from:"Giovanni Maruzzelli"

"The Revenge of the Sip"

On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks  wrote:
> Haha classic !!!
> Can't wait for the next installment in the series !!
>
> J
>
>
> On 06/10/2009, at 1:02, Anthony Minessale 
> 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 
> wrote:
>>
>> On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola  wrote:
>> > Very nice :)
>> >
>> > On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling 
>> > 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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>> http://www.freeswitch.org
>
>
>
> --
> 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] Fail over

On Mon, Oct 5, 2009 at 8:59 AM, Giovanni Maruzzelli  wrote:
> On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
>  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
 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

On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola  wrote:
> Very nice :)
>
> On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling  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 8:24 AM, srinivasula reddy
 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] 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  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] 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  
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] 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  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  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] FreeSWITCH Weekly Conference Starting, Please Call In!

2009-09-11 Thread Giovanni Maruzzelli

On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins  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] Friday Meeting at 11AM CST

2009-09-11 Thread Giovanni Maruzzelli

On Fri, Sep 11, 2009 at 4:01 PM, Brian West  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] Skypiax false DTMF event

2009-09-10 Thread Giovanni Maruzzelli

On Wed, Sep 9, 2009 at 10:04 PM, Dmitry Bely  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_machi

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 Servat 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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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 Du 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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>
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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] 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 Viola wrote:
> I'm in, very cool =D
>
> Diego
>
> On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins  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] skypiax error

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

-giovanni

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Re: [Freeswitch-users] skypiax error

On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelli 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

:-) 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 Yang 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 
> 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 Yang
>> 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

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 Yang 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] Centos 5.3 vs Ubuntu LTS

Vista!


On Fri, Aug 28, 2009 at 10:00 PM, Michal
Bielicki wrote:
> OpenSolaris :P
>
> Am 28.08.2009 um 21:50 schrieb Alberto Escudero:
>
>> Debian :)
>> -- Stopping junk mailers is good for the environment
>>
>>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>>> te...@voicetechnology.com.br> wrote:
>>>
 Hi folks,What linux distro would you suggest to deploy FS with django:
 Centos 5.3 or Ubuntu LTS?

>>> CentOS 5.3 FTW!
>>> -MC
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> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
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> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. i...@halokwadrat.de
> e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
> Hauptgeschäftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
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> EIngetragen im HRB Warszawa, KRS 153539
>
>
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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 Tom 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  (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 
> 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] Skypiax, Skype endpoint and trunk, robustness patch

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


On Fri, Aug 28, 2009 at 9:41 PM, Giovanni Maruzzelli wrote:
> On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayeb 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

On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayeb 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] 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 Sacherer 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 TP400
> * 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] Hello, and stuff.

On Fri, Aug 28, 2009 at 9:04 PM, Michael Collins wrote:
>
>
> On Fri, Aug 28, 2009 at 11:55 AM, Giovanni Maruzzelli 
> wrote:
>>
>> 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" :-)
>
> Although, in reality, some of us do actually enjoy spending time together.
> :P
> -MC

meh too! ;-)

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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 Tom 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] 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  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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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. Ebbert 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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[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] Freeswitch performance as a redirecting server

2009-08-26 Thread Giovanni Maruzzelli

netbook remix


joking! Server 64bit :-)

-gm



On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga wrote:
> Hi Giovanny,
>
> regarding ubuntu, did you mean 8.04 server or desktop ?
>
>
> On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli 
> wrote:
>>
>> 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 Culjaga
>> 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
>> > 
>> > 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 Culjaga
>> >> 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
>> >> >
>> >>

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 Schreiber 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  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 Talbot
>> 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
>> > 
>> > 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 nu

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

yes, and will support Asterisk and other engines in the future.

On Tue, Aug 25, 2009 at 8:20 PM, Michael Di
Martino wrote:
> Is FreePBX V3 based on Freeswitch?
>
>
>
> Michael DiMartino | Director of IT | Open Access, Inc.
> 115 Bi County Blvd | Farmingdale, NY 11735
> 631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
>
> www.openaccessinc.com
>
>
>
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Carlos
> Talbot
> Sent: Tuesday, August 25, 2009 2:07 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
>
>
>
> 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 
> 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.
>
> Guidi 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  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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>>> http://www.freeswitch.org
>>
>>
>> ___
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>>
>>
>
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>
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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 Talbot 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 
> 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.
>> Guidi 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  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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>
>
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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 Culjaga 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  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  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
>>  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 Culjaga
>>> 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

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 Culjaga 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 
> 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 Culjaga
>> 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
&

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 Culjaga 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.
>
>
>
>
>
>
>
> ___
> FreeS

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.
Guidi 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  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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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>
>
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Re: [Freeswitch-users] Screaming monkeys on ext 5000

2009-08-24 Thread Giovanni Maruzzelli

On Mon, Aug 24, 2009 at 5:31 AM, Brian West 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] Skypiax, Skype endpoint and trunk, robustness patch

2009-08-15 Thread Giovanni Maruzzelli

On Sat, Aug 15, 2009 at 5:41 PM, Seven Du 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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>
>
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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
 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] 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 West 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

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 Myrvold 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 Myrvold
>>> 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
>>>> ma

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 Myrvold 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 West
>> 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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>>
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>
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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 West 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

ules in the FS ecosystem:
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
>  However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what 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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>

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Re: [Freeswitch-users] if using centos you should read this

2009-08-01 Thread Giovanni Maruzzelli

http://linux.slashdot.org/story/09/08/01/1443221/CentOS-Administrator-Reappears

str8edge sends word that Lance Davis, the CentOS project administrator
who had mysteriously gone absent, has now returned and is working with
the development team to get things back on track. From their
announcement: "The CentOS Development team had a routine meeting today
with Lance Davis in attendance. During the meeting a majority of
issues were resolved immediately and a working agreement was reached
with deadlines for remaining unresolved issues. There should be no
impact to any CentOS users going forward. The CentOS project is now in
control of the CentOS.org and CentOS.info domains and owns all
trademarks, materials, and artwork in the CentOS distributions. We
look forward to working with Lance to quickly complete all the agreed
upon issues. More information will follow soon."

On Fri, Jul 31, 2009 at 8:21 AM, Giovanni Maruzzelli wrote:
> :-)!
>
>
> On Fri, Jul 31, 2009 at 7:36 AM, Muhammad
> Shahzad 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 
>> wrote:
>>>
>>>
>>> On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
>>>  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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>>
>>
>

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Re: [Freeswitch-users] if using centos you should read this

2009-07-30 Thread Giovanni Maruzzelli

:-)!


On Fri, Jul 31, 2009 at 7:36 AM, Muhammad
Shahzad 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 
> wrote:
>>
>>
>> On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
>>  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
>>
>>
>> ___
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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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>

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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=1316571&cid=28842157

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[Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped

2009-07-27 Thread Giovanni Maruzzelli

Ciao FreeSWITCHers,

please have a look at the much changed wiki page:
http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and
checkout and test the code in svn.

Much has happened, various bug fixes and features added.

Most notable:
- multiple instances of the same Skype username on Linux (eg: running
20 concurrent channels as "Bob" Skype user)
- adding and removing interfaces on the fly (patch sent by Muhammad Shahzad)
- easier creation of Skype clients configuration directory
- reduced latency
- better robustness
- running as Windows Service
- customized ALSA driver for more devices with less IRQs and context switches
- custom kernel tickless and 100HZ (eg. solves high load problems in
CentOS and in virtualization)
- interactive command line client for prototyping

Also, please note that ALSA drivers version 1.0.20 seems to be much
more stable in our kind of usage (snd-dummy).

Various other enhancements will come, but in the mean time please give
feedback on the current svn code (we want to be robust for the 1.0.4
Release :-) )

See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm !

-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] 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
Shahzad 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] 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
Shahzad 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):  [] syscall_call+0x7/0xb
>  3 Time(s):  [] sys_delete_module+0x192/0x1b8
>  3 Time(s):  [] audit_syscall_entry+0x14b/0x17d
>  3 Time(s):  [] remove_proc_entry+0x139/0x18c
>  3 Time(s):  [] alsa_sound_exit+0xa/0x30 [snd]
>  3 Time(s):  [] 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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>
>

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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 Maruzzelli 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
> Shahzad 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):  [] syscall_call+0x7/0xb
>>  3 Time(s):  [] sys_delete_module+0x192/0x1b8
>>  3 Time(s):  [] audit_syscall_entry+0x14b/0x17d
>>  3 Time(s):  [] remove_proc_entry+0x139/0x18c
>>  3 Time(s):  [] alsa_sound_exit+0xa/0x30 [snd]
>>  3 Time(s):  [] 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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>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

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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 West 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 Yang 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 
> wrote:
>>
>> Sure, I'll append to you the result tomorrow.
>>
>> Regards,
>> -Jingwei
>>
>> On Tue, Jun 16, 2009 at 4:42 PM, Giovanni Maruzzelli 
>> 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 Yang
>>> 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
>>> > 
>>> > 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 Yang
>>> >> wrote:
>>> >> > Hi Team,
>>> >> >
>>> >> > I've been using the record_session feature to record call sessions.
>>> >> > Here's
>>> >> > how I prepared the dialplan:
>>> >> >
>>> >> >     
>>> >> >   >> >> > expression="^2909/(.*)$">
>>> >> >     
>>> >> >     
>>> >> >   
>>> >> >     
>>> >> >
>>> >> > And here's how I trigger it:
>>> >> >
>>> >> >     freeswi...@localhost.localdomain>originate
>>> >> > 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_ri

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 Yang 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 
> 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 Yang
>> wrote:
>> > Hi Team,
>> >
>> > I've been using the record_session feature to record call sessions.
>> > Here's
>> > how I prepared the dialplan:
>> >
>> >     
>> >   
>> >     
>> >     
>> >   
>> >     
>> >
>> > And here's how I trigger it:
>> >
>> >     freeswi...@localhost.localdomain>originate 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

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 Yang wrote:
> Hi Team,
>
> I've been using the record_session feature to record call sessions. Here's
> how I prepared the dialplan:
>
>     
>   
>     
>     
>   
>     
>
> And here's how I trigger it:
>
>     freeswi...@localhost.localdomain>originate 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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>

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

On Wed, Jun 10, 2009 at 3:01 PM, Muhammad
Shahzad wrote:
> I am using this machine extensively for my development projects, including
> Skypiax. Yesterday i gave a presentation to the board of directors of the
> said firm, regarding existing status of my project. They tested the setup
> with 2-3 concurrent SIP -> SKYPIAX and then SKYPIAX to SIP calls without any
> problem. So, i believe this configuration works without any sound issue...!

Until some times ago, I was used to test with 32 bit centos, and I've
not seen problems (I tested with centos5.2).

Now I'm installing 5.3 32 bit, and I'll report on my results

On 64bit, on both centos5.2 and 5.3 I'm seeing problems with more than
6-10 concurrent calls, with less calls it was ok. BTW, it can be it is
a problem with my hard disk (that is seen as an ide instead than as a
SATA). I will check this too.

I'll report back soon on the various issues...


>
> The second machine is my test machine in a remote data center. I didn't
> prepare this machine, however, from SSH console i can see it has following
> specs,
>
> 1. Intel(R) Xeon(R) CPU E5405  @ 2.00GHz with 4 GB of RAM.
> 2. CentOS 5.3 with kernel 2.6.18-92.1.22.el5.centos.plusPAE.
> 3. FS SVN revision Revision ID 13613.
>
> root ~# uname -a
> Linux localhost.localdomain 2.6.18-92.1.22.el5.centos.plusPAE #1 SMP Wed Dec
> 17 11:32:56 EST 2008 i686 i686 i386 GNU/Linux
>
> root ~# cat /etc/issue
> CentOS release 5.3 (Final)
> Kernel \r on an \m
>
>
> Each machine that i use always, get update with yum update command BEFORE i
> do anything else on it.
>
> Hope this info will be helpful for you.
>
> Can you give me step by step procedure of your testing that is producing
> this bad sound result? I would like to perform this test on my both machines
> and see if i get the same results too.
>
> Thank you.
>
>
> On Wed, Jun 10, 2009 at 6:33 PM, Giovanni Maruzzelli 
> wrote:
>>
>> Ciao Muhammad,
>> you're faster than light :-)!
>>
>> the patch will be integrated very soon, I'll let you know when I'm done
>> it.
>>
>> Keep enhancements, patches, bug fixes, etc flowing!
>>
>> thanks again, and thanks to the firm that so quickly understood and
>> authorized you,
>>
>> -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 2:29 PM, Muhammad
>> Shahzad wrote:
>> > I am glad to share the patch to enable dynamic Skypiax interfaces in FS.
>> > Please do note that however, that i started working on it on May 22,
>> > 2009.
>> > So any officaily changes made to mod_skypiax.c since then will not
>> > appear in
>> > it and will be lost if you apply this patch blindly.
>> >
>> > I request Giovanni Maruzzelli to carefully merge this patch in main
>> > stream
>> > code before committing it to FS SVN.
>> >
>> > Thank you.
>> >
>> >
>> > On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad
>> >  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
>> >> 
>> >> 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

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

On Wed, Jun 10, 2009 at 3:10 PM, dujinfang wrote:
> btw, I have changed the start scripts a bit to start X and skype separately,
> glad to share it if someone interested.

I'm interested! :-)

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

Ciao Muhammad,
you're faster than light :-)!

the patch will be integrated very soon, I'll let you know when I'm done it.

Keep enhancements, patches, bug fixes, etc flowing!

thanks again, and thanks to the firm that so quickly understood and
authorized you,

-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 2:29 PM, Muhammad
Shahzad wrote:
> I am glad to share the patch to enable dynamic Skypiax interfaces in FS.
> Please do note that however, that i started working on it on May 22, 2009.
> So any officaily changes made to mod_skypiax.c since then will not appear in
> it and will be lost if you apply this patch blindly.
>
> I request Giovanni Maruzzelli to carefully merge this patch in main stream
> code before committing it to FS SVN.
>
> Thank you.
>
>
> On Wed, Jun 10, 2009 at 4:47 PM, Muhammad Shahzad
>  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 
>> 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
>>> Shahzad 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
>>> > 
>>> > 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
>>&g

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

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
Shahzad 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 
> 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
>> Shahzad 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
>> > 
>> > 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 : +39-02-8739003

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

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
Shahzad 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 
> 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
>> Shahzad 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
>> > 
>> > 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
>> >>
>> >> ___
>> >> Freeswitch-users mailing list
>> >> Freeswitch-users@list

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

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
Shahzad 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 
> 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
>>
>> ___
>> 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
>
>
>
> --
> 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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>
>

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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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[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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Re: [Freeswitch-users] (no subject)

2009-06-05 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  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  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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> -- Meet us at ClueCon!  http://www.cluecon.com
>
>
>
>
>
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Re: [Freeswitch-users] Cluecon 2009 - Hotel Special!

2009-05-12 Thread Giovanni Maruzzelli

yay Michael! Got it! I owe you one!



On Tue, May 12, 2009 at 8:24 PM, Michael Collins  wrote:
> Good news! For those of you who like to get a good deal on a hotel room
> we've got the inside scoop on a special for the Wyndham Chicago. I just did
> a search on expedia.com for a hotel with these specs:
> Hotel only
> Check-in: 8/3/2009, Check-Out: 8/6/2009
> 1 room, 2 adults
> City: Chicago
> Hotel Name: "Wyndham Chicago"
>
> The results were very nice: the non-refundable rate per night is only
> $140!!! Get to expedia.com NOW and get signed up!
>
> We look forward to seeing you in Chicago this August.
> -Michael
> http://www.cluecon.com
>
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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  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.
>

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

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

-gm

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

2009-05-06 Thread Giovanni Maruzzelli

Ciao Michael,
if you like, you can add that using mod_skypiax you have native hd
skype->FS and FS->Skype (no hardware needed) :-)



On Wed, May 6, 2009 at 6:36 PM, Michael Collins  wrote:
> FYI,
>
> Here's a nice story for you all to check out. Please check it out and pass
> it on.
>
> -Michael
>
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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  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  
> 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,   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 
>>>> 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  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,  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 t

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

2009-05-01 Thread Giovanni Maruzzelli

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  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,   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 
>>> 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  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,  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

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,   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 
>> 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  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,  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

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

2009-04-30 Thread Giovanni Maruzzelli

e_change() Send signal sofia/external/
>> 07771236...@sipgate.co.uk [BREAK]
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397
>> switch_core_session_run() (sofia/external/07771236...@sipgate.co.uk)
>> Running State Change CS_REPORTING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609
>> switch_core_session_reporting_state() (sofia/external/
>> 07771236...@sipgate.co.uk) State REPORTING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:502
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State CONSUME_MEDIA
>> going to sleep
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State
>> Change
>> CS_HANGUP
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP
>> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:228 channel_on_hangup() rev
>> 13177[(nil)|37 ][DEBUG_SKYPE  228  ][skypiax1  ][-1, 1,112] hanging up
>> skype call: 455
>> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:1262
>> skypiax_signaling_write() rev 13177[(nil)|37 ][DEBUG_SKYPE  1262
>> ][skypiax1  ][-1, 1,112] SENDING: |||ALTER CALL 455 HANGUP
>> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:235 channel_on_hangup() rev
>> 13177[(nil)|37 ][DEBUG_SKYPE  235  ][skypiax1  ][-1, 1,112]
>> skypiax/skypiax1/xyzTestUK CHANNEL HANGUP
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:46
>> switch_core_standard_on_hangup() skypiax/skypiax1/xyzTestUK Standard
>> HANGUP,
>> cause: NORMAL_CLEARING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:433
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State HANGUP going
>> to
>> sleep
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:475
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change
>> CS_HANGUP -> CS_REPORTING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:927
>> switch_core_session_signal_state_change() Send signal
>> skypiax/skypiax1/xyzTestUK [BREAK]
>> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:300 channel_kill_channel() rev
>> 13177[(nil)|37 ][DEBUG_SKYPE  300  ][skypiax1  ][-1, 1,112]
>> skypiax/skypiax1/xyzTestUK CHANNEL got SWITCH_SIG_BREAK
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:397
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) Running State
>> Change
>> CS_REPORTING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609
>> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State
>> REPORTING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:53
>> switch_core_standard_on_reporting() skypiax/skypiax1/xyzTestUK Standard
>> REPORTING, cause: NORMAL_CLEARING
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:609
>> switch_core_session_reporting_state() (skypiax/skypiax1/xyzTestUK) State
>> REPORTING going to sleep
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:410
>> switch_core_session_run() (skypiax/skypiax1/xyzTestUK) State Change
>> CS_REPORTING -> CS_DESTROY
>> 2009-04-30 17:52:51 [DEBUG] switch_core_session.c:1061
>> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK)
>> Locked,
>> Waiting on external entities
>> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1079
>> switch_core_session_thread() Session 2 (skypiax/skypiax1/xyzTestUK) Ended
>> 2009-04-30 17:52:51 [NOTICE] switch_core_session.c:1081
>> switch_core_session_thread() Close Channel skypiax/skypiax1/xyzTestUK
>> [CS_DESTROY]
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559
>> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State
>> DESTROY
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:60
>> switch_core_standard_on_destroy() skypiax/skypiax1/xyzTestUK Standard
>> DESTROY
>> 2009-04-30 17:52:51 [DEBUG] switch_core_state_machine.c:559
>> switch_core_session_destroy_state() (skypiax/skypiax1/xyzTestUK) State
>> DESTROY going to sleep
>> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:57 skypiax_signaling_read()
>> rev 13177[(nil)|37 ][DEBUG_SKYPE  57   ][skypiax1  ][-1, 1,112]
>> READING:
>> |||ERROR 559 CALL: Action failed|||
>> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:91 skypiax_signaling_read()
>> rev 13177[(nil)|37 ][DEBUG_SKYPE  91   ][skypiax1  ][-1, 1,112] Skype
>> got ERROR: |||ERROR|||
>> 2009-04-30 17:52:51 [DEBUG] skypiax_protocol.c:93 skypiax_signaling_read()
>> rev 13177[(nil)|37 ][DEBUG_SKYPE  93   ][skypiax1  ][-1, 1,110]
>> skype_call now is DOWN
>> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:672
>> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE  672
>>  ][skypiax1  ][-1, 1,110] skype call ended
>> 2009-04-30 17:52:51 [DEBUG] mod_skypiax.c:687
>> skypiax_signaling_thread_func() rev 13177[(nil)|37 ][DEBUG_SKYPE  687
>>  ][skypiax1  ][-1, 1,110] no session
>> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:53
>> switch_core_standard_on_reporting() sofia/external/
>> 07771236...@sipgate.co.uk Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:609
>> switch_core_session_reporting_state() (sofia/external/
>> 07771236...@sipgate.co.uk) State REPORTING going to sleep
>> 2009-04-30 17:52:54 [DEBUG] switch_core_state_machine.c:410
>> switch_core_session_run() (sofia/external/07771236...@sipgate.co.uk) State
>> Change CS_REPORTING -> CS_DESTROY
>> 2009-04-30 17:52:54 [DEBUG] switch_core_session.c:1061
>> switch_core_session_thread() Session 1 (sofia/external/
>> 07771236...@sipgate.co.uk) Locked, Waiting on external entities
>> 2009-04-30 17:52:54 [NOTICE] switch_core_session.c:1079
>> switch_core_session_thread() Session 1 (sofia/external/
>> 07771236...@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.!*
>> http://dslspecial.gmx.de/freedsl-surfflat/?ac=OM.AD.PD003K11308T4569a
>>
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>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com 
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
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> sip:8...@conference.freeswitch.org 
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
=
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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

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  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
>  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
>  wrote:
>>> On Fri, Apr 17, 2009 at 4:02 PM, UV  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.
>>
>> got Skypiax working as a service
>>
>> 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-B18C4790CFFD&displaylang=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 m

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

On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
 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
 wrote:
>> On Fri, Apr 17, 2009 at 4:02 PM, UV  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.
>
> got Skypiax working as a service
>
> 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-B18C4790CFFD&displaylang=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 4:02 PM, UV 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.

got Skypiax working as a service

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-B18C4790CFFD&displaylang=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 4:02 PM, UV  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 4:00 PM, UV  wrote:
> You're suggesting running two instances of FS on same machine, have them
> communicate first over TCP then UDP via SIP over socket (Skype --> Skypiax
> --> FS1 --> FS2) on a production system? Hmmm... nah... :-)
>

That's exactly what the "farming client" was supposed to attain (and
in my initial idea using TCP instead of UDP). :-)

Using FS is so much better for this purpose, instead of a clumsy
"giovanni's made, TCP based, farming client" you get carrier class
with best throughput/whatever :-)

Yes, this schema will be better employed having the Skype stuff on
different machines than the "main" FS servers.

gm


> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni
> Maruzzelli
> Sent: Friday, April 17, 2009 4:40 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Skypiax as a windows service
>
> On Fri, Apr 17, 2009 at 12:30 AM, Anthony Minessale
>  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
> 
>> 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  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).
>

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

On Fri, Apr 17, 2009 at 12:58 AM, UV  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 12:30 AM, Anthony Minessale
 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 
> 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  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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>
>
>
> --
> Anthony Minessale II
>
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Re: [Freeswitch-users] Skypiax as a windows service

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  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] [ANN] Spice Telephony 0.3 - an open source FreeSWITCH/Erlang callcenter