security policy file /usr/lib64/xserver/SecurityPolicy

This error seems not stopping xvfb from getting started. Then I started FS
and loaded mod_skypiax. However, when I initiated a test call (originate
skypiax/ANY/userAAA echo), I saw a bunch of ALSA lib errors popping up:

ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi

Again, this error doesn't prohibit the call from reaching me. It's just too
annoying and it keeps popping up after a while. Does anyone know how to get
rid of those errors?

I found a similar post here:
http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html.
Phil, if you happen to see my question, could you please reply and let me
know what the cause was and what you have done to solve it? Thanks!

By the way, I started xvfb and FS using root.

Regards,
-Jingwei

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Re: [Freeswitch-users] skypiax error

:-) My fault, I would have to document this.

I'll do pretty soon.

Sorry about that, and thanks for reporting!!!

-gm


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Fri, Sep 4, 2009 at 10:38 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
 Hi Giovanni,

 That's a big relief. Thanks a lot for the reply :)

 Regards,
 -Jingwei

 On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Jingwei,

 those are normal warnings made by the Skype client (not by
 mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment
 out hdmi lines. Is a problem with a lazy implementation of that file,
 that supposes you got an hdmi.

 The other warning is because there are some files missing from the
 Xvfb installation made by centos, but are completely harmless. In the
 future I will make the script to redirect them to /dev/null :-)

 Bottom line: all is OK.

 -giovanni

 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618




 On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yangjingwei.y...@gmail.com
 wrote:
  Hi Folks,
 
  I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine.
  After having followed the big help doc from the wiki page
 
  (http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch),
  I hit an error when running multi.sh (under
 
  freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username).
 
  Couldn't open RGB_DB '/usr/share/X11/rgb'
  error opening security policy file /usr/lib64/xserver/SecurityPolicy
 
  This error seems not stopping xvfb from getting started. Then I started
  FS
  and loaded mod_skypiax. However, when I initiated a test call (originate
  skypiax/ANY/userAAA echo), I saw a bunch of ALSA lib errors popping up:
 
  ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
 
  Again, this error doesn't prohibit the call from reaching me. It's just
  too
  annoying and it keeps popping up after a while. Does anyone know how to
  get
  rid of those errors?
 
  I found a similar post here:
 
  http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html.
  Phil, if you happen to see my question, could you please reply and let
  me
  know what the cause was and what you have done to solve it? Thanks!
 
  By the way, I started xvfb and FS using root.
 
  Regards,
  -Jingwei
 
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Re: [Freeswitch-users] skypiax error

On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelligmar...@celliax.org wrote:
 :-) My fault, I would have to document this.

http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Error_and_warnings_at_the_starting_of_Skype_clients_on_Linux

-giovanni

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Re: [Freeswitch-users] skypiax error

Updated the wiki page with references to other errors/warnings as well :-)

-giovanni

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Re: [Freeswitch-users] REMINDER: Weekly call is now happening. Join us!

For the ones SIP challenged:

call Skype the skypeuser skypiax5 and then press 1

-gm

On Fri, Sep 4, 2009 at 6:43 PM, Diego Violadiego.vi...@gmail.com wrote:
 I'm in, very cool =D

 Diego

 On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins m...@freeswitch.org wrote:

 Hello all,

 We are now on line and welcoming callers. Here's the agenda so far:
 http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04

 Come join the conference
 sip:8...@conference.freeswitch.org
 1-213-799-1400

 -MC

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Re: [Freeswitch-users] Hello, and stuff.

Hi Tom,

best is to try it for yourself, you cannot expect from the FS mailing
list an answer like: you know, fs is nor a marked improvement on
anything, we just like to spend time together :-)

-gm


On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tompaveraw...@hotmail.com wrote:
 As a background, I ran an asterisk consulting company for about 3 years that
 I gave up on 2 years ago after repeatedly failing to achieve any sort of
 stability on any sort install over about 30 phones, I gave up.

 Maybe that was wrong, I am open to the possibility that I just didn't know
 enough and I was building things wrong, but I worked inside the asterisk
 code (which I feel is a hopeless mess), I implemented a few small custom
 features, anyway...

 I'm coming back into the VoIP space now, and I'm wondering what sort of
 issues can I expect in trying to pick up and learn freeswitch?  From what
 I've read on the website, it appears to have a much more sane architecture.
 I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
 stable and worst general call quality was asterisk.  I constantly contended
 with strange call quality issues in asterisk, lots of echo (even with
 hardware echo cancellation cards), lots of jitter, lots of call break up
 (even on small systems with 10-20 users, using QoS on the network, and in
 general doing everything I could to prioritize voice over anything else).

 When I used Cisco call manager and broadsoft, the voice quality issues were
 basically non-existant, as long as the network was running QoS echo,
 stutter, calls breaking up, just didn't happen.  So, I guess my question is,
 does freeswitch show a marked improvement over asterisk in this department?
 As long as you configure QoS and have hardware echo cancellation does it
 actually work reliably?

 Thanks for any additional information about freeswitch you can provide as
 well.  I am a software developer primarily by trade, but I do lots of
 consulting type work in the SME space and I've had a couple projects thrown
 to me that require some integration with a phone system, and I just can't in
 good conscience recommend asterisk anymore.


 -Tom

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Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

Usually you don't need to worry about stability issues with FS.

For scalability, peoples tend to use openser or some other sip
loadbalancer in fron of fs, but you probably would not need that.

Live migration of calls is not yet possible, tough.

-giovanni



On Fri, Aug 28, 2009 at 8:52 PM, Raimund Sachererr...@runsolutions.com wrote:
 Hello List,

 I have read the current thread about scalability and I would need some
 advice about a callcenter setup:

 First where I come from:
 I have lot's of problems with an asterisk solution. I have regular
 crash's and lock-ups, with downgrading and other stuff i got it
 somewhat stable, but have nevertheless regular hickups. I am desperate
 and want to get rid of asterisk and I hope that freeSwitch will
 provide me with a more stable solution.


 Our Setup (really nothing special):
 * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
 GIG of RAM, Xircom analog switch connected to mobile stations for 4
 different providers, Digium 4port cards TP400something
 * 8 queues
 * ~60 agents (which logon, logoff, pause, unpause), not more than 40
 concurrently online
 * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
 * Music on Hold in the call-queues
 * No special announcement
 * Transfers between calls in queues and different agents as well as
 non agents (i mention this because we have transfer related chrashes
 in asterisk)

 The current Problems:
 * Lockups with different causes (ranging from calls not terminated to
 heavy thread locking through the AMI interface)
 * Crashes and library aborts (pthread, libc, crashes related to music
 on hold, app_queue, transfers)

 We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
 now back to 1.4.21.2 (stock debian) as anything beyond that is for
 whatever reason highly unstable for our szenario. Maybe we should have
 been segmenting the box into one asterisk dedicted to talking to the
 hardware, one especial for queue/sip handling, i do not know. (all
 issues are well documented in issues.asterisk.org, but it seems to be
 very, very difficult to get to the bottom of them as they exist since
 1.4.23 as it seems and are open until know and not fixable since month.)


 Now, I really would appreciate some success-stories on how you guys
 managed to get a stable pbx system with freeSWITCH in regard of HA and
 scalability:

 * How to segment freeSWITCH? Or is it stable enough to handle all in
 one for such a szenario as outlined above?
 * What would be the best strategie for High Availability / Failover?
        - I read in the WIKI (featurelist) that Livemigration of calls from
 one box to another should be possible?
        - I was thinking about using memcached for storing all state
 information so another freeswitch box can takeover calls from the
 first box if it dies, is this possible? If so, how?
        - Is there anotherway to somehow configure freeSWITCH that in the
 event of a crash i do not loose the current established calls?

 Basically I just want a stable PBX where I do not have to fear every
 day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
 so i can sleep at night for at least 3 month without a crash?



 Thank you for your Time and help in advance, and I am more than
 willing to take all the information gathered here and create a wiki
 page to help other people with the same questions/problems.

 best
 Ray


 --
 Raimund Sacherer
 -
 RunSolutions
     Open Source It Consulting
 -

 Parc Bit - Centro Empresarial Son Espanyol
 Edificio Estel - Local 3D
 07121 -  Palma de Mallorca
 Baleares


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Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch

On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayebtayeb.mef...@gmail.com wrote:
 if i load the module manualy, for the first load crach and don't detect
 the active runing skype client
 but if i do load mod_skypiax aguin will load and work perfectly

Meftah,

I've not seen that bug, it works first time for me.

Anyway, I'll double check it, and I open A jira for you ;-)

-giovanni



 thanks
 Giovanni Maruzzelli wrote:
 Hi FreeSWITCHers,

 all the users of mod_skypiax are kindly requested to test the svn trunk 
 14519.

 It contains a lot of changes meant to add stability and robustness,
 toward a production environment.

 Let me know how your feelings, and please add to the Jira any possible
 bug/issue/etc.

 Thanks to you all,

 -giovanni



 Sincerely,

 Giovanni Maruzzelli

 Cell : +39-347-2665618

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Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch

http://jira.freeswitch.org/browse/MODSKYPIAX-49


On Fri, Aug 28, 2009 at 9:41 PM, Giovanni Maruzzelligmar...@celliax.org wrote:
 On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayebtayeb.mef...@gmail.com wrote:
 if i load the module manualy, for the first load crach and don't detect
 the active runing skype client
 but if i do load mod_skypiax aguin will load and work perfectly

 Meftah,

 I've not seen that bug, it works first time for me.

 Anyway, I'll double check it, and I open A jira for you ;-)

 -giovanni



 thanks
 Giovanni Maruzzelli wrote:
 Hi FreeSWITCHers,

 all the users of mod_skypiax are kindly requested to test the svn trunk 
 14519.

 It contains a lot of changes meant to add stability and robustness,
 toward a production environment.

 Let me know how your feelings, and please add to the Jira any possible
 bug/issue/etc.

 Thanks to you all,

 -giovanni



 Sincerely,

 Giovanni Maruzzelli

 Cell : +39-347-2665618

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Re: [Freeswitch-users] Hello, and stuff.

Welcome on board Tom! And sorry for being witty before ;-)
-gm


On Fri, Aug 28, 2009 at 9:54 PM, Christensen Tompaveraw...@hotmail.com wrote:
 I am totally fine without a slick GUI interface.  The first 2 years of
 asterisk stuff I did was all in on the CLI in editor of your choice (I use
 vim most of the time, but not for religious reasons...).
 Anyway, thanks for the info, I'll be setting up a freeswitch system this
 weekend expect to see me on IRC and here..
 Thanks!
 -Tom
 
 Date: Fri, 28 Aug 2009 11:54:25 -0700
 From: m...@freeswitch.org
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Hello, and stuff.

 Tom,
 Welcome! Sadly, your experience is not unique...

 On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom paveraw...@hotmail.com
 wrote:

 As a background, I ran an asterisk consulting company for about 3 years that
 I gave up on 2 years ago after repeatedly failing to achieve any sort of
 stability on any sort install over about 30 phones, I gave up.

 The consensus I've seen is that the larger the install, the more likely one
 is to have inexplicable issues.


 Maybe that was wrong, I am open to the possibility that I just didn't know
 enough and I was building things wrong, but I worked inside the asterisk
 code (which I feel is a hopeless mess), I implemented a few small custom
 features, anyway...

 Any software that openly admits that a function is pure nastiness but
 doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
 leadership IMHO. (grep the Asterisk source tree for nastiness and you'll
 see it.)


 I'm coming back into the VoIP space now, and I'm wondering what sort of
 issues can I expect in trying to pick up and learn freeswitch?  From what
 I've read on the website, it appears to have a much more sane architecture.
 I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
 stable and worst general call quality was asterisk.  I constantly contended
 with strange call quality issues in asterisk, lots of echo (even with
 hardware echo cancellation cards), lots of jitter, lots of call break up
 (even on small systems with 10-20 users, using QoS on the network, and in
 general doing everything I could to prioritize voice over anything else).

 Again, your experience isn't unique...

 When I used Cisco call manager and broadsoft, the voice quality issues were
 basically non-existant, as long as the network was running QoS echo,
 stutter, calls breaking up, just didn't happen.  So, I guess my question is,
 does freeswitch show a marked improvement over asterisk in this department?
 As long as you configure QoS and have hardware echo cancellation does it
 actually work reliably?

 We receive lots of reports that FreeSWITCH is a vast improvement over not
 only Asterisk but proprietary solutions as well. The FS architecture is, as
 you mentioned, not insane. It is well thought out and therefore highly
 flexible, extensible, and scalable. I'm not aware of anything - OSS or
 proprietary - that can match FS in these three areas.

 Thanks for any additional information about freeswitch you can provide as
 well.  I am a software developer primarily by trade, but I do lots of
 consulting type work in the SME space and I've had a couple projects thrown
 to me that require some integration with a phone system, and I just can't in
 good conscience recommend asterisk anymore.

 Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
 development but the power users are quite happy with doing the emacs (or
 vim) shuffle with the XML config files. Furthermore, the ways that FS allows
 you to connect and control are fantastic: mod_xml_curl for dynamic
 configurations, event-socket for external control (think of it like AMI not
 sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
 the list is impressive.

 I can honestly say that every week we get new people looking at FreeSWITCH
 and saying, Wow, this is incredible. I can definitely, in good conscience,
 recommend you investigate FS more deeply. I'm confident you'll be happy with
 the return on your investment.

 Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if
 you want to chat in real-time.
 -Michael



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Re: [Freeswitch-users] Can a chat message be sent to a cell phone with FS?

2009-08-27 Thread Giovanni Maruzzelli

In a short while (for any value of short) will be available for
testing mod-celliax, an interface to the cellular phones networks for
voice calls and SMSs.

-giovanni


On 8/27/09, Merle J. Ebbert se02005-...@yahoo.com wrote:
 Can a chat message be sent to a cell phone with FS?

 Thanks,
 Merle

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-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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[Freeswitch-users] HOW-TO: being on trunk of FreePBX, starting from the ISO

2009-08-26 Thread Giovanni Maruzzelli

Instructions for being on trunk of FreePBX, starting from the ISO
(cut and paste to the ssh console after ISO install):

/etc/init.d/httpd stop
cd /var/www/html
mv freepbx freepbx-originale
svn co http://www.freepbx.org/v3/svn/trunk/ freepbx
chown -R apache.apache freepbx
ln -s freepbx  freepbx-v3
cd freepbx-v3/
ln -s freepbx  freepbx-v3
/etc/init.d/httpd start

then browse to:

http://192.168.0.12/freepbx-v3/index.php/installer

it will work! :-)


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?

2009-08-26 Thread Giovanni Maruzzelli

You will find all the informations, and then some, here:
http://wiki.freeswitch.org

Then, after reading and testing and experimenting, come to the IRC
channel for direct help with esoteric (or not so esoteric) problems.

You'll find a nice and friendly community.

-gm




Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Wed, Aug 26, 2009 at 3:25 PM, Merle J. Ebbertse02005-...@yahoo.com wrote:

 Hi,

 I'm trying to avoid taking up a lot of peoples valuable time.

 SIP  FS have  brought some ideas for some commercial products but I
 need to know where to start.

 Having once written a proprietary DOS  helped with writing a RTOS, I
 consider myself capable
 of learning.  I (we) just need to know where to start to come up to
 speed rapidly.

 Is there a FreeSWITCH tutorial available?
 Should someone new start with Asterisk and then possibly move to FS?

 Thanks,
 Merle


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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Re: [Freeswitch-users] Freeswitch performance as a redirecting server

Maybe your load comes from disk access?

Try putting the sql and log directories on a ramdisk.

OTH,

-giovanni

On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjagatculj...@gmail.com wrote:
 Hello,

 i'm trying to use freeswitch as a redirecting server meaning FS has to
 receive an INVITE and according to some rules it will redirect calls to
 other destinations.


 CALLING_USER    FREESWITCH    SOMEWHERE

 INVITE ---
    -- 100 Trying
    -- 302 Moved Temporary
 ACK    ---
 INVITE-



 Well, wverything works well except i have perfromance issues  on my HW
 FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary). When
 i increase the rate, FS starts delaying 302 response. Right at 50 CPS i see
 calls being build up in FS and the delay begining to grow.

 When i observe the machine, load average is almost nothing (load average:
 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread taking
 most load... all others are just sitting there with 1-5 % CPU time.
 This looks to me as FS handles 302 messages in a single thread?!?!


 tculj...@fs:/usr/local/freeswitch/conf/dialplan$ top -H

 top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61, 0.60
 Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
 Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
 0.0%st
 Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
 Swap:  2650684k total, 3020k used,  2647664k free,   153868k cached

   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
 COMMAND
  4814 root  20   0 34188  20m 3780 S   38  1.0   3:10.29
 freeswitch
  4800 root  20   0 34188  20m 3780 S    6  1.0   0:08.26
 freeswitch
  4798 root  20   0 34188  20m 3780 R    5  1.0   0:24.46
 freeswitch
  4787 root  20   0 34188  20m 3780 S    2  1.0   0:11.24
 freeswitch
  4794 root  20   0 34188  20m 3780 S    1  1.0   0:11.42
 freeswitch
  4803 root  20   0 34188  20m 3780 S    1  1.0   0:11.74
 freeswitch
  4788 root  20   0 34188  20m 3780 S    1  1.0   0:02.96
 freeswitch
  4804 root  20   0 34188  20m 3780 S    1  1.0   0:01.64
 freeswitch
  4807 root  20   0 34188  20m 3780 S    1  1.0   0:01.68
 freeswitch
  4811 root  20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch



 cat /proc/cpuinfo
 processor   : 0
 vendor_id   : GenuineIntel
 cpu family  : 6
 model   : 15
 model name  : Intel(R) Xeon(R) CPU    5140  @ 2.33GHz
 stepping    : 6
 cpu MHz : 2333.560
 cache size  : 4096 KB
 physical id : 0
 siblings    : 2
 core id : 0
 cpu cores   : 2
 apicid  : 0
 initial apicid  : 0
 fdiv_bug    : no
 hlt_bug : no
 f00f_bug    : no
 coma_bug    : no
 fpu : yes
 fpu_exception   : yes
 cpuid level : 10
 wp  : yes
 flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
 cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
 constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
 xtpr dca lahf_lm
 bogomips    : 4670.78
 clflush size    : 64
 power management:

 processor   : 1
 vendor_id   : GenuineIntel
 cpu family  : 6
 model   : 15
 model name  : Intel(R) Xeon(R) CPU    5140  @ 2.33GHz
 stepping    : 6
 cpu MHz : 2333.560
 cache size  : 4096 KB
 physical id : 0
 siblings    : 2
 core id : 1
 cpu cores   : 2
 apicid  : 1
 initial apicid  : 1
 fdiv_bug    : no
 hlt_bug : no
 f00f_bug    : no
 coma_bug    : no
 fpu : yes
 fpu_exception   : yes
 cpuid level : 10
 wp  : yes
 flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
 cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
 constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
 xtpr dca lahf_lm
 bogomips    : 4666.82
 clflush size    : 64
 power management:



 uname -a
 Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
 GNU/Linux



 Of course, i've tuned the machine up

 ulimit -c unlimited
 ulimit -d unlimited
 ulimit -f unlimited
 ulimit -i unlimited
 ulimit -n 99
 ulimit -q unlimited
 ulimit -u unlimited
 ulimit -v unlimited
 ulimit -x unlimited
 ulimit -s 240
 ulimit -l unlimited
 ulimit -a


 Started FS with minimum modules but still 40 CPS seems to be the limit.


 So, is there any way to improve performance?


 Tihomir.







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Re: [Freeswitch-users] Freeswitch performance as a redirecting server

Definitely go for 64 bit OS.

If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
used both for development and for heavy duty production.

Also Ubuntu 8.04 is good.

Other versions/distros are less used by the community.

Adding RAM and CPUs helps to scale up.

-gm



Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjagatculj...@gmail.com wrote:
 Hey Giovanni,

 thanks for the tip... indeed the db files were heavily used regardless if i
 started freeswitch with nosql option (freeswitch -nosql)... FS was not
 writing anything into that files ... instead it was just accessing it
 This behaviour leads to a waste of 40% CPU time... waiting for other
 processes (mainly disk access) to finish!!!

 I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
 CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
 minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
 irresponsive :).


 What can be wrong?
 What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
 cross?
 What fine tuning do we need in order to asure a long high load run?



 Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have sense
 to move my OS to 64 bit? ... will FS gain more preformance ?... I mean will
 FS perofomr drastically better 20%+ ?


 Tihomir.


 On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Maybe your load comes from disk access?

 Try putting the sql and log directories on a ramdisk.

 OTH,

 -giovanni

 On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjagatculj...@gmail.com
 wrote:
  Hello,
 
  i'm trying to use freeswitch as a redirecting server meaning FS has to
  receive an INVITE and according to some rules it will redirect calls to
  other destinations.
 
 
  CALLING_USER    FREESWITCH    SOMEWHERE
 
  INVITE ---
     -- 100 Trying
     -- 302 Moved Temporary
  ACK    ---
 
  INVITE-
 
 
 
  Well, wverything works well except i have perfromance issues  on my
  HW
  FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
  When
  i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
  see
  calls being build up in FS and the delay begining to grow.
 
  When i observe the machine, load average is almost nothing (load
  average:
  1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
  taking
  most load... all others are just sitting there with 1-5 % CPU time.
  This looks to me as FS handles 302 messages in a single thread?!?!
 
 
  tculj...@fs:/usr/local/freeswitch/conf/dialplan$ top -H
 
  top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61,
  0.60
  Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
  Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
  0.0%st
  Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
  Swap:  2650684k total, 3020k used,  2647664k free,   153868k cached
 
    PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
  COMMAND
   4814 root  20   0 34188  20m 3780 S   38  1.0   3:10.29
  freeswitch
   4800 root  20   0 34188  20m 3780 S    6  1.0   0:08.26
  freeswitch
   4798 root  20   0 34188  20m 3780 R    5  1.0   0:24.46
  freeswitch
   4787 root  20   0 34188  20m 3780 S    2  1.0   0:11.24
  freeswitch
   4794 root  20   0 34188  20m 3780 S    1  1.0   0:11.42
  freeswitch
   4803 root  20   0 34188  20m 3780 S    1  1.0   0:11.74
  freeswitch
   4788 root  20   0 34188  20m 3780 S    1  1.0   0:02.96
  freeswitch
   4804 root  20   0 34188  20m 3780 S    1  1.0   0:01.64
  freeswitch
   4807 root  20   0 34188  20m 3780 S    1  1.0   0:01.68
  freeswitch
   4811 root  20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch
 
 
 
  cat /proc/cpuinfo
  processor   : 0
  vendor_id   : GenuineIntel
  cpu family  : 6
  model   : 15
  model name  : Intel(R) Xeon(R) CPU    5140  @ 2.33GHz
  stepping    : 6
  cpu MHz : 2333.560
  cache size  : 4096 KB
  physical id : 0
  siblings    : 2
  core id : 0
  cpu cores   : 2
  apicid  : 0
  initial apicid  : 0
  fdiv_bug    : no
  hlt_bug : no
  f00f_bug    : no
  coma_bug    : no
  fpu : yes
  fpu_exception   : yes
  cpuid level : 10
  wp  : yes
  flags   : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
  mca
  cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
  constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
  cx16
  xtpr dca lahf_lm
  bogomips    : 4670.78
  clflush

Re: [Freeswitch-users] Freeswitch performance as a redirecting server

is a heavely multithreaded software, it benefits from number of CPUs
(or cores), RAM, and heavy duty kernel features (found in 64bit
kernels)

put all accesses on ramdisk, leave out the modules you don't use...

experiment, test, and find the best for your specific application/workload

test not only with sipp, but with real load too (often they're very different)

-gm


On Tue, Aug 25, 2009 at 3:42 PM, Tihomir Culjagatculj...@gmail.com wrote:
 thanks for the feedback... this is something im going to do tomorrow...

 what about other things?


 On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks jaybi...@gmail.com wrote:

 Everytime someone asks this ,  the resounding answer is use a 64bit os..
 No question
 Jay



 On 25/08/2009, at 23:19, Tihomir Culjaga tculj...@gmail.com wrote:

 Hey Giovanni,

 thanks for the tip... indeed the db files were heavily used regardless if
 i started freeswitch with nosql option (freeswitch -nosql)... FS was not
 writing anything into that files ... instead it was just accessing it
 This behaviour leads to a waste of 40% CPU time... waiting for other
 processes (mainly disk access) to finish!!!

 I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
 CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
 minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
 irresponsive :).


 What can be wrong?
 What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
 cross?
 What fine tuning do we need in order to asure a long high load run?



 Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
 sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
 will FS perofomr drastically better 20%+ ?


 Tihomir.


 On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
 gmar...@celliax.org wrote:

 Maybe your load comes from disk access?

 Try putting the sql and log directories on a ramdisk.

 OTH,

 -giovanni

 On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjagatculj...@gmail.com
 wrote:
  Hello,
 
  i'm trying to use freeswitch as a redirecting server meaning FS has to
  receive an INVITE and according to some rules it will redirect calls to
  other destinations.
 
 
  CALLING_USER    FREESWITCH    SOMEWHERE
 
  INVITE ---
     -- 100 Trying
     -- 302 Moved Temporary
  ACK    ---
 
  INVITE-
 
 
 
  Well, wverything works well except i have perfromance issues  on my
  HW
  FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
  When
  i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
  see
  calls being build up in FS and the delay begining to grow.
 
  When i observe the machine, load average is almost nothing (load
  average:
  1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
  taking
  most load... all others are just sitting there with 1-5 % CPU time.
  This looks to me as FS handles 302 messages in a single thread?!?!
 
 
  tculj...@fs:/usr/local/freeswitch/conf/dialplan$ top -H
 
  top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61,
  0.60
  Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
  Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
  0.0%st
  Mem:   2074520k total,   571244k used,  1503276k free,   259604k
  buffers
  Swap:  2650684k total, 3020k used,  2647664k free,   153868k cached
 
    PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
  COMMAND
   4814 root  20   0 34188  20m 3780 S   38  1.0   3:10.29
  freeswitch
   4800 root  20   0 34188  20m 3780 S    6  1.0   0:08.26
  freeswitch
   4798 root  20   0 34188  20m 3780 R    5  1.0   0:24.46
  freeswitch
   4787 root  20   0 34188  20m 3780 S    2  1.0   0:11.24
  freeswitch
   4794 root  20   0 34188  20m 3780 S    1  1.0   0:11.42
  freeswitch
   4803 root  20   0 34188  20m 3780 S    1  1.0   0:11.74
  freeswitch
   4788 root  20   0 34188  20m 3780 S    1  1.0   0:02.96
  freeswitch
   4804 root  20   0 34188  20m 3780 S    1  1.0   0:01.64
  freeswitch
   4807 root  20   0 34188  20m 3780 S    1  1.0   0:01.68
  freeswitch
   4811 root  20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch
 
 
 
  cat /proc/cpuinfo
  processor   : 0
  vendor_id   : GenuineIntel
  cpu family  : 6
  model   : 15
  model name  : Intel(R) Xeon(R) CPU    5140  @ 2.33GHz
  stepping    : 6
  cpu MHz : 2333.560
  cache size  : 4096 KB
  physical id : 0
  siblings    : 2
  core id : 0
  cpu cores   : 2
  apicid  : 0
  initial apicid  : 0
  fdiv_bug    : no
  hlt_bug : no
  f00f_bug    : no
  coma_bug    : no
  fpu

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

Carlos,

you're very kind, as always.

I'm aware that this is a dev preview, and I'm interested just in that,
to begin getting acquainted with the framework (and adding support to
the endpoints/trunk I take care of).

I probably have not got the logic right :-) (I tried both Windows
Installer and Linux ISO)

I started fpbx with fs running, it works. I create a number, then I
create a device and I connect it to that number. It works.

If I restart FS, do not works anymore, complaining no sofia profiles.

From the front page of FPBX is not clear you *must* create a trunk/trunk group.

I was thinking trunks were for outgoing calls, or for receiving from external.

I was just testing internal phones, trying an IVR, so I was thinking
trunks were not needed.

Can you explain to me?

Thanks again,

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbotcarlos.tal...@gmail.com wrote:
 Giovanni,
 you mean like this message?
 Unable to determine location for device. Voicemail password set via FreePBX
 will not be valid.
 This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
 Let's keep in ming FreePBX v3 is a developer release and as such many
 features are in flux and might not work. That being said there some features
 in the Windows build that still do not work. The biggest one right now is
 the lack of the php ESL library for Windows which affects the voicemail app.
 I'm trying to get this to compile but it's been difficult.
 I do include the .svn files with the FreePBX install so you can freely
 install TortoiseSVN and update FreePBX at your leisure.
 With regards to the sip_profiles, did you create a trunk group and trunk?
 regards,
 Carlos

 On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Windows installer does not work for me.

 I've reinstalled various times, same results.

 I can correctly create a number, but when I try to create a device for
 that number, it tells me that cannot locate the device, and the
 password for vicemail will be invalid.

 After that, it begins to give the php error page, it cannot find the
 start  tag in directory/default.xml

 Also for me there are no sofia profiles...

 So, I cannot start to test it (eg: I would like to add mod_skypiax
 support to it).


 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618




 On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
 Guidiraffaele.p.gu...@gmail.com wrote:
  This is what I was asking! :D When the installer finished it started the
  whole thing and everything got loaded fine, but when I restarted my
  system
  it didn't (and did not anymore). Well, I will try to install everything
  from
  scratch again and see...
 
  On Mon, Aug 24, 2009 at 20:30, Brian West br...@freeswitch.org wrote:
 
  If you installed FreePBX then it would be that softwares job to manage
  the sofia profiles... wouldn't it?
 
  /b
 
  On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
 
   Actually I did that and it worked fine. I had the problem the SECOND
   time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
   sip_profiles were not
 
 
  ___
  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
 
 
  ___
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  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
 
 

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 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


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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org

Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

Darren,

thanks! it helps a lot.

If we svn up in /var/www/freepbx we got the trunk?


On Tue, Aug 25, 2009 at 10:59 PM, Darren Schreiberd...@d-man.org wrote:
 Hi there...

 So a few things on this.

 1) We have a module that's still being worked on called Sip Interface that
 allows you to configure Sip Profiles in FreeSWITCH. Unfortunately we don't
 have the ability to easily import your existing SIP profiles, and by NOT
 displaying them in the UI your stock config conflicts with those profiles.
 In other words, internal.xml and external.xml define sip profiles on ports
 5060 and 5080 that the GUI is unaware. So to simplify things, we just delete
 those files and expect you to recreate them via the UI on install. We'll
 probably make this a little more obvious in the near future, but for now,
 that's what we do.

 2) The trunk creation system in the ISO does give the impression that trunk
 groups  trunks are for external. This is a design flaw I have already fixed
 in trunk (referenced above). We have now split this module into two modules
 - one for configuring Sip Interface/Sip Profiles (which are for defining
 your IPs  ports to use for sending/receiving calls and authentication
 settings) and the other for defining gateways ('trunks) which are generally
 used for making outbound calls.

 So in other words, we're aware of the issues you are detailing. We will be
 finalizing the fix for this issue hopefully on Sunday and will rebuild the
 ISO by Monday.

 Hope that helps.

 - Darren



 
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Carlos
 Talbot
 Sent: Tuesday, August 25, 2009 12:02 PM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem


 This would be a question for Darren and the FreePBX group. :) I guess it
 does not help that the User Documentation link on this page is currently
 empty: http://www.freepbx.org/v3/wiki/
 If you note the message during FreePBX initialization *all* files in the
 sip_profiles directory are removed (including internal*.xml). This causes
 'sofia status' to come back empty.

 Incompatible Configuration

 WARNING: THE FOLLOWING FILES WILL BE DELETED!

 D:/FreeSWITCH/conf/sip_profiles/external.xml
 D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
 D:/FreeSWITCH/conf/sip_profiles/internal.xml


 regards,
 Carlos

 On Tue, Aug 25, 2009 at 1:28 PM, Giovanni
 Maruzzelli gmar...@celliax.org wrote:

 From the front page of FPBX is not clear you *must* create a trunk/trunk
 group.

 I was thinking trunks were for outgoing calls, or for receiving from
 external.

 I was just testing internal phones, trying an IVR, so I was thinking
 trunks were not needed.

 Can you explain to me?

 Thanks again,

 -giovanni

 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618




 On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbotcarlos.tal...@gmail.com
 wrote:
  Giovanni,
  you mean like this message?
  Unable to determine location for device. Voicemail password set via
  FreePBX
  will not be valid.
  This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
  Let's keep in ming FreePBX v3 is a developer release and as such many
  features are in flux and might not work. That being said there some
  features
  in the Windows build that still do not work. The biggest one right now
  is
  the lack of the php ESL library for Windows which affects the voicemail
  app.
  I'm trying to get this to compile but it's been difficult.
  I do include the .svn files with the FreePBX install so you can freely
  install TortoiseSVN and update FreePBX at your leisure.
  With regards to the sip_profiles, did you create a trunk group and
  trunk?
  regards,
  Carlos
 
  On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli
  gmar...@celliax.org
  wrote:
 
  Windows installer does not work for me.
 
  I've reinstalled various times, same results.
 
  I can correctly create a number, but when I try to create a device for
  that number, it tells me that cannot locate the device, and the
  password for vicemail will be invalid.
 
  After that, it begins to give the php error page, it cannot find the
  start  tag in directory/default.xml
 
  Also for me there are no sofia profiles...
 
  So, I cannot start to test it (eg: I would like to add mod_skypiax
  support to it).
 
 
  Sincerely,
 
  Giovanni Maruzzelli
  Cell : +39-347-2665618
 
 
 
 
  On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
  Guidiraffaele.p.gu...@gmail.com wrote:
   This is what I was asking! :D When the installer finished it started
   the
   whole thing and everything got loaded fine, but when I restarted my
   system
   it didn't (and did not anymore). Well, I will try to install
   everything
   from
   scratch again and see...
  
   On Mon, Aug 24, 2009 at 20:30, Brian West br...@freeswitch.org
   wrote:
  
   If you installed

Re: [Freeswitch-users] Screaming monkeys on ext 5000

2009-08-24 Thread Giovanni Maruzzelli

On Mon, Aug 24, 2009 at 5:31 AM, Brian Westbr...@freeswitch.org wrote:
 It requires internet connectivity.  It calls a remote system to play
 which is out of our control.


Yeah, I noted this too, since a couple weeks at least...

Maybe let's Todd know it's monkeys are out of voice?

-giovanni

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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

2009-08-24 Thread Giovanni Maruzzelli

Windows installer does not work for me.

I've reinstalled various times, same results.

I can correctly create a number, but when I try to create a device for
that number, it tells me that cannot locate the device, and the
password for vicemail will be invalid.

After that, it begins to give the php error page, it cannot find the
start  tag in directory/default.xml

Also for me there are no sofia profiles...

So, I cannot start to test it (eg: I would like to add mod_skypiax
support to it).


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
Guidiraffaele.p.gu...@gmail.com wrote:
 This is what I was asking! :D When the installer finished it started the
 whole thing and everything got loaded fine, but when I restarted my system
 it didn't (and did not anymore). Well, I will try to install everything from
 scratch again and see...

 On Mon, Aug 24, 2009 at 20:30, Brian West br...@freeswitch.org wrote:

 If you installed FreePBX then it would be that softwares job to manage
 the sofia profiles... wouldn't it?

 /b

 On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:

  Actually I did that and it worked fine. I had the problem the SECOND
  time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
  sip_profiles were not


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Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch

2009-08-15 Thread Giovanni Maruzzelli

On Sat, Aug 15, 2009 at 5:41 PM, Seven Dudujinf...@gmail.com wrote:
 And as I noticed you removed the sequential line hunting methods.

Because was broken. So, I aliased it to the RR.
If you think it can be useful, add a Jira for it

 Thank you very much for merging in the sk list with statistic patch,
Thanks to you for sending the patch! I've only added the callflow of
the skype client to it


 Two features are:

 continue load on fail: make sure the module continue load even it
 failed to talk to a skype instance

h, I'm too conservative for this one: I prefer that if you
configured a skype instance, you expect it to work, so the module must
fail if there is not such instance

 auto skype user: get the user name by the returned CURRENTUSERHANDLE
 other than  from the config xml, for easier config.

the username returned by CURRENTUSERHANDLE is checked against the
config file because is the only way you can associate interface_name
with its related Skype client instance on Windoz (no multiple X
servers there).

Thanks a lot for all your efforts!!!

-giovanni





 On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
 Hi FreeSWITCHers,

 all the users of mod_skypiax are kindly requested to test the svn
 trunk 14519.

 It contains a lot of changes meant to add stability and robustness,
 toward a production environment.

 Let me know how your feelings, and please add to the Jira any possible
 bug/issue/etc.

 Thanks to you all,

 -giovanni



 Sincerely,

 Giovanni Maruzzelli

 Cell : +39-347-2665618

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[Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch

2009-08-14 Thread Giovanni Maruzzelli

Hi FreeSWITCHers,

all the users of mod_skypiax are kindly requested to test the svn trunk 14519.

It contains a lot of changes meant to add stability and robustness,
toward a production environment.

Let me know how your feelings, and please add to the Jira any possible
bug/issue/etc.

Thanks to you all,

-giovanni



Sincerely,

Giovanni Maruzzelli

Cell : +39-347-2665618

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Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch

2009-08-14 Thread Giovanni Maruzzelli

svn 14521: skypiax: compiles on windoz, not yet tested (on windoz)



On Fri, Aug 14, 2009 at 7:43 PM, Giovanni Maruzzelli
gmar...@celliax.org wrote:
 Hi FreeSWITCHers,

 all the users of mod_skypiax are kindly requested to test the svn trunk 14519.

 It contains a lot of changes meant to add stability and robustness,
 toward a production environment.

 Let me know how your feelings, and please add to the Jira any possible
 bug/issue/etc.

 Thanks to you all,

 -giovanni



 Sincerely,

 Giovanni Maruzzelli

 Cell : +39-347-2665618


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Re: [Freeswitch-users] Cluecon 2009

2009-08-12 Thread Giovanni Maruzzelli

it helped me!
oh... well, I helped myself!

-giovanni


On Wed, Aug 12, 2009 at 11:30 PM, Brian Westbr...@freeswitch.org wrote:
 And it didn't help we had an open bar two of the nights!

 /b

 On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:

 That's the trouble with a 8am conference in a town where the bars
 close at 4am :-)


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Re: [Freeswitch-users] skypiax on Mac OS X

2009-08-09 Thread Giovanni Maruzzelli

Ciao Ivan,

it seems that you do not have the libX11 **development** package installed.

Unfortunately I don't know about OSX, so I cannot help you, but many
on the list know.

BTW: it will probably be of no use to you to compile mod_skypiax on
OSX, because Skype for MACOSX works in another way than Skype for
Linux.
If you know about MacOSX programming, please have a look at
https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
probably be simple enough to add a message pump for MacOSX.

-giovanni




Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvoldi...@myrvold.org wrote:
 I tried to compile mod_skypiax, and am getting problem with X11. On OS
 X Leopard, X11 is installed in /usr/X11/lib/
 See below.

 What can I do to get past this error?

 I can also let you ssh into my machine. Contact me off list in case.

 Ivan

 making all mod_skypiax
 Compiling skypiax_protocol.c...
 Compiling mod_skypiax.c...
 mkdir .libs
 Compiling mod_skypiax.c ...
 Creating mod_skypiax.so...
 ld: library not found for -lX11
 collect2: ld returned 1 exit status
 gcc -DSKYPIAX_SVN_VERSION=\14471\ -I/Users/imyrvold/Documents/
 Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/Documents/
 Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
 fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
 ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
 o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
 mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/Freeswitch/
 freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
 imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
 expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
 libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/libiconv.dylib /
 Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr/.libs/
 libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
 lncurses -lX11
 make[5]: *** [mod_skypiax.so] Error 1
 make[4]: *** [all] Error 1
 make[3]: *** [mod_skypiax-all] Error 1
 make[2]: *** [all-recursive] Error 1


 Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:

 No, it needs implementation of the message pump between the module and
 the Skype API.

 It's probably kind of trivial, if no other problems I'm not aware of.

 I do not have a Mac to implement it, tough :-(.

 -giovanni





 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 On Thu, Aug 6, 2009 at 5:55 PM, Brian Westbr...@freeswitch.org
 wrote:
 I'm not sure about that one I haven't tried lately because the
 API
 differs on the Mac last I looked at it.

 /b

 On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:

 Is skypiax now working on Mac OS X in Freeswitch?

 Ivan


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Re: [Freeswitch-users] skypiax on Mac OS X

2009-08-09 Thread Giovanni Maruzzelli

Seven,
thanks a lot for your effort, please let your stuff be available,
maybe Ivan can make use of it!

Ivan,
in the file src/mod/endpoints/mod_skypiax/skypiax_protocol.c add you
will find #ifdef WIN32 .

it conditional compiles code between WIN32 and linux.

You need to add another #ifdef, so it will compile for OSX.

You will probably be able to use the same pipe mechanism as in Linux
(normal POSIX pipes).
You will for sure need to implement the part that deals with the Skype
API. Maybe it will be not much more than reusing the example code to
interact with the API.

Please, let us know how it goes, and feel *very* free to ask for further info.

-giovanni




Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvoldi...@myrvold.org wrote:
 Yes, I am interested in this, and if you have any source I could have
 a look at it.

 Ivan

 Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:


 On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
 Ciao Ivan,

 it seems that you do not have the libX11 **development** package
 installed.

 Unfortunately I don't know about OSX, so I cannot help you, but many
 on the list know.
 BTW: it will probably be of no use to you to compile mod_skypiax on
 OSX, because Skype for MACOSX works in another way than Skype for
 Linux.

 That's right.

 If you know about MacOSX programming, please have a look at
 https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
 probably be simple enough to add a message pump for MacOSX.

 -giovanni


 Giovanni, I have a Mac and tried to get this work yesterday, but
 haven't got it work. Will try further if I have time. However, I don't
 think it's so useful because I don't know how to run and hence control
 multi-skype instances on Mac.

 If someone interested to try this I can check the code into my branch.



 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvoldi...@myrvold.org
 wrote:
 I tried to compile mod_skypiax, and am getting problem with X11. On
 OS
 X Leopard, X11 is installed in /usr/X11/lib/
 See below.

 What can I do to get past this error?

 I can also let you ssh into my machine. Contact me off list in case.

 Ivan

 making all mod_skypiax
 Compiling skypiax_protocol.c...
 Compiling mod_skypiax.c...
 mkdir .libs
 Compiling mod_skypiax.c ...
 Creating mod_skypiax.so...
 ld: library not found for -lX11
 collect2: ld returned 1 exit status
 gcc -DSKYPIAX_SVN_VERSION=\14471\ -I/Users/imyrvold/Documents/
 Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
 Documents/
 Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
 fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
 g -
 ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
 shared -
 o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
 mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
 Freeswitch/
 freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
 imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
 expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
 libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
 libiconv.dylib /
 Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
 apr/.libs/
 libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
 lncurses -lX11
 make[5]: *** [mod_skypiax.so] Error 1
 make[4]: *** [all] Error 1
 make[3]: *** [mod_skypiax-all] Error 1
 make[2]: *** [all-recursive] Error 1


 Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:

 No, it needs implementation of the message pump between the module
 and
 the Skype API.

 It's probably kind of trivial, if no other problems I'm not aware
 of.

 I do not have a Mac to implement it, tough :-(.

 -giovanni





 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 On Thu, Aug 6, 2009 at 5:55 PM, Brian Westbr...@freeswitch.org
 wrote:
 I'm not sure about that one I haven't tried lately because the
 API
 differs on the Mac last I looked at it.

 /b

 On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:

 Is skypiax now working on Mac OS X in Freeswitch?

 Ivan


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Re: [Freeswitch-users] skypiax on Mac OS X

2009-08-06 Thread Giovanni Maruzzelli

No, it needs implementation of the message pump between the module and
the Skype API.

It's probably kind of trivial, if no other problems I'm not aware of.

I do not have a Mac to implement it, tough :-(.

-giovanni





Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Thu, Aug 6, 2009 at 5:55 PM, Brian Westbr...@freeswitch.org wrote:
 I'm not sure about that one I haven't tried lately because the API
 differs on the Mac last I looked at it.

 /b

 On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:

 Is skypiax now working on Mac OS X in Freeswitch?

 Ivan


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[Freeswitch-users] freepbx for freeswitch

2009-08-05 Thread Giovanni Maruzzelli

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.

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Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story

2009-08-04 Thread Giovanni Maruzzelli

 is going on is
 key.
 Most importantly, anthm and the core team have been super helpful in getting
 SIP to work with us.  Back in the pre 1.0 days anthm made significant
 changes to mod-sofia to enable clients behind nats without STUN.  Its
 important to point out that he didn't just make the changes -he forced us to
 really make a compelling case as to why the changes were important for
 FreeSWITCH.  This is a good thing.
 skype (mod_skypiax): Due to the facts that users prefer skype, we configured
 skypiax. It was unstable at the beginning and that's one of the reason we
 started running that separate FS instance.  To be fair, it has caused a lot
 of trouble - but we know this, its new software that takes a big risk and
 implements a complex hack.  What is important is that the author of
 skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
 fixing bugs and logging in to our box to help trouble shoot. We owe him a
 *big* thanks.
 To make Skypiax more useful, we also created some patches including the ANY
 and RR interfaces for sequential and round robin line hunting, some bug
 fixes and other features like continue-load-on-fail and auto-skype-user
 which haven't been merged into trunk yet. Thanks a community that gives us a
 platform where we can all benefit and contribute.
 erlang (mod_erlang_events): Another key enabler of the next release of our
 system is the erlang interface.  We have a complex realtime queue routing
 system has it handles input not just from freeswitch, but numerous other web
 interfaces and sockets.  Erlang was the perfect technology to implement this
 in and luckily an Erlang module for FreeSWITCH was already written.
 Beautiful.
 THE MORAL OF THE STORY:
 FreeSWITCH is a great piece of software that has enabled new technologies
 and business models.  The design has allowed (and the core team has
 nurtured) a vibrant and exciting community that has made the software even
 better.  Every day we go to work excited to push the boundaries of what can
 be done with telephony technology and are confident this is the platform of
 the future.
 Thank you all.

 Sincerely,
 Du Jinfang (Seven) - Technical Operations/VoIP Manager
 Jonathan Palley - CTO
 Idapted Ltd.
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Re: [Freeswitch-users] if using centos you should read this

2009-07-31 Thread Giovanni Maruzzelli

:-)!


On Fri, Jul 31, 2009 at 7:36 AM, Muhammad
Shahzadshaherya...@googlemail.com wrote:
 Please read my email as,

 CentOS has been a trusted platfrom for me from last 3+ years. I have
 developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
 boxes, and 27 out of 49 Asterisk box are still running on CentOS in
 production environment. I really wish and hope this great project continues.

 I don't know any of its developers personally but i am quite sure they
 will resolve their differences professionally and put this project back on
 track.

 This damn Google Spell made meaning of my entire post the possite. ;-(

 Thank you.


 On Fri, Jul 31, 2009 at 11:21 AM, Michael Collins m...@freeswitch.org
 wrote:


 On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
 shaherya...@googlemail.com wrote:

 CentOS has been a trusted platfrom for me from last 3+ years. I have
 developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
 boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in
 production environment. I really wish and hope this great project continues.

 I don't know any of its developers personally but i am quite sure they
 will resolve their differences professionally and put this project back on
 track.

 The guys doing the work have vowed to continue the project. The only real
 issues are who controls the centos.org domain name and how to handle
 donations to the project. CentOS isn't going anywhere but forward.
 -MC


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 --
 Muhammad Shahzad
 ---
 CISCO Rich Media Communication Specialist (CRMCS)
 CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
 MSN: shari_78...@hotmail.com
 Email: shaherya...@googlemail.com

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[Freeswitch-users] Slashdot: How-to-Help-With-a-University-ICT-Strategy

2009-07-27 Thread Giovanni Maruzzelli

http://ask.slashdot.org/story/09/07/27/1652247/How-to-Help-With-a-University-ICT-Strategy

 An anonymous reader writes I have been asked to contribute to my
university's revised ICT (Information and Communication Technology)
strategy and I am curious what fellow Slashdot members consider to be
the main advice in this context. What are the major mistakes that
organizations like universities make? Given the complexity of the
different participants in a university, how does one have a coherent
strategy that fulfills the needs of such a wide audience? How does one
promote open source in a managerial culture? How does one deal with
the curse of the virtual learning environment?

http://ask.slashdot.org/comments.pl?sid=1316571cid=28842157

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Re: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash

2009-07-26 Thread Giovanni Maruzzelli

Performance problems and other issues (eg crashes on ALSA drivers) has
been reported for Skypiax on CentOS, albeit various users got good
success on same CentOS. The section down below, Extreme Performances
on Linux solves all problems for the user that got issues on CentOS.

http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#.22Extreme.22_Performances_on_Linux



On Sun, Jul 26, 2009 at 2:37 PM, Giovanni Maruzzelligmar...@celliax.org wrote:
 Ciao Muhammad,

 I've got many problems with ALSA drivers, including various kind of crashes.

 To make a looong story short, use the alsa_drivers version 1.0.20,
 they have not yet crashed on me.

 Also, if you want to test it, you can compile the customized snd-dummy
 driver you find in the svn code, it is a try to have much more
 efficiency bot in softirqs and context switches, allows for 64 Skype
 instances (128 subdevices), etc. it is to be compiled with
 alsa_drivers 1.0.20 too.

 Is my feeling (I mean, almost sure) they got spin_locking wrong in
 previous versions, and it crashes the kernel when you really use it
 (Skype clients have a demented usage of alsa).

 BTW, I'm in the process of revamp the code, fix the bugs and apply
 patches. Please, have a look at the new wiki page with lots of new
 content, I'll send a mail to the ML tomorrow :-)


 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 On Sun, Jul 26, 2009 at 2:19 PM, Muhammad
 Shahzadshaherya...@googlemail.com wrote:
 Hi,

 I am having random Linux Kernel crash problems while running FreeSWITCH as
 Skype to/from SIP gateway on one of our production servers. This machine is
 running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS
 svn revision number 13754.

 At time of Kernel crash i could find following crash messages which point to
 some source code file in FS source tree.

  - Kernel Begin 


  3 Time(s):  ===
  3 Time(s):  [c0404eff] syscall_call+0x7/0xb
  3 Time(s):  [c043ed22] sys_delete_module+0x192/0x1b8
  3 Time(s):  [c0449011] audit_syscall_entry+0x14b/0x17d
  3 Time(s):  [c049f4fe] remove_proc_entry+0x139/0x18c
  3 Time(s):  [f8d96281] alsa_sound_exit+0xa/0x30 [snd]
  3 Time(s):  [f8d96304] snd_info_done+0x46/0x49 [snd]
  3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not
 tainted)
  1 Time(s): snd-malloc: Memory leak?  pages not freed = 1

  -- Kernel End -

 While the problem seems to arise from ALSA kernel module but it blames FS
 file fs/proc/generic.c:732 for this. The only FS module that is using ALSA
 is mod_skypiax but as far as i remember that module is using FS internal
 routines to allocate and de-allocate sound driver services for Skype client.

 Please suggest a solution.

 Thank you.


 --
 Muhammad Shahzad
 ---
 CISCO Rich Media Communication Specialist (CRMCS)
 CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
 MSN: shari_78...@hotmail.com
 Email: shaherya...@googlemail.com

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Re: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash

2009-07-26 Thread Giovanni Maruzzelli

Ciao Muhammad,

I've got many problems with ALSA drivers, including various kind of crashes.

To make a looong story short, use the alsa_drivers version 1.0.20,
they have not yet crashed on me.

Also, if you want to test it, you can compile the customized snd-dummy
driver you find in the svn code, it is a try to have much more
efficiency bot in softirqs and context switches, allows for 64 Skype
instances (128 subdevices), etc. it is to be compiled with
alsa_drivers 1.0.20 too.

Is my feeling (I mean, almost sure) they got spin_locking wrong in
previous versions, and it crashes the kernel when you really use it
(Skype clients have a demented usage of alsa).

BTW, I'm in the process of revamp the code, fix the bugs and apply
patches. Please, have a look at the new wiki page with lots of new
content, I'll send a mail to the ML tomorrow :-)


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Jul 26, 2009 at 2:19 PM, Muhammad
Shahzadshaherya...@googlemail.com wrote:
 Hi,

 I am having random Linux Kernel crash problems while running FreeSWITCH as
 Skype to/from SIP gateway on one of our production servers. This machine is
 running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS
 svn revision number 13754.

 At time of Kernel crash i could find following crash messages which point to
 some source code file in FS source tree.

  - Kernel Begin 


  3 Time(s):  ===
  3 Time(s):  [c0404eff] syscall_call+0x7/0xb
  3 Time(s):  [c043ed22] sys_delete_module+0x192/0x1b8
  3 Time(s):  [c0449011] audit_syscall_entry+0x14b/0x17d
  3 Time(s):  [c049f4fe] remove_proc_entry+0x139/0x18c
  3 Time(s):  [f8d96281] alsa_sound_exit+0xa/0x30 [snd]
  3 Time(s):  [f8d96304] snd_info_done+0x46/0x49 [snd]
  3 Time(s): BUG: warning at fs/proc/generic.c:732/remove_proc_entry() (Not
 tainted)
  1 Time(s): snd-malloc: Memory leak?  pages not freed = 1

  -- Kernel End -

 While the problem seems to arise from ALSA kernel module but it blames FS
 file fs/proc/generic.c:732 for this. The only FS module that is using ALSA
 is mod_skypiax but as far as i remember that module is using FS internal
 routines to allocate and de-allocate sound driver services for Skype client.

 Please suggest a solution.

 Thank you.


 --
 Muhammad Shahzad
 ---
 CISCO Rich Media Communication Specialist (CRMCS)
 CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
 MSN: shari_78...@hotmail.com
 Email: shaherya...@googlemail.com

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Re: [Freeswitch-users] A possible bug in FS causing Linux Kernel crash

2009-07-26 Thread Giovanni Maruzzelli

On Sun, Jul 26, 2009 at 2:19 PM, Muhammad
Shahzadshaherya...@googlemail.com wrote:
 Hi,

 I am having random Linux Kernel crash problems while running FreeSWITCH as
 Skype to/from SIP gateway on one of our production servers. This machine is
 running CentOS 5.2, Kernel version 2.6.18-92.1.22.el5.centos.plusPAE with FS
 svn revision number 13754.

 While the problem seems to arise from ALSA kernel module but it blames FS
 file fs/proc/generic.c:732 for this. The only FS module that is using ALSA
 is mod_skypiax but as far as i remember that module is using FS internal
 routines to allocate and de-allocate sound driver services for Skype client.

Also, please note that neither mod_skypiax nor FreeSWITCH have nothing
to do with ALSA (eg: no ALSA code at all in mod_skypiax or
FreeSWITCH).

Is the Skype client instance that uses the sound driver, just like on
a desktop Skype client usage

The Skype client instances are started by a shell script, but you
could as well start them from the command line, and are completely
autonomous from FreeSWITCH (FS do not allocate or deallocate sound
driver services for them).

Summary: it's just the ALSA drivers that are to blame :-)

-giovanni

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Re: [Freeswitch-users] Baby Update!

2009-07-03 Thread Giovanni Maruzzelli

Yeeh!


On Fri, Jul 3, 2009 at 10:31 PM, Brian Westbr...@freeswitch.org wrote:
 Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz
 YAY... Congrats mr Lanman!
 /b
 On Jul 3, 2009, at 8:58 AM, David Knell wrote:

 On Fri, 2009-07-03 at 17:43 +0600, Muhammad Shahzad wrote:

 Congratulations to Ray and Samantha. Lets see what new features and

 bug fixes we will get in their new version..! ;-)

 Bug fixes..?!  I'd refer you to Philip Larkin (went to my school, a bit
 before my time, poet, deceased, recently voted Britain's favourite
 poet) whose This Be The Verse suggests otherwise:
 http://www.artofeurope.com/larkin/lar2.htm

 [as a recent father myself, I'm trying to prove him wrong..]

 --Dave

 --
 David Knell, Director, 3C Limited
 T: +44 20 3298 2000
 E: d...@3c.co.uk
 W: http://www.3c.co.uk


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Re: [Freeswitch-users] Segmentation fault with record_session

2009-06-17 Thread Giovanni Maruzzelli

Thanks Jingwei,
have a good night!

-giovanni

On Wed, Jun 17, 2009 at 11:39 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
 Hi Giovanni,

 Sorry, pretty busy and fully occupied by other stuff today. Have to delay
 the testing and give you the result tomorrow.

 Regards,
 -Jingwei

 On Tue, Jun 16, 2009 at 5:32 PM, Jingwei Yang jingwei.y...@gmail.com
 wrote:

 Sure, I'll append to you the result tomorrow.

 Regards,
 -Jingwei

 On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Hi Jingwei,

 Thanks a lot! I'll take care of as soon as possible.

 Btw, before I read the Jira, are you testing in linux?

 If you are testing on linux, would you please report how it is
 performing under load? I mean, what is the load average with, let say,
 10 or 20 or more concurrent Skype call?

 This has nothing to do with your bug, but will help me in getting
 better performances.

 Ciao for now, and thanks again for reporting!

 -giovanni




 On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yangjingwei.y...@gmail.com
 wrote:
  Hi Giovanni,
 
  I've reported it in Jira. Here's the bug url:
 
  http://jira.freeswitch.org/browse/MODSKYPIAX-35
 
  Thanks,
  -Jingwei
 
  On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli
  gmar...@celliax.org
  wrote:
 
  Hi Jingwel,
  thanks for reporting.
 
  Could you please add a Jira issue with as much details as possible?
 
  general guide for reporting bugs:
  http://wiki.freeswitch.org/wiki/Reporting_Bugs
 
  what to add for skypiax:
 
 
  http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests
 
  mod_skypiax Jira:
  http://jira.freeswitch.org/browse/MODSKYPIAX
 
 
  Sincerely,
 
  Giovanni Maruzzelli
  =
  www.celliax.org
  via Pierlombardo 9, 20135 Milano
  Italy
  gmaruzz at celliax dot org
  Cell : +39-347-2665618
  Fax : +39-02-87390039
 
 
 
 
  On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com
  wrote:
   Hi Team,
  
   I've been using the record_session feature to record call sessions.
   Here's
   how I prepared the dialplan:
  
       extension name=skypiax
     condition field=destination_number
   expression=^2909/(.*)$
       action application=record_session data=/tmp/data.wav/
       action application=bridge data=skypiax/ANY/$1/
     /condition
       /extension
  
   And here's how I trigger it:
  
       freeswi...@localhost.localdomainoriginate
   skypiax/skypiax2/userAAA
   2909/userBBB
  
   The call can be established and the data.wav file was generated
   without
   any
   problem. However, once userAAA hung up, a segmentation fault
   occurred
   and
   freeswitch was automatically shut down. Here are what I got from the
   console:
  
   freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA
   2909/userBBB
   2009-06-15 17:25:07 [NOTICE] switch_channel.c:602
   switch_channel_set_name()
   New Channel skypiax/skypiax2/userAAA
   [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b]
   2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270
   remote_party_is_ringing()
   Ring-Ready skypiax/skypiax2/userAAA
   2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333
   outbound_channel_answered()
   Channel [skypiax/skypiax2/userAAA] has been answered
   2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349
   switch_ivr_session_transfer()
   Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default]
   API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output:
   +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b
  
   freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO]
   mod_dialplan_xml.c:252 dialplan_hunt() Processing
   FreeSWITCH-2909/userBBB
   in context default
   2009-06-15 17:25:10 [NOTICE] switch_channel.c:602
   switch_channel_set_name()
   New Channel skypiax/ANY/userBBB
   [4a8b36a4-85d6-4735-98df-dde1a32ac66a]
   2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270
   remote_party_is_ringing()
   Ring-Ready skypiax/ANY/userBBB!
   2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333
   outbound_channel_answered()
   Channel [skypiax/ANY/userBBB] has been answered
   2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680
   skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA
   [CS_EXECUTE]
   [NORMAL_CLEARING]
   2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505
   audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB
   [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
   2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085
   switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA)
   Ended
   2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087
   switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA
   [CS_DESTROY]
   2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085
   switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended
   2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087
   switch_core_session_thread() Close Channel skypiax/ANY/userBBB
   [CS_DESTROY

Re: [Freeswitch-users] Segmentation fault with record_session

2009-06-16 Thread Giovanni Maruzzelli

Hi Jingwei,

Thanks a lot! I'll take care of as soon as possible.

Btw, before I read the Jira, are you testing in linux?

If you are testing on linux, would you please report how it is
performing under load? I mean, what is the load average with, let say,
10 or 20 or more concurrent Skype call?

This has nothing to do with your bug, but will help me in getting
better performances.

Ciao for now, and thanks again for reporting!

-giovanni




On Tue, Jun 16, 2009 at 10:15 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
 Hi Giovanni,

 I've reported it in Jira. Here's the bug url:

 http://jira.freeswitch.org/browse/MODSKYPIAX-35

 Thanks,
 -Jingwei

 On Mon, Jun 15, 2009 at 8:16 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Hi Jingwel,
 thanks for reporting.

 Could you please add a Jira issue with as much details as possible?

 general guide for reporting bugs:
 http://wiki.freeswitch.org/wiki/Reporting_Bugs

 what to add for skypiax:

 http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests

 mod_skypiax Jira:
 http://jira.freeswitch.org/browse/MODSKYPIAX


 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com
 wrote:
  Hi Team,
 
  I've been using the record_session feature to record call sessions.
  Here's
  how I prepared the dialplan:
 
      extension name=skypiax
    condition field=destination_number expression=^2909/(.*)$
      action application=record_session data=/tmp/data.wav/
      action application=bridge data=skypiax/ANY/$1/
    /condition
      /extension
 
  And here's how I trigger it:
 
      freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA
  2909/userBBB
 
  The call can be established and the data.wav file was generated without
  any
  problem. However, once userAAA hung up, a segmentation fault occurred
  and
  freeswitch was automatically shut down. Here are what I got from the
  console:
 
  freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA
  2909/userBBB
  2009-06-15 17:25:07 [NOTICE] switch_channel.c:602
  switch_channel_set_name()
  New Channel skypiax/skypiax2/userAAA
  [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b]
  2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270
  remote_party_is_ringing()
  Ring-Ready skypiax/skypiax2/userAAA
  2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333
  outbound_channel_answered()
  Channel [skypiax/skypiax2/userAAA] has been answered
  2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349
  switch_ivr_session_transfer()
  Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default]
  API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output:
  +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b
 
  freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO]
  mod_dialplan_xml.c:252 dialplan_hunt() Processing
  FreeSWITCH-2909/userBBB
  in context default
  2009-06-15 17:25:10 [NOTICE] switch_channel.c:602
  switch_channel_set_name()
  New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a]
  2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270
  remote_party_is_ringing()
  Ring-Ready skypiax/ANY/userBBB!
  2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333
  outbound_channel_answered()
  Channel [skypiax/ANY/userBBB] has been answered
  2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680
  skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA
  [CS_EXECUTE]
  [NORMAL_CLEARING]
  2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505
  audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB
  [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
  2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085
  switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended
  2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087
  switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA
  [CS_DESTROY]
  2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085
  switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended
  2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087
  switch_core_session_thread() Close Channel skypiax/ANY/userBBB
  [CS_DESTROY]
  Segmentation fault (core dumped)
 
  Please kindly let me know whether there's anything wrong with the
  dialplan
  or the way how I originated the call.
 
  Thanks!
  -Jingwei
 
 
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Re: [Freeswitch-users] Segmentation fault with record_session

2009-06-15 Thread Giovanni Maruzzelli

Hi Jingwel,
thanks for reporting.

Could you please add a Jira issue with as much details as possible?

general guide for reporting bugs:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

what to add for skypiax:
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#How_To_Report_BUGS_and_Feature_Requests

mod_skypiax Jira:
http://jira.freeswitch.org/browse/MODSKYPIAX


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Mon, Jun 15, 2009 at 11:40 AM, Jingwei Yangjingwei.y...@gmail.com wrote:
 Hi Team,

 I've been using the record_session feature to record call sessions. Here's
 how I prepared the dialplan:

     extension name=skypiax
   condition field=destination_number expression=^2909/(.*)$
     action application=record_session data=/tmp/data.wav/
     action application=bridge data=skypiax/ANY/$1/
   /condition
     /extension

 And here's how I trigger it:

     freeswi...@localhost.localdomainoriginate skypiax/skypiax2/userAAA
 2909/userBBB

 The call can be established and the data.wav file was generated without any
 problem. However, once userAAA hung up, a segmentation fault occurred and
 freeswitch was automatically shut down. Here are what I got from the
 console:

 freeswi...@localhost.localdomain originate skypiax/skypiax2/userAAA
 2909/userBBB
 2009-06-15 17:25:07 [NOTICE] switch_channel.c:602 switch_channel_set_name()
 New Channel skypiax/skypiax2/userAAA [66195ba1-b609-4f7f-b6cf-4a7e79fdf24b]
 2009-06-15 17:25:07 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing()
 Ring-Ready skypiax/skypiax2/userAAA
 2009-06-15 17:25:10 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered()
 Channel [skypiax/skypiax2/userAAA] has been answered
 2009-06-15 17:25:10 [NOTICE] switch_ivr.c:1349 switch_ivr_session_transfer()
 Transfer skypiax/skypiax2/userAAA to XML[2909/user...@default]
 API CALL [originate(skypiax/skypiax2/userAAA 2909/userBBB)] output:
 +OK 66195ba1-b609-4f7f-b6cf-4a7e79fdf24b

 freeswi...@localhost.localdomain 2009-06-15 17:25:10 [INFO]
 mod_dialplan_xml.c:252 dialplan_hunt() Processing FreeSWITCH-2909/userBBB
 in context default
 2009-06-15 17:25:10 [NOTICE] switch_channel.c:602 switch_channel_set_name()
 New Channel skypiax/ANY/userBBB [4a8b36a4-85d6-4735-98df-dde1a32ac66a]
 2009-06-15 17:25:11 [NOTICE] mod_skypiax.c:1270 remote_party_is_ringing()
 Ring-Ready skypiax/ANY/userBBB!
 2009-06-15 17:25:20 [NOTICE] mod_skypiax.c:1333 outbound_channel_answered()
 Channel [skypiax/ANY/userBBB] has been answered
 2009-06-15 17:25:27 [NOTICE] mod_skypiax.c:680
 skypiax_signaling_thread_func() Hangup skypiax/skypiax2/userAAA [CS_EXECUTE]
 [NORMAL_CLEARING]
 2009-06-15 17:25:27 [NOTICE] switch_ivr_bridge.c:505
 audio_bridge_on_exchange_media() Hangup skypiax/ANY/userBBB
 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085
 switch_core_session_thread() Session 1 (skypiax/skypiax2/userAAA) Ended
 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087
 switch_core_session_thread() Close Channel skypiax/skypiax2/userAAA
 [CS_DESTROY]
 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1085
 switch_core_session_thread() Session 2 (skypiax/ANY/userBBB) Ended
 2009-06-15 17:25:27 [NOTICE] switch_core_session.c:1087
 switch_core_session_thread() Close Channel skypiax/ANY/userBBB [CS_DESTROY]
 Segmentation fault (core dumped)

 Please kindly let me know whether there's anything wrong with the dialplan
 or the way how I originated the call.

 Thanks!
 -Jingwei


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Re: [Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems

2009-06-10 Thread Giovanni Maruzzelli

Ciao Muhammad!

What a good news!

Centos is the most stable and performing platform for FS, so I would
really love to test and document on the wiki how to have a stable
centos mod_skypiax installation.

I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE
), and begin to test. In the mean time, do you have any hint, special
procedure, etc you have done for having skypiax working well?

Please, please, please let be in contact! :-)


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Wed, Jun 10, 2009 at 8:33 AM, Muhammad
Shahzadshaherya...@googlemail.com wrote:
 Sorry, i didn't visited the Jira link you mentioned. Now i know the issue
 and I have replied it there.

 Thank you.


 On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Hi all,

 there are problems for mod_skypiax in recent centos,  with more than a
 handful of concurrent Skype calls.

 Probably the problem is ALSA-related.

 Until it is solved, for production please use Ubuntu 8.04 (see below),
 some other Linux distro (and please write here your experience), or
 Windows.

 I modified the wiki page to reflect this (
 http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk )

 If someone with CentOS knowledge can chime in I'll be grateful :-).

 Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for
 all infos, and feel free to contact me directly.

 -giovanni







 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039

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 ---
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 CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
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Re: [Freeswitch-users] [Freeswitch-dev] mod_skypiax (Skype Endpoint and Trunk) on centos problems

2009-06-10 Thread Giovanni Maruzzelli

Thanks a lot Muhammad,

and please let that firm know the advantages of having customizations
included into mainstream ;-).

OK, I will try 32 bit too, and see if there is differences.

So, you started fom a fresh install of centos5.2, then you installed
the PAE kernel. Is this right?

Pay attention, because if you do an yum update now, it will install
the 128 kernel, no more the 92, and maybe this will break
something.

Anyway, I'm investigating, and please let me know if you'll have
additional infos.

Hope to hear from you soon,

-giovanni

On Wed, Jun 10, 2009 at 12:47 PM, Muhammad
Shahzadshaherya...@googlemail.com wrote:
 I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable
 kernel. I have heard 64bit ALSA drivers have bad sound issues, but never
 used it personally.

 As for source code of my modifications, i made those change to develop a
 customized commercial solution for large European firm, so i would need
 their permissions to provide you the required official patch. Let me write
 them an offical request for this.

 Thank you.


 On Wed, Jun 10, 2009 at 3:47 PM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 Ciao Muhammad,

 first thanks a lot for sharing your experience and help us in making a
 better software!

 From the name of the kernel, seems that you are using centos5.2 is this
 correct?

 I just tried centos5.3 (64bit) with centosplus kernel, but no luck.

 I'm now installing a centos5.2 (64), I will test it with centosplus
 kernel and with its normal kernel.

 BTW, I would like *really* a lot to have and integrate your addition
 to the code (also if it needs some labor from me, no problem). Would
 you like to send it to me, so I will integrate in the main trunk and
 you don't have no more to maintain it? (so you can develop other cool
 features for mod_skypiax ;-) )?

 -giovanni

 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 On Wed, Jun 10, 2009 at 11:16 AM, Muhammad
 Shahzadshaherya...@googlemail.com wrote:
  Thanks. I didn't make any special arrangements for FS or Skypiax to work
  on
  CentOS 5.3. I only enabled CentOS Plus yum repository and then install
  PAE
  kernel with following commands,
 
  root ~# yum update
  root ~# yum install kernel-PAE
 
  i installed PAE kernel just because i wanted to increase System RAM to
  8GB
  before i deploy it for production use, so i can double or even triple
  Skypiax channels whenever i need so, without system or FS shutdown.
 
  As far as a skypiax configuration is concerned, i did modified
  mod_skypiax.c
  to add a couple of commands to dynamically add and remove Skypiax
  interfaces
  in a running FS process. However, this code does not replaces or changes
  any
  previous code. Other then that there is no significant change in
  configuration steps. Though i did use mod_xml_curl to dynamically update
  skypiax interface configuration in FS.
 
 
  Thank you.
 
 
  On Wed, Jun 10, 2009 at 2:37 PM, Giovanni Maruzzelli
  gmar...@celliax.org
  wrote:
 
  Ciao Muhammad!
 
  What a good news!
 
  Centos is the most stable and performing platform for FS, so I would
  really love to test and document on the wiki how to have a stable
  centos mod_skypiax installation.
 
  I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE
  ), and begin to test. In the mean time, do you have any hint, special
  procedure, etc you have done for having skypiax working well?
 
  Please, please, please let be in contact! :-)
 
 
  Sincerely,
 
  Giovanni Maruzzelli
  =
  www.celliax.org
  via Pierlombardo 9, 20135 Milano
  Italy
  gmaruzz at celliax dot org
  Cell : +39-347-2665618
  Fax : +39-02-87390039
 
 
 
 
  On Wed, Jun 10, 2009 at 8:33 AM, Muhammad
  Shahzadshaherya...@googlemail.com wrote:
   Sorry, i didn't visited the Jira link you mentioned. Now i know the
   issue
   and I have replied it there.
  
   Thank you.
  
  
   On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli
   gmar...@celliax.org
   wrote:
  
   Hi all,
  
   there are problems for mod_skypiax in recent centos,  with more than
   a
   handful of concurrent Skype calls.
  
   Probably the problem is ALSA-related.
  
   Until it is solved, for production please use Ubuntu 8.04 (see
   below),
   some other Linux distro (and please write here your experience), or
   Windows.
  
   I modified the wiki page to reflect this (
   http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk )
  
   If someone with CentOS knowledge can chime in I'll be grateful :-).
  
   Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for
   all infos, and feel free to contact me directly.
  
   -giovanni
  
  
  
  
  
  
  
   Sincerely,
  
   Giovanni Maruzzelli
   =
   www.celliax.org
   via

[Freeswitch-users] broken compilation on windows?

2009-06-09 Thread Giovanni Maruzzelli

Hi all,

I cannot compile on Windows the current svn, 13722.

The first error it gives is:
freeswitch\libs\pcre\pcre_internal.h(368) : fatal error C1189: #error
:  LINK_SIZE must be either 2, 3, or 4

then it fails 81 projects (42 succeeded), because no
freeswitchcore.lib (obviously)

I tried both the Freeswitch.2008.sln and the
freeswitch.express.2008.sln, I'm using VC Express 2008.

-giovanni


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

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[Freeswitch-users] mod_skypiax (Skype Endpoint and Trunk) on centos problems

2009-06-09 Thread Giovanni Maruzzelli

Hi all,

there are problems for mod_skypiax in recent centos,  with more than a
handful of concurrent Skype calls.

Probably the problem is ALSA-related.

Until it is solved, for production please use Ubuntu 8.04 (see below),
some other Linux distro (and please write here your experience), or
Windows.

I modified the wiki page to reflect this (
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk )

If someone with CentOS knowledge can chime in I'll be grateful :-).

Please see Jira: http://jira.freeswitch.org/browse/MODSKYPIAX-34 for
all infos, and feel free to contact me directly.

-giovanni







Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

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Re: [Freeswitch-users] (no subject)

2009-06-06 Thread Giovanni Maruzzelli

I agree!


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sat, Jun 6, 2009 at 7:23 AM, Mitul Limbani mi...@enterux.com wrote:
 Ttrfrtttgteruoywtklou

 Regards,juuyuuu
 Mitul Limbani,
 Founder 
 CEO,
 iuokljkknnvvfcxzasqwwhjhyljljjifkkkljjyjjjkkjllgjjggllyjkljkokjkjjjujkmktdswwdsflyjhhbhh
 mlkkkjjjhhhjykvytyyp
 Enterux Solutions Pvt Ltd,bu. B.  P
 The Enterprise Linux Company(r),
 http://www.enterux.com/i
 Pio

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Re: [Freeswitch-users] Documentation error?

2009-05-21 Thread Giovanni Maruzzelli

fixed



On Thu, May 21, 2009 at 6:31 PM, Brian West br...@freeswitch.org wrote:
 Its an error on the wiki you should have $${domain} in there
 /b
 On May 21, 2009, at 11:22 AM, Larry Marshall wrote:

 On the page http://wiki.freeswitch.org/wiki/Configuring_SIP under
 Configuration, it speaks about the vars.xml file. Specifically it states,
 “In this file, there is only one parameter that you need to specify. That
 parameter is $${sip_profile}.”

 I can’t find the variable, nor can I grep for its assignment in conf. Am I
 missing something?

 Lars


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Re: [Freeswitch-users] [Freeswitch-dev] Interesting Blog About HD Telephony

2009-05-06 Thread Giovanni Maruzzelli

On Wed, May 6, 2009 at 8:23 PM, Michael Collins m...@freeswitch.org wrote:
 FYI,
 I made a comment on Dave's blog extolling the virtues of FS and I mentioned
 Skype support. I didn't specifically mention mod_skypiax but I didn't
 specifically mention any mods.


blushI was suggesting to put mod_skypiax in the
http://www.freeswitch.org/node/182 page, for ourselves /blush

BTW: Very nice comment, it sure will attract attention!

-gm

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Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error

2009-05-01 Thread Giovanni Maruzzelli

...@sipgate.co.uk) Ended
 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1081
 switch_core_session_thread() Close Channel sofia/external/
 07771236...@sipgate.co.uk [CS_DESTROY]
 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559
 switch_core_session_destroy_state() (sofia/external/
 07771236...@sipgate.co.uk) State DESTROY
 2009-04-30 17:52:54 [DEBUG] mod_sofia.c:240 sofia_on_destroy()
 sofia/external/07771236...@sipgate.co.uk SOFIA DESTROY
 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:60
 switch_core_standard_on_destroy()
 sofia/external/07771236...@sipgate.co.ukstandard DESTROY
 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:559
 switch_core_session_destroy_state() (sofia/external/
 07771236...@sipgate.co.uk) State DESTROY going to sleep
 --
 Neu: GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate +
 Telefonanschluss für nur 17,95 Euro/mtl.!*
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 --
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 AIM: anthm
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 pstn:213-799-1400


-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

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Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error

2009-05-01 Thread Giovanni Maruzzelli

Gruss Phil,

actually it was shooting in the dark from my side, because I not yet
tested centos5.3, only centos5.2

As soon as I test it out I'll be back to you.
Thanks for filing the Jira.

-giovanni


On Fri, May 1, 2009 at 1:19 PM,  can_...@gmx.de wrote:
 Ciao Giovanni,

 grazie per la tua risposta. Removing 'hdmi' did make some changes, but it
 still doesn't work. I have filed a jira:

 http://jira.freeswitch.org/browse/MODSKYPIAX-33

 Buon primo maggio anche a te,
 Phil

  Original-Nachricht 
 Datum: Fri, 1 May 2009 08:20:10 +0200
 Von: Giovanni Maruzzelli gmar...@celliax.org
 An: freeswitch-users@lists.freeswitch.org
 Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O   
  error

 Have a happy MayDay!

 I cannot see the whole mail now, it's clipped for my mobile, but it
 seems the nth bizarry of new alsa config file, that creates an hdmi
 device even if you do not have one. Try to edit
 /usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and
 delete any mention of 'hdmi'.
 If this do not works, please file a jira or write again.
 Giovanni



 On 5/1/09, Anthony Minessale anthony.miness...@gmail.com wrote:
  if you put that info in a jira ticket
 
  http://jira.freeswitch.org
 
  and route it to skypeiax , the guy who maintains that module will see
 it.
 
 
  On Thu, Apr 30, 2009 at 5:37 PM, can_...@gmx.de wrote:
 
 
  Hello,
 
  I am trying to get skypiax working, but I am having trouble with the
  sound.
  The calls fail with CALL FAILUREREASON 7 = Sound I/O error and
  I am getting the following error:
 
         ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM
  cards.pcm.hdmi
 
 
  I am running centos 5.3 and have followed the installation guide on the
  wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When
  saving
  the configuration on my desktop I have set the sound card to snd_dummy.
 On
  the server the startup script load snd-dumy like this /sbin/modprobe
  snd-dummy enable=1.
  Below is the output of lsmod and the debug output from FS. It would be
  great if someone could help me fix my problem.
 
  Thank you very much.
  Best wishes,
  Phil
 
 
 
 
  -bash-3.2# lsmod
  Module                  Size  Used by
  snd_dummy              12416  0
  snd_seq_oss            32832  0
  snd_seq_midi_event      7744  1 snd_seq_oss
  snd_seq                55200  4 snd_seq_oss,snd_seq_midi_event
  snd_seq_device          7120  1 snd_seq_oss
  snd_pcm_oss            44480  0
  snd_mixer_oss          16512  1 snd_pcm_oss
  snd_pcm                79624  2 snd_dummy,snd_pcm_oss
  snd_timer              22088  2 snd_seq,snd_pcm
  snd                    55976  8
 
 snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
  soundcore               7456  1 snd
  snd_page_alloc          8720  1 snd_pcm
 
 
 
  freeswi...@voipserverserverfreeswitch load mod_skypiax
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:718 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  718  ][none      ][-1,-1,-1]
  globals.debug=0
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:720 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  720  ][none      ][-1,-1,-1]
  globals.debug=8
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:731 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  731  ][none      ][-1,-1,-1]
  codec-master
  globals.debug=8
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:734 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  734  ][none      ][-1,-1,-1]
  globals.dialplan=XML
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:740 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  740  ][none      ][-1,-1,-1]
  globals.context=default
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:743 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  743  ][none      ][-1,-1,-1]
  globals.codec_string=gsm,ulaw
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:750 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  750  ][none      ][-1,-1,-1]
  globals.codec_rates_string=8000,16000
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:723 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  723  ][none      ][-1,-1,-1]
  globals.hold_music=
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:737 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  737  ][none      ][-1,-1,-1]
  globals.destination=5000
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:847 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  847  ][none      ][-1,-1,-1]
  interface_id=1
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:870 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  870  ][none      ][-1,-1,-1]
  name=skypiax1
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:876 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  876  ][none      ][-1,-1,-1]
 Initialized
  XInitThreads!
  2009-04-30 17:47:35 [DEBUG] mod_skypiax.c:897 load_config() rev
  13177[(nil)|37     ][DEBUG_SKYPE  897  ][skypiax1  ][-1, 0, 0]
 CONFIGURING
  interface_id=1
  2009-04-30 17:47:35 [DEBUG

Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O error

2009-05-01 Thread Giovanni Maruzzelli

Hi Phil,

I had to close the Jira, please try again with your original
alsa.conf. Your editing of it was probably causing some of the new
problems.

I just tested it all in a virtual machine (using virtualbox) and it
worked for me.

Only things that comes at my mind is that I used the 32bit, not the
64bit version.

You are using 64bit in a Xen environment (if I understood correctly),
but others have done it with success (btw, various deployment in
Amazon ec2).

The error you was receiving in the original post (ERROR 7) is the
Skype client not finding the sound device.

Maybe is just a problem of permissions? The user the Skype client
instance is started as has permission to read/write on the sound
device?

Have you tried it starting Skype instance as root user?

In my test deployment here, ls -l /dev/snd/* shows that the devices
are r/w only by root...
Change the permission of the devices if you start Skype as another user.

chmod -R a+rw /dev/snd

So, please go back to the original alsa.conf ( I will mail it to your
address), then be sure to follow all the steps.

Then, as a first test, try a call to echo123 that is the test call
answering machine made available by Skype.

Let me know.

-giovanni

On Fri, May 1, 2009 at 9:25 PM, Giovanni Maruzzelli gmar...@celliax.org wrote:
Hi Phil,

I just tried all the steps (exactly, just cut and paste) from the wiki page:
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS_5.2.2C_from_scratch

I substituted 5.3 instead of 5.2.

I'm afraid it worked flawlessly for me. (shocked about: Anthony is
right about CentOS being boring and predictable, good qualities for
a server OS!)

At the start of Skype clients it will tell bizarre things about hdmi,
but they are unharmful (I've not edited the alsa stuff, it still groak
about non-existent hdmi, but it works nonetheless).

So, I suspect your problems have some other cause.

Now I go read the Jira and the attached files, and I hope to be more of help.

-giovanni

On Fri, May 1, 2009 at 4:14 PM, Giovanni Maruzzelli gmar...@celliax.org
wrote:
Gruss Phil,

actually it was shooting in the dark from my side, because I not yet
tested centos5.3, only centos5.2

As soon as I test it out I'll be back to you.
Thanks for filing the Jira.

-giovanni

On Fri, May 1, 2009 at 1:19 PM, can_...@gmx.de wrote:
Ciao Giovanni,

grazie per la tua risposta. Removing 'hdmi' did make some changes, but it
still doesn't work. I have filed a jira:

http://jira.freeswitch.org/browse/MODSKYPIAX-33

Buon primo maggio anche a te,
Phil

Original-Nachricht
Datum: Fri, 1 May 2009 08:20:10 +0200
Von: Giovanni Maruzzelli gmar...@celliax.org
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] skypiax - CALL FAILUREREASON 7 = Sound I/O
error

Have a happy MayDay!

I cannot see the whole mail now, it's clipped for my mobile, but it
seems the nth bizarry of new alsa config file, that creates an hdmi
device even if you do not have one. Try to edit
/usr/share/alsa/alsa.conf or any other file in /usr/share/alsa dir and
delete any mention of 'hdmi'.
If this do not works, please file a jira or write again.
Giovanni

On 5/1/09, Anthony Minessale anthony.miness...@gmail.com wrote:
if you put that info in a jira ticket

http://jira.freeswitch.org

and route it to skypeiax , the guy who maintains that module will see
it.

On Thu, Apr 30, 2009 at 5:37 PM, can_...@gmx.de wrote:

Hello,

I am trying to get skypiax working, but I am having trouble with the
sound.
The calls fail with CALL FAILUREREASON 7 = Sound I/O error and
I am getting the following error:

ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM
cards.pcm.hdmi

I am running centos 5.3 and have followed the installation guide on the
wiki. CaptureDevice, RingDevice and SoundDevice are all set to 2. When
saving
the configuration on my desktop I have set the sound card to snd_dummy.
On
the server the startup script load snd-dumy like this /sbin/modprobe
snd-dummy enable=1.
Below is the output of lsmod and the debug output from FS. It would be
great if someone could help me fix my problem.

Thank you very much.
Best wishes,
Phil

-bash-3.2# lsmod
Module Size Used by
snd_dummy 12416 0
snd_seq_oss 32832 0
snd_seq_midi_event 7744 1 snd_seq_oss
snd_seq 55200 4 snd_seq_oss,snd_seq_midi_event
snd_seq_device 7120 1 snd_seq_oss
snd_pcm_oss 44480 0
snd_mixer_oss 16512 1 snd_pcm_oss
snd_pcm 79624 2 snd_dummy,snd_pcm_oss
snd_timer 22088 2 snd_seq,snd_pcm
snd 55976 8

snd_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
soundcore 7456 1 snd
snd_page_alloc

Re: [Freeswitch-users] Ideas for my presentation

2009-04-19 Thread Giovanni Maruzzelli

we want slides!
we want slides!
we want slides!

:-)
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Re: [Freeswitch-users] Skypiax as a windows service

2009-04-19 Thread Giovanni Maruzzelli

I tried it on Vista, and it works.

The trick for me (on Vista) is to use the same local system account
for both the Skype and the FS services, and *NOT* to use a personal
account. Go figure... :-)

BTW: I tried to use the *.bat you can find in mod_skypiax/configs/
renamed as *.cmd for starting multiple Skype client instances with a
single service, and it works.


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Apr 19, 2009 at 2:40 PM, UV u...@yuvalhertzog.com wrote:
 The service creation steps you described are identical to what we've done.
 The only difference, as having the Skype service running on local system
 doesn't seem to work on Win2K3 server...
 Maybe this works on XP but on Win2K3 it behaves as if it doesn't find the
 audio devices.
 It does work on a user (such as administrator) account.

 We tried it on few servers.
 Any insight?

 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni
 Maruzzelli
 Sent: Saturday, April 18, 2009 8:11 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Skypiax as a windows service

 On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
 gmar...@celliax.org wrote:
 Then you install FS as service (freeswitch.exe -install servicename),
 start FS as a service (under local system), manually (again, from
 the services applet).

 make sure the FS service is owned by local system and that Access
 desktop is ticked.


 gm

 On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
 gmar...@celliax.org wrote:
 On Fri, Apr 17, 2009 at 4:02 PM, UV u...@yuvalhertzog.com wrote:
 Give a shout if you get Skypiax working as a service.
 I'll be happy to contribute to its wiki about it once you get it
 working.

 shoutgot Skypiax working as a service/shout

 I will document this better in the future, but following is the
 general idea, from a Vista Home machine:

 I assume you have FS configured and working with mod_skypiax (if run
 from the command line). I mean, first you have to make sure all is
 working as a normal non-service application, as documented in the
 wiki here http://wiki.freeswitch.org/wiki/Skypiax#Skypiax_on_Windows
 and in the video here
 http://wiki.freeswitch.org/wiki/Skypiax#Windows_Video_How_To

 To start the Skype client instances as services, you need to use
 instsrv and srvany from Windows Server 2003 Resource Kit Tools:

 http://www.microsoft.com/downloads/details.aspx?FamilyID=9D467A69-57FF-4AE7-
 96EE-B18C4790CFFDdisplaylang=en

 Procedure for creating a service is detailed here:
 http://support.microsoft.com/kb/137890 (or more shortly here:

 http://www.sixxs.net/wiki/Configuring_Windows_Vista#.2816.29__Installing_AIC
 CU_Utility_as_a_Service
 )

 You create an (empty) service with those tools, then you follow the
 procedure steps and as Parameters -Application you put the string
 C:\Program Files\Skype\Phone\Skype.exe /secondary /username:skypiax1
 /password:xxx

 *use your username and password in the string*

 Then, from the services applet in Control Center -administrative
 tools, you make sure the service is owned by local system and that
 Access desktop is ticked.

 Start the service manually from the services applet.
 Maybe it will appear a the service wants to access the desktop. Go
 to show message to see what Skype wants, and give some
 configurations if needed.

 Then you install FS as service (freeswitch.exe -install servicename),
 start FS as a service (under local system), manually (again, from
 the services applet).
 It will appear the service wants to access the desktop. Go there and
 give Skype authorization to be connected by FS, forever.

 Stop both services.

 Restart both services, manually. First the Skype clients,  then after
 a while, FS.

 From another machine, make a Skype call to FS.

 If all works as expected, stop both services, make sure (via
 services applet) the FS service will retry three times to start,
 with a minute pause (just to allow for the Skype clients to start and
 settle their connection with the network, to be on the safe side).

 Make the services to start automatic.

 Reboot the machine, don't log in, make another test call to FS via
 Skype, and... shout :-)

 PS: instead of having the service to start one only instance of skype,
 you can probably make the service to start a .CMD file that will start
 many instances, a la startskype.bat

 I'll look into this soon.


 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039


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Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 12:30 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
 are you planning on just signaling on TCP or both audio and signalling
 cos realtime audio over TCP kinda stinks.

 you may find that just running FS as the farm and calling to it with sip is
 more or less the same idea with no work ;)


Hi Anthony,
yes, TCP is not the best for audio. But it's the only way to route
audio from/to the Skype client instance. I mean, it's the only way the
Skype client allows you to access its audio streams.

This is the current situation:
1) mod_skypiax use native signaling (Windows messages, or X events) to
interact with the Skype client through the Skype API.
2) one of the Skype API commands allows for telling to the Skype
client: please, use this TCP port for audio in, and that TCP port for
audio out, instead of the soundcard.
3) the TCP ports must be on the local IP interface (127.0.0.1)
4) mod_skypiax and the Skype client(s) exchange audio samples through
TCP on the local machine, while signaling is platform native

I would like to have the Skype client instances on another machine,
for security and stability purposes (I'm not trusting consumer grade
Skype client to run on production main FS server).

That's why I was writing the farming client, for rerouting both the
signaling commands and the audio streams back and forth between two
separate machines.

Now I understand what you wrote: I can use FS itself (with
mod_skypiax) as a farming client, and connect with the main FS via
SIP. So I can achieve the original aim of having a separate machine(s)
with the Skype instances. Obviously, if that's a requirements, I can
optimize the footprint of the farming client FS loading only the
modules needed for SIP-Skype interaction.

Thanks a lot Anthony, this cuts the Gordian knot and spare me lots of
pathetic efforts :-)

UV, is this solution practical for you?

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039







 On Thu, Apr 16, 2009 at 10:09 AM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 EG: in the farm out scenario there will be FS talking via TCP to a
 farm client (on local machine or remote). The farm client talks
 with Skype client instances running on the same machine the farm
 client is running on.

 On Thu, Apr 16, 2009 at 1:47 PM, UV u...@yuvalhertzog.com wrote:
  Decoupling the Skyiax from FS will solve the problem as I assume it'll
  use
  TCP/IP (winsock) to interface with FS - therefore, I can run it still on
  the
  same machine but two separate sessions.

 yes, it uses TCP for this. So you would end up with FS (with Skypiax
 module) running on RDP while the Skype client instances are running as
 services, on the same machine (or in different machines). FS will talk
 to Skype client instances via TCP.
 Is this acceptable to you?

 Other question: why not running FS as a service too? If you run FS as
 a service and Skype clients as services, all things would works? Why
 you want to use RDP for? (sorry for the silly questions, I just want
 to understand better).

  However, I think getting the Skypiax
  to work as a service will be more beneficial regardless if it's
  decoupled or
  not.

 What do you mean? I believe that Skypiax (as an FS module) works when
 FS is run as service. Your problem seems to me that you cannot run
 Skype instances under RDP because they cannot access the sound device.
 Is this correct?

 gm

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Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 12:58 AM, UV u...@yuvalhertzog.com wrote:
 Ok, I think I know where's the confusion here. Let me clarify:
 1. FS run beautifully as a service - that's why I assumed it should work.
 2. Skype client runs as a service very well too.
 3. When running FS as a service with Skypiax (hence Skypiax as a service),
 Skypiax doesn't seem to find the SkypeAPI.

Why mod_skypiax do not find the API? I know for sure that other
services can access the API on Skype clients running as services. So
mod_skypiax is encountering some specific problem.

I will explore into this one and I'll be back to you.


 In the Wiki page
 http://wiki.freeswitch.org/wiki/Skypiax#Running_Skypiax_on_Windows_as_a_Serv
 ice it's says that Running Skypiax on Windows as a Service is Not yet
 written therefore I assumed it's a known limitation.

 Are you saying it isn't?

Was just the documentation not yet written, I corrected the wiki
page, now reads: This part of the How To documentation has not yet
been written. Please, feel free to contribute.


 Anyway, the farming solution you suggested should solve the problem - I'd
 assume.

As per the previous Anthony's post, you can use FS itself as a farming solution.

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Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 4:02 PM, UV u...@yuvalhertzog.com wrote:
 Give a shout if you get Skypiax working as a service.
 I'll be happy to contribute to its wiki about it once you get it working.

Yes, definitely!
gm

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Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 4:02 PM, UV u...@yuvalhertzog.com wrote:
Give a shout if you get Skypiax working as a service.
I'll be happy to contribute to its wiki about it once you get it working.

shoutgot Skypiax working as a service/shout

I will document this better in the future, but following is the
general idea, from a Vista Home machine:

I assume you have FS configured and working with mod_skypiax (if run
from the command line). I mean, first you have to make sure all is
working as a normal non-service application, as documented in the
wiki here http://wiki.freeswitch.org/wiki/Skypiax#Skypiax_on_Windows
and in the video here
http://wiki.freeswitch.org/wiki/Skypiax#Windows_Video_How_To

To start the Skype client instances as services, you need to use
instsrv and srvany from Windows Server 2003 Resource Kit Tools:
http://www.microsoft.com/downloads/details.aspx?FamilyID=9D467A69-57FF-4AE7-96EE-B18C4790CFFDdisplaylang=en

Procedure for creating a service is detailed here:
http://support.microsoft.com/kb/137890 (or more shortly here:
http://www.sixxs.net/wiki/Configuring_Windows_Vista#.2816.29__Installing_AICCU_Utility_as_a_Service
)

You create an (empty) service with those tools, then you follow the
procedure steps and as Parameters -Application you put the string
C:\Program Files\Skype\Phone\Skype.exe /secondary /username:skypiax1
/password:xxx

*use your username and password in the string*

Then, from the services applet in Control Center -administrative
tools, you make sure the service is owned by local system and that
Access desktop is ticked.

Start the service manually from the services applet.
Maybe it will appear a the service wants to access the desktop. Go
to show message to see what Skype wants, and give some
configurations if needed.

Then you install FS as service (freeswitch.exe -install servicename),
start FS as a service (under local system), manually (again, from
the services applet).
It will appear the service wants to access the desktop. Go there and
give Skype authorization to be connected by FS, forever.

Stop both services.

Restart both services, manually. First the Skype clients, then after
a while, FS.

From another machine, make a Skype call to FS.

If all works as expected, stop both services, make sure (via
services applet) the FS service will retry three times to start,
with a minute pause (just to allow for the Skype clients to start and
settle their connection with the network, to be on the safe side).

Make the services to start automatic.

Reboot the machine, don't log in, make another test call to FS via
Skype, and... shout :-)

PS: instead of having the service to start one only instance of skype,
you can probably make the service to start a .CMD file that will start
many instances, a la startskype.bat

I'll look into this soon.

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

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Re: [Freeswitch-users] Skypiax as a windows service

On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
gmar...@celliax.org wrote:
Then you install FS as service (freeswitch.exe -install servicename),
start FS as a service (under local system), manually (again, from
the services applet).

make sure the FS service is owned by local system and that Access
desktop is ticked.

On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
gmar...@celliax.org wrote:
On Fri, Apr 17, 2009 at 4:02 PM, UV u...@yuvalhertzog.com wrote:
Give a shout if you get Skypiax working as a service.
I'll be happy to contribute to its wiki about it once you get it working.

shoutgot Skypiax working as a service/shout

I will document this better in the future, but following is the
general idea, from a Vista Home machine:

*use your username and password in the string*

Then, from the services applet in Control Center -administrative
tools, you make sure the service is owned by local system and that
Access desktop is ticked.

Stop both services.

Restart both services, manually. First the Skype clients, then after
a while, FS.

From another machine, make a Skype call to FS.

Make the services to start automatic.

Reboot the machine, don't log in, make another test call to FS via
Skype, and... shout :-)

PS: instead of having the service to start one only instance of skype,
you can probably make the service to start a .CMD file that will start
many instances, a la startskype.bat

I'll look into this soon.

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

Re: [Freeswitch-users] Skypiax as a windows service

2009-04-16 Thread Giovanni Maruzzelli

EG: in the farm out scenario there will be FS talking via TCP to a
farm client (on local machine or remote). The farm client talks
with Skype client instances running on the same machine the farm
client is running on.

On Thu, Apr 16, 2009 at 1:47 PM, UV u...@yuvalhertzog.com wrote:
 Decoupling the Skyiax from FS will solve the problem as I assume it'll use
 TCP/IP (winsock) to interface with FS - therefore, I can run it still on the
 same machine but two separate sessions.

yes, it uses TCP for this. So you would end up with FS (with Skypiax
module) running on RDP while the Skype client instances are running as
services, on the same machine (or in different machines). FS will talk
to Skype client instances via TCP.
Is this acceptable to you?

Other question: why not running FS as a service too? If you run FS as
a service and Skype clients as services, all things would works? Why
you want to use RDP for? (sorry for the silly questions, I just want
to understand better).

 However, I think getting the Skypiax
 to work as a service will be more beneficial regardless if it's decoupled or
 not.

What do you mean? I believe that Skypiax (as an FS module) works when
FS is run as service. Your problem seems to me that you cannot run
Skype instances under RDP because they cannot access the sound device.
Is this correct?

gm

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Re: [Freeswitch-users] skypiax Round Robin interface

2009-04-14 Thread Giovanni Maruzzelli

Hi Seven,

thanks a lot for the patch and all the Skypiax action.

I'm just back from Eastern vacations, let me clear the backlog and
I'll be back on this in a couple days.

Thanks again!
gm


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Fri, Apr 10, 2009 at 8:38 PM, dujinfang dujinf...@gmail.com wrote:
 Hi,

 I made a patch, so skypiax is possible to do a RR hunt besides the
 sequential interface ANY.

 Usage:

 originate skypiax/RR/other_skype_name
 sk list

 http://jira.freeswitch.org/browse/MODENDP-211

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Re: [Freeswitch-users] Skypiax as a windows service

2009-04-14 Thread Giovanni Maruzzelli

Hi UV,

seems a difficult one this one.

I have no much experience in RDP/terminal server.

If there is no way to have (or fake) audio driver on RDP/terminal
server apps, probably the Skype clients will not works (as you
experienced).

I'm sure, I've read it (:-) ), that Skype clients can be run on a
Windows machine as services, without any user logged in.

That is what I would explore in the future, just adding the How To to
the wiki page.

What you are experiencing seems to be different, seems to be specific
to the RDP/terminal server usage. I'm I understanding you correctly
(that this is specific to RDP)?

Can you send me more info/hints?

In parallel, I'm slowly working on a way to farm out the Skype clients
from the FS servers, so to have the Skype clients running on different
machines on the same LAN. I've a proof of concept working on Linux for
one channel.

You think this would solve your problems (having the Skype clients
running on separate machines other than the machines running FS)?

I'm just back from Easter vacations, please allow a couple days for
the accumulated backlog ;-)

Thanks a lot for taking the time to explore Skypiax and report this,
gm


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Mon, Apr 13, 2009 at 1:32 PM, UV u...@yuvalhertzog.com wrote:
 Great work on Skypiax, Giovanni.



 We’ve tested it in our lab for sometime and it works very well.

 Unfortunately, when we tried deploying it on a production environment
 (running Win2K3 server farm), we ran into a barrier:

 FS is running as terminal server console application (to be easily
 maintained remotely by RDP)
 This is because Win2K does not allow RDP to access system console (session
 /userid 0)
 Skype does not work on terminal server due to a well known disappearing
 audio drivers problem, therefore it has to run either as a console or a
 service (both on session 0).
 FS can run well as a windows service
 Skypiax seem to load as service, but it can’t find the skype client and exit
 with the following error:

 2009-04-13 20:54:14 [ERR] mod_skypiax.c:990 load_config() rev
 13006M[|37 ][ERRORA  990  ][skype_user    ][-1, 0, 0] Failed to
 connect to a SKYPE API for interface_id=1, no SKYPE client running, please
 (re)start Skype client. Skypiax exiting



 This situation prevents me to run skypiax in production.



 I understand from the wiki page that windows service is not done yet – so I
 presume this is a predicted outcome.



 Any idea when and if this is planned to be implemented?



 Keep up the good work!



 Cheers,

 UV



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Re: [Freeswitch-users] How to design my project ?

2009-04-08 Thread Giovanni Maruzzelli

Ciao Michele,

as a start is definitely better (and more gratifying) that you runs FreeSWITCH.

Then, if (and only if) there is a compelling reason that justify the
amount of time needed to develop a standalone application, go for it.


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Wed, Apr 8, 2009 at 12:34 PM, Michele M mchl...@gmail.com wrote:
 Hi there,

 I'm quite a newbie about freeswitch. I have an  application  (IVR) that
 needs to have endpoints SIP to register,answer the calls and transfer them
 to the right phones.(I( have my own SIP server).Moreover it needs also a
 ASR/TTS API' set  to communicate with my ASR/TTS engine ( just for example
 let's assume it is Cepstral). I'd wouldn't want to have freeswitch running
 and communicate with it to accomplish that but just to use the libfreeswitch
 library embedded. As I don't know that much about freeswitch can it be done?
 or just I need to have freeswitch running as a must? Can somebody point me
 to the right place where to find example of using library embedded (best
 examples for what I'm trying to do) as I have not found that many?

 Thanks in advance

 Miki




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Re: [Freeswitch-users] Skype interaction commands on skypiax

2009-04-07 Thread Giovanni Maruzzelli

svn commit -mskypiax: modified configs/startskype.sh to specify which
unix user will start the Skype client instance. Thx to
mbrancale...@voismart.it
Sendingconfigs/startskype.sh
Transmitting file data .
Committed revision 12937.

:-)

On Tue, Apr 7, 2009 at 10:13 AM, Matteo mbrancale...@voismart.it wrote:
 Ciao Giovanni,

 I suggest to update the startskype.sh script by adding a su username 
 statement,
 in this way:

 instead of starting skype as

 echo myskypeuser xxx | DISPLAY=:101 /usr/bin/skype --pipelogin 

 is better to do:

 su unixusername -c echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype 
 --pipelogin 

 for two reason:
 you can easily put config into a non-root user
 AND
 the startskype.sh will work also if called from init.

 in fact, a plain

 echo myskypeuser xxx | DISPLAY=:101 /usr/bin/skype --pipelogin 

 will not work when called from init script,
 you have to do (even with root)

 su root -c echo 'myskypeuser xxx' | DISPLAY=:101 /usr/bin/skype --pipelogin 
 

 in any other way skype will not get the user home directory...

 This is my 2c experience on centos 5.2.

 regards,
 matteo.

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[Freeswitch-users] Skype interaction commands on skypiax

2009-04-02 Thread Giovanni Maruzzelli

Hi all,

background:
mod_skypiax is Skype compatible endpoint that allows incoming and
outbound calls to/from the Skype network and SkypeOut service. It's
seen by FS like other endpoints, so you can originate from sofia,
bridge to skypiax, and connect the call to a landline number via
SkypeOut service, for eg.
skypiax endpoint use a Skype client to interact with the Skype network
(see the wiki page for more details
http://wiki.freeswitch.org/wiki/Skypiax).

The news are:
now you can send commands to the skype client related to a skyiax
interface, both from the FS command line and programmatically
(socket/API/esl/whatever)
http://wiki.freeswitch.org/wiki/Skypiax#API_and_CLI_Commands

This allow you to use directly the entire power of the Skype API (
https://developer.skype.com/Docs/ApiDoc ), for eg to send chat
messages, interact with the buddy list, etc etc.
Typing console loglevel 9 at the FS command line allows you to see
the Skype API answers from the Skype client instance.

So, in short: you bring loglevel to 9 (so you can see the Skype API
messages going back and forth), you use sk or skypiax to send
Skype API commands to the Skype client instance.

This way you can prototype extensions to the current mod_skypiax, that
can then be implemented in C directly into the mod_skypiax source
code.

Please, let me know of extensions you would like to be integrated into
the mod_skypiax code ;-).


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

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[Freeswitch-users] FS and Skypiax on Windows Video How To

2009-03-31 Thread Giovanni Maruzzelli

Kulwinder Singh contributed this HOW TO: Freeswitch  Skype- OS
Microsoft Windows
Download 118MB HD: http://www.celliax.org/final.avi

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039

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Re: [Freeswitch-users] FS and Skypiax on Windows Video How To

2009-03-31 Thread Giovanni Maruzzelli

Ciao Bipin,
there is both video and audio.

Use vlc (http://www.videolan.org/vlc/) or mplayer
(http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-).

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




2009/3/31 xbipin bi...@xbipin.com:

 hi,

 is it just audio or is it that im having broken codecs so cant view any
 video?

 Regards,
 Bipin



 Giovanni Maruzzelli-3 wrote:

 Kulwinder Singh contributed this HOW TO: Freeswitch  Skype- OS
 Microsoft Windows
 Download 118MB HD: http://www.celliax.org/final.avi

 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039

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 View this message in context: 
 http://www.nabble.com/FS-and-Skypiax-on-Windows-Video-How-To-tp22799792p22800505.html
 Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] FS and Skypiax on Windows Video How To

2009-03-31 Thread Giovanni Maruzzelli

Oooops,

I was not aware you cannot see the video on Windows (I use mplayer and
vlc on windows, and never bother to start windows media player :-) ).

I agree that the best would be youtube or so.

I don't know how to upload video on youtube, and I'll be not in my
office for a week.

Can one of you kind souls upload the video to youtube? It would be s nice!

I'll try to do that when I'm back if nobody steps out.

gm



Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




2009/3/31 Brian West br...@freeswitch.org:
 You shouldn't have to go get anything :P  If you have to spend time to get
 something to watch the video it sometimes isn't a good thing... have you
 tried YouTUBE?
 http://www.perian.org/
 /b
 On Mar 31, 2009, at 3:45 PM, Nik Middleton wrote:

 Worked for me, just needed to add the missing codec for media player


 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
 Giovanni Maruzzelli
 Sent: 31 March 2009 21:09
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] FS and Skypiax on Windows Video How To

 Ciao Bipin,
 there is both video and audio.

 Use vlc (http://www.videolan.org/vlc/) or mplayer
 (http://www.mplayerhq.hu/design7/dload.html), and you'll be ok :-).

 Sincerely,

 Giovanni Maruzzelli
 =
 www.celliax.org
 via Pierlombardo 9, 20135 Milano
 Italy
 gmaruzz at celliax dot org
 Cell : +39-347-2665618
 Fax : +39-02-87390039




 2009/3/31 xbipin bi...@xbipin.com:

 hi,

 is it just audio or is it that im having broken codecs so cant view

 any

 video?

 Regards,

 Bipin



 Giovanni Maruzzelli-3 wrote:

 Kulwinder Singh contributed this HOW TO: Freeswitch  Skype- OS

 Microsoft Windows

 Download 118MB HD: http://www.celliax.org/final.avi

 Sincerely,

 Giovanni Maruzzelli

 =

 www.celliax.org

 via Pierlombardo 9, 20135 Milano

 Italy

 gmaruzz at celliax dot org

 Cell : +39-347-2665618

 Fax : +39-02-87390039

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 View this message in context:

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 22800505.html

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 -- Meet us a ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] live iso image with freeswitch

There is none yet available.

If you have patience, I suspect that one will be out in the next weeks, tough.

Watch the website and the mailing list for announcement.


Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Mon, Mar 30, 2009 at 4:33 PM, xbipin bi...@xbipin.com wrote:

 can any1 tell me where can i find a live cd image with the basic stuff to run
 FS and FS with all it tools installed and WITH A GUI, something like a pbx
 in a flash iso image so windows users like me find it easier to get testing
 with FS as the support for windows SIP proxy or any SIP related tool for
 windows platform is just about nil so i realized FS on windows also wont
 make much sense coz the rest of the developers etc use linux for FS and if i
 simply keep waiting for FS to actually do something productive on windows
 platform then it might take long or forever.

 if any i can provide me a live CD image with just enough tools to run FS to
 its fullest coz till date i have been only using Voipswitch and its time i
 need to implement TLS or any such type of encryption to reach new markets.
 --
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Re: [Freeswitch-users] live iso image with freeswitch

Hi all,

What I would like to stress is:

1) FreeSwitch is working on Windows, natively, without hacks

2) This is a huge advantage for a free software that want to be
*really* popular (eg: be capable of running on an already working
office machine, without dedicated hardware/expertise)

3) This is very important for people that are not hard core, but
just enthusiast, or just wannabe. Why they have to go for a
proprietary solution, maybe cracked?

4) This is very important for people/situation that just cannot afford
another nmachine, or to dedicate a machine

5) Freeswitch is tested on Windows, albeit less than on *nix

6) This gap will be closing as the curve of adoption go further up

7) The efforts toward a GUI are proceeding, and in a multiplatform
way, so they'll be working on Windows too

I'm a *nix guy like you all, but let's bring closer to us, and us
closer to, the vast majority of people/situations in the world.
Especially making mind at the fact that the big effort, building FS
multiplatform since the beginning, has been YET done :-) !

Ah,

Sincerely,

Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




2009/3/30 Michael Collins m...@freeswitch.org:
 On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel bi...@xbipin.com wrote:

 hi,

 i currently live in a country called UAE - united arab emirates and a
 city called Dubai.

 Hehe, Dubai is quite a popular place - even a number of us ignorant
 Americans have heard of it! :) We would love to see FreeSWITCH become more
 popular in Dubai since it is such an important business hub in the Arab
 world. Please keep checking back for updates on the subject of live CDs or
 ISO install images. They'll be ready sooner or later, hopefully sooner. :)

 -MC

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Re: [Freeswitch-users] live iso image with freeswitch

Yes Kristian, please!


On Mon, Mar 30, 2009 at 7:47 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
 The AstLinux ISO with FreeSWITCH is a live cd.

 http://mirror.astlinux.org/freeswitch/

 This one is a little old but I could easily compile a new one...

 2009/3/30 Brian West br...@freeswitch.org:

 On Mar 30, 2009, at 9:33 AM, xbipin wrote:

 can any1 tell me where can i find a live cd image with the basic stuff to
 run
 FS and FS with all it tools installed and WITH A GUI, something like a pbx
 in a flash iso image so windows users like me find it easier to get testing
 with FS as the support for windows SIP proxy or any SIP related tool for
 windows platform is just about nil so i realized FS on windows also wont
 make much sense coz the rest of the developers etc use linux for FS and if i
 simply keep waiting for FS to actually do something productive on windows
 platform then it might take long or forever.

 Well this is a tall order... You know people ask for it.. or shall I say
 demand it... but nobody really steps up to help out at all on the GUI
 requests.
 FreeSWITCH on windows is equally capable minus a couple of things like TLS
 since nobody will actually DO the work required to make it happen.  This
 isn't a buffet where you pull up and demand things be one way or the
 other... this is a community where you start helping.  I would love to see
 more helping and less demanding!

 if any i can provide me a live CD image with just enough tools to run FS to
 its fullest coz till date i have been only using Voipswitch and its time i
 need to implement TLS or any such type of encryption to reach new markets.

 You could follow the linux how to and install CentOS and be done already.
 Brian West
 br...@freeswitch.org
 -- Meet us a ClueCon!  http://www.cluecon.com




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 --
 Kristian Kielhofner
 http://blog.krisk.org
 http://www.submityoursip.com
 http://www.astlinux.org
 http://www.star2star.com

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Re: [Freeswitch-users] live iso image with freeswitch

no one would do that!


On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
 Are you saying they should configure SIP TLS to run on port 443? :)

 2009/3/30 Brian West br...@freeswitch.org:
 Really hard to inspect packets when they run on port 443 and are encrypted
 :P
 /b

 --
 Kristian Kielhofner
 http://blog.krisk.org
 http://www.submityoursip.com
 http://www.astlinux.org
 http://www.star2star.com

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Re: [Freeswitch-users] live iso image with freeswitch