Re: [Freeswitch-users] MacOSX

[Seven Du]

I had wrote a Air based GUI, is it make sense?

http://wiki.freeswitch.org/wiki/FsAir

2009/12/30 Brian West :
> Guessing the biggest issue is I want to create a softphone project using 
> FreeSWITCH as the core of the project... is this something people would be 
> interested in joining?
>
> /b
>
> On Dec 29, 2009, at 3:35 PM, Ivan C Myrvold wrote:
>
>> I am using iSoftPhone, works great with FreeSWITCH.
>>
>> Ivan
>>
>> Den 29. des. 2009 kl. 22.14 skrev Brian West:
>>
>>> Does it only do IAX?  If so we'll need someone to re-write an IAX2 stack 
>>> since the libiax2 from Digium is no longer updated to keep pace with 
>>> Asterisk and is now incompatible.  Which is the main reason we are thinking 
>>> about dropping IAX support unless someone writes a license compatible lib 
>>> or updates and takes over mod_iax aka owns it as their own.
>>>
>>> /b
>>>
>>> On Dec 29, 2009, at 3:08 PM, Martin Joseph wrote:
>>>
 There is one softphone for OSX that doesn't suck.  It's called
 JackenIAX.
>>
>
>
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Re: [Freeswitch-users] How can I join two freeswitch on two servers?

[Seven Du]

I couldn't guess what you want, pastbin your full config and logs and
give more detail of your story perhaps someone can help you.

2009/12/18 yvonne ding :
>
> param name="username" value="1101"
> param name="password" value="1234"
> param name="proxy" value="192.168.129.194:5060"
> param name="register" value="false"
>
>
> Hi,
>
> If I configure data as following, why FS A "1001" call FS B "1003" failed ?
> Thank you!
>
> FS A: 192.168.129.168, DN=1001
> FS B: 192.168.129.194, DN=1003
>
> In FS A add /conf/sip_proifles/external/gwfsa.xml
>  
>    
>
>
>
>
>    
> 
>
> 1101 is configured in FS B /conf/directory/default/1101.xml, FS A don't have
> 1101 number
>
>
>
>
>
> Dan Le wrote:
>>
>> If you want FS server A to be able to call FS server B, you can set up a
>> user account in server B's FS directory configs, and then just treat
>> server
>> B as a normal gateway by adding a gateway definition in server A. That
>> will
>> allow you to route calls to server B from A; to do the reverse, just
>> mirror
>> the configs the other direction.
>>
>> On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz  wrote:
>>
>>>
>>> I like to connect two freeswitch, call each other, communicate and vice
>>> versa.
>>> Can you give me an example for that?
>>>
>>> Thanks
>>> --
>>> View this message in context:
>>> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
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>
>
>
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>
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Re: [Freeswitch-users] sip message logging and analysis

[Seven Du]

I'm using contrib/seven/sip/sip2db.rb

2009/12/18 David Villasmil :
> i agree with christian, though i would use tshark. you can actually
> get the fields you want (method and callid) and store them in a dB.
> then you need to match them with a query. it is simple but Lots of work.
>
> look into -e and -E of tshark separate the fields by ","
>
> have fun!
>
> David
>
> El 18/12/2009, a las 01:27, Kristian Kielhofner   > escribió:
>
>> Frank,
>>
>>  Probably the cleanest (albeit non-FreeSWITCH) way to implement this
>> would be to use OpenSIPS/SER/etc between you and the carrier with the
>> siptrace module.
>>
>>  But that's probably more work than you want.  There's always tcpdump
>> with a decent filter (udp port 5060 and host x.x.x.x) and then
>> something like http://www.badpenguin.co.uk/files/pcap-util2
>>
>>  Both will allow you to search for BYEs and who is sending them.
>>
>>  Also keep in mind that they (or you) may just be dropping the RTP
>> without ever sending a BYE.  Setting the various RTP timeouts in
>> FreeSWITCH can help with that.  You can then look for logs/events (are
>> there any for RTP timeout?) to see who's dropping RTP.
>>
>> On Thu, Dec 17, 2009 at 7:01 PM, Frank @ Impact
>>  wrote:
>>> I bit off topic but…
>>>
>>>
>>>
>>> Using FS to send calls sip to the LD carrier.
>>>
>>>
>>>
>>> Some calls have problems where they drop the call or audio drops or
>>> whatever.
>>>
>>> The carrier’s first response is that we dropped the call.  But thi
>>> s is  a
>>> day later after the trouble has been reported.
>>>
>>>
>>>
>>> I am looking for guidance on how to log all sip message traffic and
>>> then be
>>> able to easily retrieve to find a call and look at what sip
>>> messages really
>>> were being based and by whom.  Maybe store them in a database or
>>> some other
>>> file that might be opened by an analysis tool.
>>>
>>>
>>>
>>> Any suggestions on how to log this information and then what tool
>>> to use for
>>> later analysis?
>>>
>>>
>>>
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>>>
>>
>>
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
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Re: [Freeswitch-users] Sofia performance

[Seven Du]

you can use the same ip with different port

2009/12/13, Yehavi Bourvine :
> Hello,
>
>   In the WIKI page that talks about Freeswitch performance there is a
> sentence:
>
> *libsofia only handles 1 thread per profile, so if that is your bottle neck
> use more profiles*
>
> How can I enable more than one profile on the same interface? Won't they
> colide when using the same IP and port?
>
>  Thanks! __Yehavi:
>

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Re: [Freeswitch-users] fifo caller id

[Seven Du]

I think if you listen to CUSTOM FIFO::INFO you can get
Caller-Caller-ID-Number on event socket.

2009/12/13 Diego Toro 

> Hello,
>
> I want to know how can I get caller id after call is out queue fifo, I read
> about fifo_caller_consumer_import and  fifo_consumer_caller_import
> variables, but i don't know use it.
>
> I appreciate any suggestion
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
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Re: [Freeswitch-users] Zombie Records in core db

[Seven Du]

I also have this problem on a trunk version more than 1000 revisions
behind, so I think the best way is to upgrade to trunk and report this
again if still have problem.

2009/12/8 DJB :
> We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
> traffic.  I normally check the concurrent calls by looking at the number of
> sessions from status command.  However, the number of concurrent calls in FS
> is normally higher than it's supposed to be after we ran traffic for about a
> week.  Thus, I routed the traffic away from the FS and found out from "show
> calls" that there were so many old calls from previous days.  We are running
> a pass-thru traffic in signaling only.  I wonder whether there is a way to
> have those "zombied" records clean up automatically.  Also, what should I do
> to prevent this problem?
> Thank you,
> Dorn B.
>
>
>
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Re: [Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario

[Seven Du]

Thanks, done.

2009/12/7 Michael Jerris :
> Please report bugs to jira.freeswitch.org.
>
> Mike
>
> On Dec 6, 2009, at 11:45 PM, Seven Du wrote:
>
>> Hi,
>>
>> I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
>> to trunk code, no sound after att_xfer.
>>
>> Then I rebuild FS 15807 with a fresh checkout, but still using the old
>> conf/ settings, sound is ok, but there are other problems:
>>
>> A call B, and B att_xfer C
>>
>> 1) origination_cancel_key not working. no even no DTMF log in FS when
>> I press # or any other key, I tried with Zoiper and Snom(on the B leg)
>> 2) when C answers, B immediately hangup, so B has no chance talk to C
>>
>> Could this be a problem? I pasted logs:
>>
>> http://pastebin.freeswitch.org/11417
>
>
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[Freeswitch-users] att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario

[Seven Du]

Hi,

I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
to trunk code, no sound after att_xfer.

Then I rebuild FS 15807 with a fresh checkout, but still using the old
conf/ settings, sound is ok, but there are other problems:

A call B, and B att_xfer C

1) origination_cancel_key not working. no even no DTMF log in FS when
I press # or any other key, I tried with Zoiper and Snom(on the B leg)
2) when C answers, B immediately hangup, so B has no chance talk to C

Could this be a problem? I pasted logs:

http://pastebin.freeswitch.org/11417

Thanks.

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Re: [Freeswitch-users] Generate cdrs

[Seven Du]

2009/12/4 Mouncifbb :
> I don't want to use XML cdr it puts each call on individual files so

It posts to a http server, and fall back to a xml file if server fails

> is it possible to include a JavaScript at the end of dialplan to
> collect info about the session?
>

I think the answer is yes but where would you store the collected info?

> Thanks
>
>
> On Dec 3, 2009, at 7:02 PM, Seven Du  wrote:
>
>> why not try mod_xml_cdr?
>>
>> 2009/12/4 Mouncif Benniane :
>>> is it possible to run a javascript at the end of dialplan to
>>> generate cdrs?
>>> because (mod_cdr_csv) is giving me hard time as it rotates Master
>>> file on
>>> machine reboots or shutdown signals.
>>> javascript or LUA for preferences?
>>>
>>> thank you
>>>
>>>
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>>
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Re: [Freeswitch-users] Cannot Do this Basic thing

[Seven Du]

You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ?

2009/12/4 Samuel Abekah-Mensah :
> Hi
>
> Sorry .xm is a typo. I actually shut down the server and restarted. The
> log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
> server IP address ) and then put the user in that domain.  Isn't the
> default domain that of the server FS is running on ?
> 2319.xml is in /usr/local/freeswitch/conf/directory/default/
>
> Thanks for your time
>
>
>
> Michael Collins wrote:
>>
>>
>> On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah
>> mailto:ab...@greatiam.com>> wrote:
>>
>>     I have copied 1001.xml in directory/default to a test user 2319.xm
>>     changing or instances of 1001 in the file to 2319. I then went into
>>     default.xml  in directory folder and in one of the groups  just
>>     mimicked
>>     1001 details by changing 1001 to 2319.
>>
>>     Connecting  to FS gives Forbidden message. However 1001 connects
>>     without
>>     a problem.  What have I missed ?
>>
>>     Is there a place that just puts things in do this and that and that to
>>     create a new user ?
>>
>>
>> Did you execute "reloadxml" from the fs cli before trying to connect
>> with 2319? Also I'm assuming that "2319.xm" is a typo and you actually
>> created "2319.xml" in the default/directory subdir.
>> -MC
>>
>
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Re: [Freeswitch-users] Generate cdrs

[Seven Du]

why not try mod_xml_cdr?

2009/12/4 Mouncif Benniane :
> is it possible to run a javascript at the end of dialplan to generate cdrs?
> because (mod_cdr_csv) is giving me hard time as it rotates Master file on
> machine reboots or shutdown signals.
> javascript or LUA for preferences?
>
> thank you
>
>
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Re: [Freeswitch-users] How to run a JS script periodically

[Seven Du]

Not sure about js, but in lua, you can use luarun to run a
long-running script like


loop
do sth.
sleep 5min
end

and also it can be set to start with freeswitch in lua.conf.xml

I guess you can also use jsrun to run js.

And, if you run every 5 min, why not use crontab?

fs_cli -x "jsrun xx.js"


2009/12/3 Oscav :
>
> Hi,
>
> Someone knows how to run periodically a JS script ?? The purpose is to write
> to a db some global informations (Global Variables) about FS like every 5
> minutes.
>
> Thanks.
>
>
> --
> View this message in context: 
> http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Recording with Native File PCMU

[Seven Du]

Yeah, that's why I had to record to two files(read&write) and need to mix
together by using sox.  Do you only try to using PCMU to save CPU power
matt? As Anthony said, the difference can be ignored. And you also need to
take extra effort to make sure transcoding will not happen on a
conversation.

But it maybe useful for expensive codecs like g729, iLBC, speex etc for
recording heavy scenarios. I'd like to take a look if there is a 5k bounty
;)

2009/11/26 Anthony Minessale 

> The processor power saved is negligible between PCMU and raw PCM and not
> worth the fuss.
> If you didn't decode the audio first you would not be able to mix the
> stream to produce a single file.
> So if we went to the trouble of making native media bugs to be able to do
> that you could barely use them so it would not be worth the 5k or more
> bounty to develop that functionality.
>
>
>
> On Sun, Nov 22, 2009 at 12:48 PM, Matthew Fong wrote:
>
>> I'm trying to conserve processor power by recording in native file format,
>> PCMU in my case. It works great with the following line
>>
>> session:execute("record",
>> "/tmp/my_recording."..session:getVariable("read_codec"));
>>
>> however it fails to work with
>>
>> session:execute("record_session",
>> "/tmp/my_recording."..session:getVariable("read_codec"));
>> or
>> record = api:execute("sched_api", '+1 none uuid_record
>> '..session:getVariable("uuid")..' start
>> /tmp/my_recording.'..session:getVariable("read_codec"));
>>
>> Why is it that it works with record, but not with record_session or
>> uuid_record? Is there something I'm over looking? In the latter two the
>> consul reports
>>
>> 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File
>> [/tmp/my_recording.PCMU] 8000hz
>>
>> as if it's recording, but /tmp/my_recording.PCMU never shows up. However
>> if I change it to .wav instead of .PCMU it works. Any ideas?
>>
>> --matt
>>
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>
>
> --
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>
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Re: [Freeswitch-users] how to enable short recordings

[Seven Du]

And you may also would like to update the wiki as well if the var is not
there.

2009/11/26 kokoska rokoska 

> Thank you very much, Anthony, for your help!
>
> I'm nearly at current trunk (15653) and
> 
> works great :-)
>
> Many thanks once more!
>
> Best regards,
>
> kokoska.rokoska
>
>
> Anthony Minessale napsal(a):
> > use the variable RECORD_MIN_SEC
> >
> > This was added in revision 15271 so if you are below that I recommend
> > updating to latest trunk.
> >
> >
> > On Wed, Nov 25, 2009 at 10:03 AM, kokoska rokoska
> > mailto:kokoska.roko...@post.cz>> wrote:
> >
> > Yes, Brian, I need them :-)
> >
> > They don't contain speech - instead, they contain few "computer
> > generated" tones and I should store them in max quality for later
> > proccessing (i.e. analysis)...
> >
> > Best regards,
> >
> > kokoska.rokoska
> >
> >
> > Brian West napsal(a):
> > > Really you want to keep 1-3 second files around?
> > >
> > > /b
> > >
> > > On Nov 25, 2009, at 9:46 AM, kokoska rokoska wrote:
> > >
> > >> Thank you very much, Brian, for your interest!
> > >>
> > >> It is standard recording:
> > >>
> > >> 
> > >>
> > >> Best regards,
> > >>
> > >> kokoska.rokoska
> > >
> > >
> > > ___
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users@lists.freeswitch.org
> > 
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
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> > 
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> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
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> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
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> > MSN:anthony_miness...@hotmail.com 
> >  >
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> >  >
> > IRC: irc.freenode.net  #freeswitch
> >
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> >  >
> > iax:gu...@conference.freeswitch.org/888
> > 
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> >  >
> > pstn:213-799-1400
> >
> >
> > 
> >
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Re: [Freeswitch-users] Recording with Native File PCMU

[Seven Du]

http://code.google.com/p/mod-recpld/

It's out-dated. I originally wrote it to record raw G.729 codec on
passthrough mode. It worked before and then we abandoned that since We felt
G729 cannot deliver good sound particularly on a cross-continent network.

The code is written when I don't know much about FS internals, perhaps it's
easier to write some mod with indicate no-transcoding in switch_rtp.c.

2009/11/25 Rupa Schomaker 

> Yes
>
>
> On Wed, Nov 25, 2009 at 9:44 AM, Matthew Fong  wrote:
>
>> so is using session_record with .wav my best option for recording bridged
>> calls?
>>
>> --matt
>>
>>
>> On Wed, Nov 25, 2009 at 7:18 AM, Brian West  wrote:
>>
>>> These two options attach media bugs on to the session.   Which doesn't
>>> work with native files as far as I know.
>>>
>>> /b
>>>
>>> On Nov 25, 2009, at 8:53 AM, Matthew Fong wrote:
>>>
>>> > record WORKS!, but uuid_record and session_record do not want to
>>> > record in native format. do uuid_record and session_record work with
>>> > native format? or is it not going to be possible to record a bridged
>>> > call in native format?...maybe because there are two different
>>> > channels with a bridged call?
>>>
>>>
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>>
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>
>
> --
> -Rupa
>
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Re: [Freeswitch-users] Business/holiday hours routing

[Seven Du]

2009/11/24 Adam Ford 

> Yes, that wiki page is what I was referring to when I said it seems simple
> enough to do it in the XML dialplan. So other than the one lua suggestion,
> it seems the majority say XML is the way to go eh?
>

Yeah, I made that lua script when the latest XML time routing feature was
unavailable.

>
> -Adam
>
> Andrew - that does sound like a useful patch, is it in svn or unpublished?
>
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Andrew
> Thompson
> Sent: Monday, November 23, 2009 6:48 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Business/holiday hours routing
>
> On Mon, Nov 23, 2009 at 06:26:47PM -0700, Adam Ford wrote:
> > Is there a standard module for FreeSWITCH out there that people use for
> > routing calls based on business hours and a holiday schedule? Or is
> everyone
> > just creating their own in the XML dialplan?(which seems pretty simple)
> >
> >
> >
> > I can't seem to find anything on the wiki, but might just be searching
> for
> > the wrong thing.  I am relatively new at FreeSWITCH and would rather
> follow
> > what the community has decided is the best practice, instead of trying to
> > reinvent the wheel myself.
> >
> I assume you've seen this:
>
> http://wiki.freeswitch.org/wiki/Time_of_Day_Routing
>
> I have a patch that'll let you specify the nth day of the nth week via
> wday="3,4" for the 4th tuesday in the month. This willl let you do
> vacations like thanksgiving, MLK day, etc as well.
>
> Andrew
>
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Re: [Freeswitch-users] Business/holiday hours routing

[Seven Du]

XML has basic conditioning, but lua rocks.

-- Time condition for sales 1

--session:setAutoHangup(false)

function do_transfer(extn)

--print(extn)
session:transfer(extn, "XML", "sales")
end


now = os.date("%H:%M")
w = tonumber(os.date("%w"))

if w >= 1 and w <=5 then

if ( now >= "09:00" and now < "20:30" ) then
do_transfer("sales_fifo_1")
elseif ( now >= "20:30" and now < "22:30" ) then
do_transfer("sales_fifo_2")
else
do_transfer("sales_fifo_cellphone")
end

else

if ( now >= "10:00" and now < "19:00" ) then
do_transfer("sales_fifo_1")
elseif (now >= "20:00" and now < "22:30" ) then
do_transfer("sales_fifo_2")
else
do_transfer("sales_fifo_cellphone")
end
end


2009/11/24 Adam Ford 

>  Is there a standard module for FreeSWITCH out there that people use for
> routing calls based on business hours and a holiday schedule? Or is everyone
> just creating their own in the XML dialplan?(which seems pretty simple)
>
>
>
> I can’t seem to find anything on the wiki, but might just be searching for
> the wrong thing.  I am relatively new at FreeSWITCH and would rather follow
> what the community has decided is the best practice, instead of trying to
> reinvent the wheel myself.
>
>
>
> Thanks for any input,
>
> -Adam
>
>
>
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Re: [Freeswitch-users] FIFO Orgination_caller_id

[Seven Du]

Yes, that's what we are doing.

2009/11/24 Brian West 

> You do realize that the whole concept is OLD skewl.  You should be
> popping this info via external resources when the agent is bridged to
> the caller and the info is there before they are done saying "thanks
> for calling spacely sprockets, this is George how may I help you  "
>
> /b
>
>
> On Nov 23, 2009, at 7:07 PM, Michael Collins wrote:
>
> > Tony and Brian were discussing this today. They bring up a really
> > good point: do you want to risk having calls remain on hold as they
> > bounce around looking for an agent? This can happen if you pre-
> > determine which caller goes to which agent and the agent doesn't
> > answer. I do understand why this feature matters to many people -
> > it's how old school ACD systems work. However, mod_fifo is more
> > efficient. It's hard to justify decreasing call routing efficiency
> > in order to display the caller's info to the on-hook agent prior to
> > answering.
> >
> > -MC
>
>
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Re: [Freeswitch-users] FIFO Orgination_caller_id

[Seven Du]

And because it's static string for on-hook members, it's hard to set
dynamically. For now, I'm using a callback way - whenever the sip client
answered the call, it fetch the real connected number from a http server.
That's not ideal because not only it add the complexity but also the callee
have no idea what the number is before answer.

The problem for on-hook agent is that it call the agent first, and then get
one customer from the fifo queue, so it is not possible to let the agent
know the real caller-id before answer. Ideas?

2009/11/24 Anthony Minessale 

> if you add
> {origination_caller_id_name=foo,origination_caller_id_number=123} before the
> static entries for the on hook agent it will prevail over the default one.
>
> If you are using 1.0.4, this feature is only available in trunk or one of
> the 1.0.5 pre releases.
>
>
> On Mon, Nov 23, 2009 at 4:49 PM, Adam Ford  wrote:
>
>>  Is there any way to set the origination_caller_id for a FIFO outbound
>> call to an on-hook agent?  I can’t find anything in the wiki about a FIFO or
>> member variable to set this. It seems to be set to ‘Queue’ by default, and
>> appears to be hardcoded in the module source.  It would be nice to be able
>> to change per FIFO queue.  That way agents that handle multiple companies
>> can more easily see which queue is calling and answer accordingly.
>>
>>
>>
>> It is not a big deal, since it does automatically set the
>> origination_caller_id_number to ‘fifo+’.  However, depending on
>> the phone, the caller ID number is not always readily shown, and must be
>> looked for.
>>
>>
>>
>> Thanks to anyone who has some insight on this,
>>
>> -Adam
>>
>>
>>
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>
>
> --
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>
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>
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Re: [Freeswitch-users] Recording with Native File PCMU

[Seven Du]

did you try without any .wav or .PCMU?

2009/11/23 Matthew Fong 

> I'm trying to conserve processor power by recording in native file format,
> PCMU in my case. It works great with the following line
>
> session:execute("record",
> "/tmp/my_recording."..session:getVariable("read_codec"));
>
> however it fails to work with
>
> session:execute("record_session",
> "/tmp/my_recording."..session:getVariable("read_codec"));
> or
> record = api:execute("sched_api", '+1 none uuid_record
> '..session:getVariable("uuid")..' start
> /tmp/my_recording.'..session:getVariable("read_codec"));
>
> Why is it that it works with record, but not with record_session or
> uuid_record? Is there something I'm over looking? In the latter two the
> consul reports
>
> 2009-11-22 18:39:04.265284 [INFO] mod_native_file.c:82 Opening File
> [/tmp/my_recording.PCMU] 8000hz
>
> as if it's recording, but /tmp/my_recording.PCMU never shows up. However if
> I change it to .wav instead of .PCMU it works. Any ideas?
>
> --matt
>
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Re: [Freeswitch-users] Changing User-Agent String

[Seven Du]

lol:

2009/11/19 Anthony Minessale 

> maybe you could send them 183 then 4 180's or send them an invite and
> pretend to deadlock and not send any more sip traffic as a way of
> identifying yourself
>
> On Wed, Nov 18, 2009 at 4:05 PM, Brian West  wrote:
>
>> Well this is a bit more informal vs the wiki where people take it as
>> fact!  :)  Plus its a little humpday humor!
>>
>> /b
>>
>> On Nov 18, 2009, at 3:30 PM, Rob Forman wrote:
>>
>> > lol!
>> >
>> > we have to play nice in the wiki but the mailing list is another
>> > story.
>> >
>> >
>> > On Nov 18, 2009, at 3:20 PM, Brian West wrote:
>> >
>> >> Sounds like you need to take a baseball bat to their forehead.
>> >>
>> >> /b
>>
>>
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>
>
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>
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Re: [Freeswitch-users] mod_fifo and multi-tenancy

[Seven Du]

I once wrote a patch for "fifo delete", but didn't submit to jira. If
someone think it's useful to merge into trunk, I think I can still find the
code, but sure need to test with the current trunk.

2009/11/19 Michael Collins 

>
>
> On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi wrote:
>
>>
>>
>> On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins wrote:
>>
>>>
>>>
>>> On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi 
>>> wrote:
>>>
 About mod_fifo, it would be safe to use it in multi-tenancy scenarios
 where domains are created and deleted all the time and in consequence, 
 fifos
 are created all the time? I mean, fifos are eventually destroyed by 
 mod_fifo
 itself. Is this correct?

 br,
 takeshi


>>> No, FIFOs are not "destroyed" automatically just because the last member
>>> goes away. Tony says that an empty FIFO takes up almost no memory so
>>> performance shouldn't be an issue. You can always issue the API command:
>>> *fifo reparse del_all* to clean everything out if you feel like things
>>> are getting out of hand.
>>>
>>
>> Thanks,
>> I have updated the mod_fifo wiki page.
>>
>>
> FYI, I was doing some other research and I noticed this on the mod_fifo
> wiki page:
>
> *fifo_destroy_after_use*: FreeSWITCH automatically create a new FIFO when
> the first time use it, and keep in the memory hash. This var tell FreeSWITCH
> destroy it to save memory. Using for a one time FIFO.
>
> So... you do have that option as well.
> -MC
>
>
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[Freeswitch-users] prefix Freeswitch-users vs. FreeSWITCH-Users

[Seven Du]

Would it be better to change the list subject prefix from [Freeswitch-users]
to FreeSWITCH-Users?
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Re: [Freeswitch-users] mod_skypiax for OSX?????

[Seven Du]

2009/11/6 Giovanni Maruzzelli 

> On Thu, Nov 5, 2009 at 6:57 PM, Seven Du  wrote:
> > Ciao Giovanni,
> >
> > Do you still plan to merge this?
>
> Sorry Seven,
>
> I've lost track of this, and now I'm so sick I'm completely un-useful ;).
>
>
That's OK, we all have a lot of things to do each day.


> But yes, I would like to do it, if you think it is in a useful state.
>
> Can you please create a Jira and attach an svn diff, so in the next
> days I can merge it?
>
> I'd like to create a jira and I think it would be easier if you can
directly merge from branch. However the branch is a bit old and it would
need some days if you need svn diff based on the current trunk.

Thanks.


> -giovanni
>
> >
> > 2009/9/5 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
> >> >>
> >> >> ___
> >> >> 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
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> >> >
> >> >
> >> > ___
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> >> >
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>
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> Cell : +39-347-2665618
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Re: [Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session

[Seven Du]

No, I'm Seven and never used hangup hook. you must had though I was Dome.

Sorry, I'm not tend to hijack this thread, just though it's the same topic.

2009/11/2 Anthony Minessale 

> We already concluded its your unacceptabe use of originate in hangup hook
> right?
>
> On Nov 1, 2009 7:45 PM, "Seven Du"  wrote:
>
> Just suspicious would be possible that happened on sqlite stage? I
> manually deleted the channels from sqlite and nothing bad happend.
> just FYI.
>
> -- Forwarded message --
> From: Seven Du 
> Date: Sun, 1 Nov 2009 10:24:32 +0800
> Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session
> To:
>
> Thank you Anthony. We are on r14696 and no non-standard mods loaded. we
> even
> unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both
> inbound
> and outbound. I must be very careful if I upgrade to trunk and turn on
> rwlock debug because it's on production and the the problem not happening
> that much so would be hard to trace. But I will find time to test and
> report
> back.
>
> FYI, I also noticed that in some zombile channels, couple of INVITEs sent
> but never got a response. However, it sent out an ACK at last.
>
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>
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[Freeswitch-users] Fwd: Many CS_REPORTING state Zombie session

[Seven Du]

Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.

-- Forwarded message --
From: Seven Du 
Date: Sun, 1 Nov 2009 10:24:32 +0800
Subject: Fwd: [Freeswitch-users] Many CS_REPORTING state Zombie session
To:

Thank you Anthony. We are on r14696 and no non-standard mods loaded. we even
unloaded mod_xml_cdr. But we are heavily using mod_erlang_event both inbound
and outbound. I must be very careful if I upgrade to trunk and turn on
rwlock debug because it's on production and the the problem not happening
that much so would be hard to trace. But I will find time to test and report
back.

FYI, I also noticed that in some zombile channels, couple of INVITEs sent
but never got a response. However, it sent out an ACK at last.

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Re: [Freeswitch-users] Many CS_REPORTING state Zombie session

[Seven Du]

I also got some zombie channels, if someone can help me take a look that's
really nice.

http://pastebin.freeswitch.org/10912

I only loaded mod_cdr_csv
Is it ok to use mod_xml_cdr?

Thanks.


2009/11/1 Dome Charoenyost 

> I found bug in fscore_pb   $pwd should be $mypwd
>
> Now i post already. please help me check.
>
> Dome C.
>
>
> 2009/11/1 João Mesquita :
> > No, mod_nibblebill definetely needs to be enhanced but it is not the
> problem
> > and it can be used with high load traffic.
> >
> > The one I am not sure about is odbc_query since it was not developed for
> > that.
> >
> > Do what Rupa said, please.
> >
> > Regards,
> >
> > JM
> >
> > On Sat, Oct 31, 2009 at 3:11 PM, Brian West 
> wrote:
> >>
> >> I think once you get the backtrace like rupa said we can see that maybe
> >> odbc_query is really hanging or something similar.
> >> /b
> >> On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
> >>
> >> 2009/10/31 Brian West :
> >>
> >> You should never do billing inline with the session thread is all I'm
> >>
> >> saying.
> >>
> >> ___
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> >>
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Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)

[Seven Du]

not sure about this, but did you try send dtmf to uuid
723f3dbb-b87b-4cd4-98fc-698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
>
> 3d18cc6e8454 ?
>


2009/10/20 Nikita Belov 

> Yeah. I'm using it for starting eavesdrop:
> SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: 723f3dbb-b87b-4cd4-98fc-698eed7f2bdb
>
> But we are discussing how to switch eavesdrop to allow "spy" to speak with
> one of the sides. Do you know any application which can I use for this?
> Mike
> said I should use sendevent command to control eavesdrop, but I still
> confused how can I do it.
>
> > -Original Message-
> > From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-
> > users-boun...@lists.freeswitch.org] On Behalf Of Brian West
> > Sent: Monday, October 19, 2009 8:59 PM
> > To: freeswitch-users@lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference
> > call)
> >
> > Please review this:
> >
> > http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg
> >
> > SendMsg 
> > call-command: execute
> > execute-app-name: 
> > execute-app-arg: 
> >
> >
> >
> > /b
> >
> > On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:
> >
> > > And what event name to use for sendevent command? Sorry for
> > > importunity.
> > > sendevent ???
> > > Unique-ID: b9d6a35c-ee0c-4203-8d65-ed816e0a9c19
> > > eavesdrop-command: 1
> > >
> >
> >
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Re: [Freeswitch-users] freeswitch french community

[Seven Du]

There are some plugins for mediawiki to support multilanguage, either inline
or on a separate page. However, both have Pros and Cons. e.g.

multi-language support this with a separate page

some_page #default to en
some_page/fr Franch
some_page/zh Chinese

Pros: clear, available languages can be listed in "Other languages for this
page"
Cons: hard to keep all languages in sync

other plugins can do inline translation:

<@en|hello>
<@fr|Ca Va>
<@zh|你好>

Pros: untranslated items remains in default lang, no sync headache
Cons: It breaks a page into sentences or items, also if we have more than 10
languages the source code maybe really a mess.


2009/10/19 Diego Viola 

> The other way we could do is set up a fr.freeswitch.org with mediawiki on
> it and you start to translate from the English wiki.
>
> But I think inline translation would be better for this.
>
> Just my two cents.
>
> Diego
>
>
>
> On Sun, Oct 18, 2009 at 11:13 PM, Diego Viola wrote:
>
>> Hi Meftah,
>>
>> I'm more than happy to help with the documentation and anything else
>> related to FreeSWITCH.
>>
>> As you mention the documentation and translation, there are some things
>> that I would really like to improve in the documentation.
>>
>> I propose that we start a new documentation from the ground up, we really
>> need a better wiki engine that supports inline translation and a better
>> search engine. I'm open to do all the hard work like porting the entire
>> documentation to it.
>>
>> We could put the new wiki on a sub-domain like: newdocs.freeswitch.organd I 
>> do all the work there, without affecting others.
>>
>> That way we could have inline documentation like this:
>>
>> status - show status information
>> status - mostrar informacion de estado
>>
>> It's up to others to decide if this is really worth or not.
>>
>> I'm open to help and to give my time to improve things like documentation,
>> etc :).
>>
>> Diego
>>
>>
>> On Sun, Oct 18, 2009 at 8:08 PM, Meftah Tayeb wrote:
>>
>>> hello folk
>>> we are starting a new freeswitch community for french users
>>> the newly created Irc channel is #freeswitch-fr
>>> freeswitch staf, please can we register this channel officialy to the
>>> freeswitch freenode group?
>>> also can we build a french documentation and host it somehere? in the
>>> wiki?
>>> my long time friend, diegoviola i going to help about the freeswitch
>>> wiki markup
>>> we are waiting for french users to join, and welcome
>>> thank toAnthony Minessale for accepting the community
>>> welcome!
>>>
>>>
>>> __ Information from ESET NOD32 Antivirus, version of virus
>>> signature database 4520 (20091018) __
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
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>>
>>
>
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Re: [Freeswitch-users] mod_opal - call charged before H.225 connect

[Seven Du]

that will make life easier.

2009/10/13 Brian West 

> Does anyone see a problem with hosting mod_h323 in our SVN?  I would
> like to centralize everything we can to reuse our issue tracking
> resources and not fragment the community if possible.
>
> /b
>
> On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
>
> > hi,
> >
> > finally i compiled it right ... had a stupid issue with ekiga and
> > wrong ptlib in place...
> >
> > anyhow, i loaded the module and will continue the tests
> > tomorrow ...first thing i arrive in my office :P
> >
> >
>
>
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Re: [Freeswitch-users] Re corded file as voicemail.

[Seven Du]

It was a problem and has been fixed in the last trunk. Just update to the
latest code should be ok.

btw, the developers using jira to track bugs, so feel free to report one (as
you see http://jira.freeswitch.org/browse/FSCORE-463) if you think it's a
bug next time.

2009/10/12 Nagalenoj 

>
> Whats the conclusion.?!
>
>
> Brian West-3 wrote:
> >
> > Please open a jira please this did work but a recent change in
> > switch_core_codec caused this to appear I usually test this regularly
> > but haven't run thru a full run of tests lately.
> > /b
> >
>
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Re: [Freeswitch-users] Re corded file as voicemail.

[Seven Du]

I won't try until I need that, but I believe it works. Thanks Brian.

2009/10/13 Brian West 

> Fixed... svn up.
> /b
>
> On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
>
> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.
>
>
>
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Re: [Freeswitch-users] mod_sofia.c registered calls how to know

[Seven Du]

try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open them.
not sure how to do that on windows, but on linux:


# sqlite3 xx.db
sqlite> select * from sip_registration;

2009/10/12 srinivasula reddy 

> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
> dont have any databasae installed in my system, then from sqllite will come
> into picture, and how can i see the registered users data from sqllite.
>
>
> Thanks
> Srinivas
>
> On Mon, Oct 12, 2009 at 3:04 AM, Michael Jerris  wrote:
>
>>
>> On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
>>
>>
>> Hi
>>
>> can any please tell me where registered calls are stored, so when incoming
>> call came to mod_sofia.c how it will check it is registered or not?\\
>>
>>
>> Calls are not registered and calls have nothing to do with registration.
>>  Users are registered so that you may send calls to them.  Registration data
>> is stored either in a sqlite database, or optionally if you setup odbc, in
>> another database of your choice.  If you try to send a call to an
>> unregistered user in the dialplan using the proper syntax to send calls to
>> registered users (see the wiki for more details), and that user is not
>> registered, the bridge app will fail, optionally letting you continue on in
>> the dialplan based on variables such as continue_on_fail and
>> hangup_after_bridge.  You can use the sofia_contact function to see if there
>> is anyone registered to a specific user.
>>
>> Mike
>>
>>
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>
>
> --
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Re: [Freeswitch-users] Re corded file as voicemail.

[Seven Du]

http://jira.freeswitch.org/browse/MODCODEC-15

Is it ok I assigned to you ?

Thanks.

2009/10/12 Brian West 

> It was possible but we have a regression in the code that isn't
> letting that happen right now... hence the reason i said Open a jira
> so we could fix it.
>
> IS THAT not clear?
>
> /b
>
> On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote:
>
> >
> > Whats the conclusion.?!
>
>
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Re: [Freeswitch-users] Re corded file as voicemail.

[Seven Du]

I set to true because brian said it can play stereo files but no lucky for
me.

2009/10/12 Jason White 

> Seven Du  wrote:
> > originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
> > &bridge(sofia/gateway/yy/yy)
>
> Shouldn't that be record_stereo=false for mono recording?
>
>
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Re: [Freeswitch-users] Re corded file as voicemail.

[Seven Du]

e Change CS_CONSUME_MEDIA -> CS_EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:503
(sofia/eqenglish/seven1240) State CONSUME_MEDIA going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491
(sofia/eqenglish/seven1240) State EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:173 sofia/eqenglish/seven1240
SOFIA EXECUTE
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:151
sofia/eqenglish/seven1240 Standard EXECUTE
EXECUTE sofia/eqenglish/seven1240 playback(/tmp/a.wav)
2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2
channels, muxing to mono will occur.
2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently
unsupported. please downsample audio source to mono.
2009-10-12 01:41:30.349961 [DEBUG] switch_ivr_play_say.c:1113 Raw Codec
Activation Failed l...@8000hz 2 channels 20ms
2009-10-12 01:41:30.349961 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/eqenglish/seven1240 [CS_EXECUTE] [NORMAL_CLEARING]
2009-10-12 01:41:30.349961 [DEBUG] switch_channel.c:1715 Send signal
sofia/eqenglish/seven1240 [KILL]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:491
(sofia/eqenglish/seven1240) State EXECUTE going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_HANGUP
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434
(sofia/eqenglish/seven1240) State HANGUP
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:338 Channel
sofia/eqenglish/seven1240 hanging up, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:376 Sending BYE to
sofia/eqenglish/seven1240
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:46
sofia/eqenglish/seven1240 Standard HANGUP, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:434
(sofia/eqenglish/seven1240) State HANGUP going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:476
(sofia/eqenglish/seven1240) State Change CS_HANGUP -> CS_REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:398
(sofia/eqenglish/seven1240) Running State Change CS_REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613
(sofia/eqenglish/seven1240) State REPORTING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:53
sofia/eqenglish/seven1240 Standard REPORTING, cause: NORMAL_CLEARING
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:613
(sofia/eqenglish/seven1240) State REPORTING going to sleep
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:411
(sofia/eqenglish/seven1240) State Change CS_REPORTING -> CS_DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:932 Send signal
sofia/eqenglish/seven1240 [BREAK]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_session.c:1068 Session 2178
(sofia/eqenglish/seven1240) Locked, Waiting on external entities
2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1086 Session 2178
(sofia/eqenglish/seven1240) Ended
2009-10-12 01:41:30.349961 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/eqenglish/seven1240 [CS_DESTROY]
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:556
(sofia/eqenglish/seven1240) Running State Change CS_DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565
(sofia/eqenglish/seven1240) State DESTROY
2009-10-12 01:41:30.349961 [DEBUG] mod_sofia.c:255 sofia/eqenglish/seven1240
SOFIA DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:60
sofia/eqenglish/seven1240 Standard DESTROY
2009-10-12 01:41:30.349961 [DEBUG] switch_core_state_machine.c:565
(sofia/eqenglish/seven1240) State DESTROY going to sleep
2009-10-12 01:41:30.349961 [NOTICE] switch_cpp.cpp:1130 ==DebugVar==
gateway_name: skype


2009/10/12 Brian West 

> FreeSWITCH can play back stereo files it'll just mux them down to mono
> before playing... can you elaborate on the error you're getting?
> /b
>
> On Oct 10, 2009, at 12:40 AM, Seven Du wrote:
>
> Yes, it's discussed before.
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
>
> set that var to false before you record.
>
>
>
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Re: [Freeswitch-users] Difference between park and valet_park

[Seven Du]

search this list, just has been discussed.

2009/10/10 velusamy velu 

> Dear All,
>  Could you please any one explain the difference between parking and
> valet parking?
>
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Re: [Freeswitch-users] Re corded file as voicemail.

[Seven Du]

Yes, it's discussed before.

http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO

set that var to false before you record.


2009/10/10 Jason White 

> Nagalenoj  wrote:
> >
> > No, When I do voicemail_inject and check through voicemail, it is not
> playing
> > the file instead reporting the following error.
> >
> > 'Stereo is currently not supported, please downsample to mono.'
>
> This has been discussed on the list before. I think the solution was to use
> sox to convert the files to mono. You might need a script to do this before
> injecting them into the voicemail.
>
>
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Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

[Seven Du]

It's very cool. But think about the following scenario:

Alice answers a call from Bob.
Alice transfers Bob to 6001 and hangs up.
Bob is now in parking stall 6001, hearing MOH.
Alice calls Charlie and tells him that he has a call parked in 6001.

  Dee call in and answered by Alice  # at that time, Alice don't know the
lot is empty or not
  If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
expected

Charlie calls 6000, waits for prompt, then dials 6001#.
Charlie Listening to music.

Event it's rarely happen, I do think Alice would like to now the if lot is
empty or not before transfer. Make sense?

2009/10/9 Michael Collins 

> FYI,
>
> The FreeSWITCH devs have added valet parking! Check it out:
> http://www.freeswitch.org/node/207
>
> Let us know what you think.
>
> Thanks,
> Michael
>
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Re: [Freeswitch-users] Dynamic updation of groups in default.xml

[Seven Du]

change the xml and execute "reloadxml" in FS console or fs_cli

or you can check mod_xml_curl

2009/10/6 srinivasula reddy 

> Hi,
> Can any one tell me how to add users dynamically to groups in default.xml,
> with out restart the freeswitch.
>
>
> Thanks
> Srinivasula Reddy K
>
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Re: [Freeswitch-users] Mobile Phone As GSM Gateway....

[Seven Du]

maybe you can check this: http://www.gsmopen.org/

2009/10/6 Moiz Chinoy 

> Hi,
>
> Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
> use this as a GSM gateway?
>
> --
> Regards,
> Moiz Chinoy.
>
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Re: [Freeswitch-users] wav files compression

[Seven Du]

FS support recording to mp3 directly through mod_shout but you might not
want to use that for performance reason.

You can use lame to convert .wav to .mp3 regularly( by crontab if you on
linux) or immediately after record(by using iwatch, or listening to event
socket to see when the record is done ).

2009/10/3 Keith Wood 

>
> I am working on an implementation for managing thousands of IVR within an
> organization.  Right now, I am storing all audio files in wav format, but it
> quickly become unmanagable because the size of these wav files ( 8 bits mono
> ) quickly consuming a lot of the disk space.
>
> Is there anyway I can store those audio files and still have high quality
> audio for IVR?  I know mp3 is smaller but freeswitch does not support it.
>
> any ideas?
>
> keith
>
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Re: [Freeswitch-users] confuse of regex command with "|"

[Seven Du]

Thanks. But I think it would be nicer if the regex looks the same as in
dialplan. can we add a optional separator arg on this case?

regex | [seperator]
regex "10":"09|10" :

2009/9/29 Brian West 

> Yep escape it.
>
> /b
>
> On Sep 28, 2009, at 10:47 AM, Michael Collins wrote:
>
> > regex "10"|"09\|10"
>
>
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[Freeswitch-users] confuse of regex command with "|"

[Seven Du]

Hi, is this a bug?

freeswi...@internal> regex "10"|"09|10"
false
freeswi...@internal> regex "10"|"10"
true
freeswi...@internal> regex "10"|"(09|10)"
false
freeswi...@internal> 2009-09-27 11:47:00.815355 [ERR] switch_regex.c:101
COMPILE ERROR: 4 [missing )]["(09]


the first one should be true?
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Re: [Freeswitch-users] Mod_vmd and Session Locking.

[Seven Du]

what's your rev? I think rev14494 might related to you.

2009/9/27 Vinuth Madinur 

> Hello All,
> I'm trying to do a simple dialer, where I am:
>
> 1. Initiating mod_vmd on channel answer.
> 2. Staying quiet until there is a beep.
> 3. Leave a message on beep.
> 4. Hangup.
>
> (in my scenario it's guaranteed to hit voicemail.)
>
> But the session isn't exiting properly after "hangup", with last statement
> being: " switch_core_session.c:1068 Session 1 (sofia/external/1002) Locked,
> Waiting on external entities".
>
> It doesn't go to CS_DONE state. I see [CRIT] level logs when shutting down
> freeswitch waiting for all these sessions to complete.
> But all these sessions are getting stuck while attempting to get a write
> lock on session object.
> I tried this with both 1.0.3 and 1.0.4 and the behavior is similar.
> Can someone please help? Or point me toward fixing it?
>
> Thanks,
> Gubbi.
>
>
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Re: [Freeswitch-users] Sofia regiter to provider with multiple account

[Seven Du]

I would write a simple lua script to do a round robin hunt.

2009/9/27 Dome Charoenyost 

> 2009/9/26 Brian West :
> > Why would they require you to have 50 accounts?  Doesn't seem sane to
> > me.
> They provide for pc to phone user and i want to use for my corp.
>
> :)
>
> >
> > /b
> >
> > On Sep 25, 2009, at 9:48 PM, Dome Charoenyost wrote:
> >
> >> 2009/9/26 Michael Jerris :
> >>>
> >>> On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote:
> >>>
>  Dear All,
> How to config freeswitch for support this case ?
>  1. FS register to provider about 50 user account. (Each account
>  can't support multiple call in same time)
> >>>
> >>> Sofia gateways
> >>>
>  2. FS Check account not inuse before call out.
> >>>
> >>> mod_limit
> >>>
>  3. User account should be round-robin
> >>>
> >>> what does this mean?
> >>>
> >>
> >> Each user account have own balance (in provider). So i want to use
> >> all user
> >
> >
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Re: [Freeswitch-users] Bind to more than one ethernet interface

[Seven Du]

It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.

2009/9/24 Yehavi Bourvine 

> Hello,
>
>   I am trying to run FreeSwitch on a machine which has more than one
> interface, all of them should be used for SIP. The FreeSwitch binds only to
> the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0
> but it doesn't help.
>
> Any idea what to do?
>
> Thanks! _Yehavi:
>
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Re: [Freeswitch-users] Bind to more than one ethernet interface

[Seven Du]

sorry when I said "on profile" I want to say "one profile"

2009/9/24 Seven Du 

> It not possible to use 0.0.0.0 for on profile. however, you can create more
> sip profiles for each of your interfaces. Search freeswitch-users archievs
> then you will find similar topics.
>
> 2009/9/24 Yehavi Bourvine 
>
>> Hello,
>>
>>   I am trying to run FreeSwitch on a machine which has more than one
>> interface, all of them should be used for SIP. The FreeSwitch binds only to
>> the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0
>> but it doesn't help.
>>
>> Any idea what to do?
>>
>> Thanks! _Yehavi:
>>
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Re: [Freeswitch-users] session record does not for very short calls

[Seven Du]

I think the file was there but deleted by FreeSWITCH if it thinks it  
was too short (like 3 seconds?). If I'm not wrong, someone requested  
this feature becuase FreeSWITCH left too many small recordings.


On Sep 17, 2009, at 1:27 AM, João Mesquita wrote:
> I think you need to upgrade your version before we even take a look  
> at that... You are so far behind trunk right now and lots of things  
> have been changed since then.
>
> Not sure if this would solve your problem but not a lot of ppl will  
> look at your problem when you run this version.
>
> jmesquita
>
> On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact  
>  wrote:
> FreeSWITCH Version 1.0.trunk (12790M)
>
>
> I have this in my DP
>
>   
>
>   
>
>   
>
>
> works fine as long as the call is long enough.  But if the call is  
> only, say, 3-4 seconds long (or something very short like that),  
> then the wav file is never created with the audio in it.
>
>
> Is there a work around for this?
>
>
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Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects

[Seven Du]

Hi MC,

Months ago we had tried the multi-language plugin on MediaWiki, I know  
you are still planning to do this but I just want how far it goes.  
Count me in when you are short of hand.

On Sep 15, 2009, at 11:25 PM, Michael Collins wrote:
> Demuel,
> Thanks for the input. Yes, we want to avoid chaos. I will work to  
> keep everyone organized. I would welcome a mass of people all trying  
> to do different things because the challenge would simply be to keep  
> them organized. Right now the challenge is in recruiting people to  
> stay with the not-so-glorious aspects of the project, namely  
> documentation, maintenance, and janitorial style subprojects.
>
> Please keep the comments and suggestions coming!
>
> Thanks,
> MC
>
> On Tue, Sep 15, 2009 at 12:38 AM,  wrote:
> Hi Michael / FS enthusiast,
>
> In my opinion, it will be an undeniably perfect chaos if everyone  
> wants to do everything. One way
> I can suggest out of this is the following:
>
> - sets of persons that does all the stuff on the FS core. I'm not  
> sure if this one will work out but
> I think these are the only persons who has commit access to the  
> source code and will also be willing
> to accept modifications, enhancements, etc.
>
> - individuals can ask if they can be the maintainer/tester for a  
> particular module. Bug fixing,
> sending modifications and enhancements will still be subject to the  
> approval of anthm.
>
> - an array of persons that will take ownership on what FS can do and  
> provide realiable working
> examples for it. Currently, it takes amount of pain for a newbie on  
> how to configure SIP because
> the examples are too confusing and sometimes there is not certainty  
> if this will work on what
> release of FS or not.
>
> - any individual that has time and interest can start owning the  
> porting of FS to other operating
> system. Like in my case, I am working on making FS ported to FreeBSD  
> ports but I don't know if
> somebody did it or if he is currently doing it, as to what stage he  
> is in?
>
> - there should be a release engineering team. In my opinion, a  
> stable and current release will be
> much more sane.
>
> As with any other successful open source project, we won't be  
> starting to say I want to contribute
> this and start working on that. We should indicate that you take  
> ownership of this and any
> comments should be forwarded right unto you.
>
> Again, this is just my opinion. You can take some of it or leave it  
> using "sudo rm -rf blah/*" .
>
> There and back again,
> Demuel I. Bendano
> a.k.a engrxyz
>
>
> > Hi Michael,
> >
> > You can count with me for anything else, like documentation,
> > coding/scripting, or any other FreeSWITCH related stuff.
> >
> > Regards,
> >
> > Diego
> >
> > 2009/9/14 João Mesquita 
> >
> >> You can assign two things to me.
> >>
> >> 1. libesl code documentation (partially done and Doxygened - needs
> >> cleaning)
> >> 2. Bug marshal. I am setting up the proper lab environment here  
> to be able
> >> to test most stuff.
> >>
> >> Count me in for any questions I can answer and I am _always_ on IRC
> >>
> >> jmesquita
> >>
> >> On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins  >wrote:
> >>
> >>> Hello FreeSWITCHers!
> >>>
> >>> We are looking for people who are in a position to help out with  
> various
> >>> subprojects that will help FreeSWITCH to keep growing. We need  
> people to
> >>> help out in these basic areas:
> >>>
> >>> Bug marshals (people who watch JIRA and test bug reports,  
> patches, etc.)
> >>> Documentation maintainers (people who update the wiki when new  
> stuff comes
> >>> out, also those familiar with mediawiki administration)
> >>> Documentation authors (people who write new docs, how-to's,  
> tutorials,
> >>> examples, etc.)
> >>> Package maintainers (people who manage Debian debs, RPMs, etc.)
> >>>
> >>> Additionally, we are always looking for more folks to assist with
> >>> answering questions on IRC and the mailing list. It is  
> definitely nice to
> >>> have people who've gone through the pains of switching to  
> FreeSWITCH (or
> >>> learning it from scratch) who can assist the steady stream of  
> new users.
> >>>
> >>> If you want to help and aren't sure where to go from here then  
> please at
> >>> least do the following:
> >>> #1 - Join #freeswitch on irc.freenode.net and hang out as much as
> >>> possible
> >>> #2 - Check the recent changes link on wiki.freeswitch.org each day
> >>> #3 - Join the Friday public conference call and listen in
> >>> These three things, in addition to the mailing list, will keep  
> you well in
> >>> tune with the FreeSWITCH community and what's happening.
> >>>
> >>> Next, make a note of the parts of FS that you use frequently,  
> know a lot
> >>> about, or are particularly passionate about. Those are the items  
> we'd love
> >>> to have you help us with. For example: if you use mod_xml_curl  
> frequently
> >>> and have been through the set up process then you're a p

Re: [Freeswitch-users] Recording Only 1 Leg of a Call

[Seven Du]

As a work around, record to stereo, and use sox to split channes ?

On Sep 9, 2009, at 12:44 AM, Anthony Minessale wrote:
> that would have to be filed as a feature request as we do not  
> currently have a way to do that.
>
>
> On Mon, Sep 7, 2009 at 11:50 PM, Matthew Fong   
> wrote:
> I want to record without the telephone user's interaction.
>
> I think uuid_record should have the option to only record the audio  
> of the uuid channel that is being specified, and as a secondary  
> option combine the audio of the b leg (since uuid_record only  
> specifies one uuid anyway--this seems logical).
>
> Anyway, just my wish list :)
>
> --matt
> http://www.hellohunter.com
> voice broadcasting & hosted dialer
>
> On Tue, Sep 8, 2009 at 2:12 AM, Milena  wrote:
> Hello,
> What about this?:
>
> " 
>
>  "
>
> the person would have to press *2 during the call to start the  
> recording.
>
> 2009/9/7 Matthew Fong 
> Whats the best way to record only one leg of a call?
>
> uuid_record records both channels
> session_record does the same (but has a stereo option)
>
> is there any way to record only an a-leg of the call? Thanks so much.
>
> --matt
> http://www.hellohunter.com
> hosted dialer & voice broadcasting
>
> ___
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>
>
>
> ___
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>
>
>
> ___
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
> ___
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Re: [Freeswitch-users] mod_skypiax for OSX?????

[Seven Du]

On Sep 6, 2009, at 7:25 AM, Tapan Parikh wrote:
>
> Yes, thanks a lot!
>
> Im still having a bit of trouble getting it working though.  When  
> skypiax_proxy starts, I get the following errors:
>
> 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init  
> theDOProxy
> 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init  
> theDOProxy
>
I don't have that problem. google found this: 
https://developer.skype.com/jira/browse/SPA-418

> Then, when an incoming call comes in, I see the messages:
>
Please try set Auto-Answer on Skype Preference and retry.


> Message received  CALL 683 STATUS RINGING
> Sent 23 bytes to FreeSWITCH
> Sending to skype: GET CALL 683 PARTNER_HANDLE
> Send to skype: GET CALL 683 PARTNER_HANDLE
> Message received  CALL 683 CONF_ID 0
> Sent 18 bytes to FreeSWITCH
> Message received  CALL 683 FAILUREREASON 14
> Sent 25 bytes to FreeSWITCH
> Message received  CALL 683 STATUS MISSED
> Sent 22 bytes to FreeSWITCH
> Message received  CALL 683 SEEN FALSE
> Sent 19 bytes to FreeSWITCH
>
> But the call is not actually picked up by the proxy.  Any ideas?  Im  
> on Mac OS X 10.5, w/ freeswitch from SVN.  Thanks!
>
>
> On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli  > wrote:
> 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
> >>
> >> ___
> >> 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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Re: [Freeswitch-users] mod_skypiax for OSX?????

[Seven Du]

I'm not sure this and don't have time to debug. But last time I tried  
incoming calls worked by setting skype to Auto-answer calls.  Can you  
try that?

On Sep 6, 2009, at 3:19 PM, Ivan C Myrvold wrote:
> I have got outgoing call to Skype to work, and the audio quality is  
> excellent. But I also have problems with incoming call, looks like  
> it doesn't get bridged by FreeSWITCH. I put sofia on debug, but  
> couldn't see any Sofia messages at all. Only message I saw in the  
> console was this:
>
> 2009-09-05 20:44:39.720943 [WARNING] skypiax_protocol.c:375 rev  
> 14772[0x0|37 ][WARNINGA  375  ][interface1][-1, 0, 0]  
> skype_call: 332, STATUS: RINGINGCALL is not recognized
>
> But I am thrilled that Seven have got skypiax to work this far, and  
> am confident that we will have it worked both incoming and outgoing  
> soon.
>
> Ivan
>
> Den 6. sep.. 2009 kl. 01:25 skrev Tapan Parikh:
>
>>
>> Yes, thanks a lot!
>>
>> Im still having a bit of trouble getting it working though.  When  
>> skypiax_proxy starts, I get the following errors:
>>
>> 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init  
>> theDOProxy
>> 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init  
>> theDOProxy
>>
>> Then, when an incoming call comes in, I see the messages:
>>
>> Message received  CALL 683 STATUS RINGING
>> Sent 23 bytes to FreeSWITCH
>> Sending to skype: GET CALL 683 PARTNER_HANDLE
>> Send to skype: GET CALL 683 PARTNER_HANDLE
>> Message received  CALL 683 CONF_ID 0
>> Sent 18 bytes to FreeSWITCH
>> Message received  CALL 683 FAILUREREASON 14
>> Sent 25 bytes to FreeSWITCH
>> Message received  CALL 683 STATUS MISSED
>> Sent 22 bytes to FreeSWITCH
>> Message received  CALL 683 SEEN FALSE
>> Sent 19 bytes to FreeSWITCH
>>
>> But the call is not actually picked up by the proxy.  Any ideas?   
>> Im on Mac OS X 10.5, w/ freeswitch from SVN.  Thanks!
>>
>>
>> On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli > > wrote:
>> 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
>> >>
>> >> ___
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users@lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitc

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

[Seven Du]

gm,

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

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

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

-7-

On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote:
> Seeeven!
>
> I saw the modification you made on the wiki page...
>
> You made it, mod_skypiax runs on OSX
>
> Let's merge your mods on the mainline, plese ;-)))
>
> -giovanni
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
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Re: [Freeswitch-users] restart when convenient

[Seven Du]

freeswi...@foosball> fsctl
-USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap| 
restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration  
[num]|loglevel [level]]


On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote:
> After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing
> that all active calls are dropped and the freeswitch is restarted
>
>
>
> On Fri, Sep 4, 2009 at 10:14 AM, Michael Collins  
> wrote:
>>
>>
>> 2009/9/3 Christian Löschenkohl 
>>>
>>> hello
>>>
>>> i'm looking for a possibility to restart freeswitch like it is  
>>> possible
>>> with
>>> asterisk.
>>> for me i tried to created a script that looks for open channels  
>>> and if no
>>> channel
>>> is open it restarts freeswitch with the init script (not the most
>>> efficient way).
>>>
>>> i think i would be great if we would have a buildin function for  
>>> this, i
>>> think such
>>> command would help with maintenance and not only for me.
>>>
>>> br
>>>
>> Like JM said, the fsctl API can help. If you're in Linux you can do  
>> a shell
>> script with a command like ths:
>>
>> fs_cli -x 'fsctl shutdown restart'
>>
>> -MC
>>
>>
>> ___
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>>
>>
>
>
>
> -- 
> Anatoliy Kounitskiy
> -
> E-mail: anato...@kounitskiy.com
> Mobile: +359898913540
>
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Re: [Freeswitch-users] select batchfile after call

[Seven Du]

hangup hook api? system or system_bg ? not sure


On Sep 3, 2009, at 9:08 PM, NOx-WHV wrote:
>
> Hi,
>
> does anybody have a tip how to start a batchfile after hanging up.
>
> After ext. 1000 calls 1001 and hang up, i need a request to call:
>
> /../../FS/batchfile 1000
>
> if 1001 calls 1000 i need:
>
> /../../FS/batchfile 1001
>
> and so on...
>
>
> Thanks for help
> -- 
> View this message in context: 
> http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey

[Seven Du]

I run into this problem before. Don't remember the exact error but  
might be segfault of lame runing in freeswitch-lua.

If you use Linux you would like to try iwatch. It's a perl program  
watching your file system and can execute the lame command as soon as  
it got the CLOSE_WRITE(or other)  filesystem event.

On Aug 26, 2009, at 4:44 PM, Michael Jerris wrote:
> Running out of stack space?  The stack space we run freeswitch in is
> fairly small.  Programs launched from the freeswitch process inherit
> this.
>
> Mike
>
> On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote:
>
>> I ran strace from freeswitch and from the command line. lame  
>> segfaults
>> when run from system FS.
>>
>> The only obvious different i see is in the execve() /* XX vars */
>> apart
>> from the final Segfault
>>
>> From
>> execve("/usr/local/freeswitch/bin/lame",
>> ["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav", "/tmp/foo.mp3", "-
>> S"],
>> [/* 16 vars */]) = 0
>>
>>
>>> From FS
>> execve("/usr/local/freeswitch/bin/lame",
>> ["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav",
>> "/tmp/fooo.mp3", "-S"], [/* 14 vars */]) = 0
>>
>> I am attaching the full straces in case they are of any help. Not
>> sure if
>> this deserves a jira
>>
>> /aep
>> -- 
>> Stopping junk mailers is good for the environment
>>
>>> maybe it's writing some err to stderr that is being suppressed
>>> somehow
>>>
>>> On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
>>> aep.li...@it46.se> wrote:
>>>
 Hi Brian,

> From the CLI>

 freeswi...@open46> system /usr/local/freeswitch/bin/lame -V2
 /tmp/foo.wav
 /tmp/foo.mp3 -S
 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing
 command:
 /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
 API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
 /tmp/foo.mp3 -S)] output:
 +OK

 open46:/tmp# ls
 foo.wav


 and running the command from the command line:


 open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
 /tmp/foo.mp3
 -Sopen46:/tmp# ls
 foo.mp3  foo.wav


 If I do the same with lame397

 freeswi...@open46> system /usr/local/freeswitch/bin/lame397 -V2
 /tmp/foo.wav /tmp/foo.mp3 -S
 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing
 command:
 /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
 API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
 /tmp/foo.mp3 -S)] output:
 +OK

 open46:/tmp# ls
 foo.mp3  foo.wav


 Highly paranormal! Sorry for hijacking the previous thread.

 /aep

 --
 Stopping junk mailers is good for the environment

> Try running it at the CLI and see if you see any errors.  Also
> please
> do not hijack threads.  The original thread "[Freeswitch-users]
> XML-
> RPC on different ip than 0.0.0.0" which was hijacked by clicking
> reply, changing the subject and clicking send.  Please in the
> future
> do not do that as it clutters up the threading and could get your
> query lost in the noise.
>
> Thanks,
> Brian
>
> On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists)
> wrote:
>
>> Here it comes the mystery. I am use lame 3.98.2 the mp3 file  
>> never
>> appears, if I use version 3.97 (older version), it does!.
>
>
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>>>
>>>
>>> --
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>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
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>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:8...@conference.freeswitch.org >> %3a...@conference.freeswitch.org>
>>> iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] how to use freeswitch as a sip client

[Seven Du]

http://wiki.freeswitch.org/wiki/Mod_portaudio
http://wiki.freeswitch.org/wiki/FsAir


On Aug 18, 2009, at 11:22 PM, Tzury Bar Yochay wrote:
> Hi,
>
> Can someone instruct me (or point to a page where I can found this  
> information) how can freeswitch be used as a sip client.
> I would like to test a server transporting SIP packets over TCP and  
> was reported on this mailing list that it is possible to do so with  
> FS.
>
> thanks,
> Tzury
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Re: [Freeswitch-users] opal build error

[Seven Du]

Thanks, will try later.

On Aug 16, 2009, at 2:20 PM, Peter Olsson wrote:
> Make sure to do a complete rebuild. And also read the comments in  
> jira MODOPAL-10.
>
> /Peter
>
>
> On 09-08-16 03.51, "Seven Du"  wrote:
>
> Hi,
>
> According to wiki it still in development status, but should compile
> right? Any idea about this? thanks.
>
> make
>
> In file included from mod_opal.cpp:25:
> mod_opal.h:151: error: conflicting return type specified for 'virtual
> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
> /usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
> ptlib_virtual_function_changed_or_removed**
> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
> mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&,
> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
> switch_channel_t*)':
> mod_opal.cpp:564: error: no matching function for call to
> 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&,  
> NULL)'
> /usr/include/opal/opal/localep.h:290: note: candidates are:
> OpalLocalConnection::OpalLocalConnection(OpalCall&,
> OpalLocalEndPoint&, void*, unsigned int,
> OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
> 276: note:
> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
> mod_opal.cpp: In member function 'switch_status_t
> FSConnection::receive_message(switch_core_session_message_t*)':
> mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
> declared in this scope
> make[1]: *** [mod_opal.lo] Error 1
> make: *** [all] Error 1
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[Freeswitch-users] opal build error

[Seven Du]

Hi,

According to wiki it still in development status, but should compile  
right? Any idea about this? thanks.

make

In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for ‘virtual  
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)’
/usr/include/opal/opal/localep.h:267: error:   overriding ‘virtual  
ptlib_virtual_function_changed_or_removed**  
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)’
mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall&,  
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,  
switch_channel_t*)’:
mod_opal.cpp:564: error: no matching function for call to  
‘OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)’
/usr/include/opal/opal/localep.h:290: note: candidates are:  
OpalLocalConnection::OpalLocalConnection(OpalCall&,  
OpalLocalEndPoint&, void*, unsigned int,  
OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h: 
276: note:  
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
mod_opal.cpp: In member function ‘switch_status_t  
FSConnection::receive_message(switch_core_session_message_t*)’:
mod_opal.cpp:1037: error: ‘SWITCH_CHANNEL_SESSION_LOG’ was not  
declared in this scope
make[1]: *** [mod_opal.lo] Error 1
make: *** [all] Error 1
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Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, robustness patch

[Seven Du]

On Aug 16, 2009, at 12:02 AM, Giovanni Maruzzelli wrote:
> 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
>

Ok, as you think it's broken, better to leave it as is.
>> 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
>
I think maybe you merged in an old version of the patch, the last  
should be skype_status_new1.diff, which I added one more statistic  
line and the unsigned long should be uint32_t I think. however, I can  
add another patch if you think it's useful. Oh, one more, it also set  
channel name to skypiax/RR/sk_1/other_skype_name for easy check in log. 
(As I also would like to see sofia/external/ to be sofia/gateway/ 
gw/ :) ).

>>
>> 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
>

One of the reason I think it's useful is one can configure to load  
everything on server boot time.  I run two skypiax servers, one can  
start 20 instances in batch without any problem but the other only  
starts 50%, then I need manually start them over and over until I  
confirmed all works with client or skypiax_auth. The two servers are  
not in the same datacenter but all have public ip. Once it started  
working we never never experienced a skype instance stoped working.  
But I experienced that kill a skype instance immediately caused  
skypiax core dump.

Sure it might cause other bugs even it is configured to false by  
default. I gona merge in my branch in case others using that.

>> 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).
>

It by default disabled so I guess nothing will break.

> 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

[Seven Du]

Great works. I tested and reported results in jira.

And as I noticed you removed the sequential line hunting methods.  
Though I don't use that I think someone else may need that. Think  
about the guy want skypeout accounts in a round robin manner, others  
might use that in a priority manner, that means sequential search. So  
it maybe a good idea to add it back. At the same time I think "ANY" is  
not a good name for that method, so named it "SEQ" maybe better. Then  
"ANY" can be removed later by announcing here or keep there for  
backwards compatibility.

Thank you very much for merging in the sk list with statistic patch,  
and for the other two features, I found it's a little hard to split  
codes, so, can make two jira, and upload one patch file?

Two features are:

continue load on fail: make sure the module continue load even it  
failed to talk to a skype instance
auto skype user: get the user name by the returned CURRENTUSERHANDLE  
other than  from the config xml, for easier config.

Thanks.

-7-

On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn  
> trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
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[Freeswitch-users] 403 forbidden on files.freeswitch.org

[Seven Du]

Got this:

Forbidden
You don't have permission to access /cluecon_2009/presentations/ 
Dale_Building_FreeSWITCH_App_Lua.pptx on this server.

Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80

$ ping files.freeswitch.org
PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56 data bytes
64 bytes from 69.174.57.101: icmp_seq=0 ttl=46 time=224.900 ms
64 bytes from 69.174.57.101: icmp_seq=1 ttl=46 time=221.597 ms
^C

$ ping files-sync.freeswitch.org
PING www-01.freeswitch.org (216.82.231.69): 56 data bytes
64 bytes from 216.82.231.69: icmp_seq=0 ttl=44 time=271.147 ms
64 bytes from 216.82.231.69: icmp_seq=1 ttl=44 time=273.161 ms
^C


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Re: [Freeswitch-users] answer command

[Seven Du]

It's not Eyebeam but FS hung up the call because it have nothing to do  
after answer.

You should either playback a sound, do the echo command, record, hold  
the call, bridge to another channel or transfer somewhere else.

On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
>
> I've tried to use "answer" command from outbound event socket and it's
> working, but
> the problem is that FS answering the call, but SIP Client (we tried  
> this
> with EyeBeam and CISCO 7960)
> doesn't know that call was answered. So, as long as FS doesn't know  
> what to
> do with this number it then disconnects the call.
>
> 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
> [sofia/internal/sip:1...@10.107.181.160:42840] has been answered
> 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015  
> Channel
> [sofia/internal/1...@10.107.249.12] has been answered
> 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
> sofia/internal/sip:1...@10.107.181.160:42840 has no read codec.
> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
> sofia/internal/sip:1...@10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
> sofia/internal/1...@10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086  
> Session 133
> (sofia/internal/sip:1...@10.107.181.160:42840) Ended
>
>
> Maybe there is the way to acknowledge SIP client that call was  
> answered?
>
> Regards,
> Maxim Tsvetov
>
> Diego Viola wrote:
>>
>> I suggest that you learn the differences between mod_commands  
>> commands and
>> mod_dptools applications, and also the interfaces where you can  
>> access and
>> use them.
>>
>> As said before, mod_dptools is accessible from dialplan, event socket
>> outbound, etc. and mod_commands is accessible from the CLI, event  
>> socket
>> (inbound/outbound), XML RPC, etc.
>>
>> That's all described in the wiki I think.
>>
>> Let us know if you have any questions =D.
>>
>> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola 
>> wrote:
>>
>>> Michael, you're welcome :).
>>>
>>> Milena, answer is a mod_dptools command, you can use it from the XML
>>> dialplan or from the event socket outbound. mod_commands API are  
>>> APIs
>>> that
>>> you execute from the socket, event socket inbound, etc. But you  
>>> can also
>>> execute them from event socket outbound using the "api" command.
>>>
>>> I hope that makes sense, correct me if I'm wrong =D.
>>>
>>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>>> wrote:
>>>


 On Tue, Aug 11, 2009 at 9:05 AM, Milena   
 wrote:

>
> Hello Brian,
>
> I wanna fix the wiki, but to make sure i got it right, does it  
> only
> work
> on outbound event socket? or is there any other scenario where  
> it would
> work.
>

 FYI, Diego Viola fixed the wiki. (Thanks Diego!)
 -MC


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>>>
>>
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> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
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Re: [Freeswitch-users] problem when adding more extension

[Seven Du]

On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
> Hi,
>
> I wanted to add more extension to freeswitch.
> to add extension 1050 with password 1234 I did the following:
>
> $ cd /usr/local/freeswitch/conf/directory/default
>
> created 1050.xml having all '1000' strings replaced by '1050' by  
> typing
> $ sed s/1000/1050/g < 1000.xml > 1050.xml
>

just run reloadxml should be ok no need to rescan the profile

> rescan and reload the xml by typing into the CLI
>
> freeswi...@internal> sofia profile internal rescan reloadxml
>
> However, when I tried to login with these credentials I got the
> following in the fs_cli:
>
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
> SIP username 1050 does not match auth username
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
> sofia_reg_handle_register() Send challenge for
> [1...@server_address.net]
>
> below are the content of 1000 and 1050 xml files
>
> please advise.
>
>
> $ cat 1050.xml
> 
> 
>   
> 
> 
>   
>   
> 
> 
> 
> 
> 
>  value="$${outbound_caller_name}"/>
>  value="$${outbound_caller_id}"/>
> 
>   
> 
> 
>
> $ cat 1000.xml
> 
> 
>   
> 
> 
>   
>   
> 
> 
> 
> 
> 
>  value="$${outbound_caller_name}"/>
>  value="$${outbound_caller_id}"/>
> 
>   
> 
> 
>
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Re: [Freeswitch-users] answer command

[Seven Du]

answer only works on outbound event socket. why you don't answer in a  
dialplan? what's scenario you use this?

On Aug 11, 2009, at 9:31 PM, Maxim Tsvetov wrote:
>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>
>
> daqiang wang wrote:
>>
>> why not use:
>>session:answer()
>>
>>
>> 2009/8/11 Maxim Tsvetov 
>>
>>>
>>> Thank you.
>>>
>>> Is it possible to use this command from command line or via  
>>> event_socket
>>> interface?
>>>
>>>
>>>
>>> Szymon Olko wrote:

 Maxim Tsvetov pisze:
> Hello,
>
> I found in WIKI list of supported commands
> http://wiki.freeswitch.org/wiki/Mod_commands.
>
> I need command "answer" but it doesn't exist (Freeswitch 1.04,  
> Win32)
>
> That's what Freeswitch replies for this command:
>
> "answer: Command not found!"
>
> Сould you please help?

 Answer is in dialplan commands.

 http://wiki.freeswitch.org/wiki/Mod_dptools

 Szymon

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

[Seven Du]

2009/8/10 Ivan C Myrvold 

> Seven,
> I am afraid I will not be able to help you much with the Carbon code,
> as I am only good at Cocoa programming. You said you chose Carbon
> because you only needed low level API, and that is fair enough, but I
> will also add that you can do the same with only linking to the
> Foundation framework in Cocoa.
>

Link with Cocoa is OK but just need extra hack to the standard Makefile and
not so necessary. And I think Carbon code is more C friendly.
I made a jira to Skype-Dev, hope someone can help us.


>
> I looked a little at the diff file yesterday, and will investigate
> more today, to try to understand how you have done the Skype
> integration to the Freeswitch in the Carbon code.
> And I am glad that someone have contributed to get skypiax working in
> OS X. Great work so far!
>
> Ivan
>
> Den 9. aug.. 2009 kl. 20:02 skrev Seven Du:
>
> > Ivan,
> >
> > Good to know you are a cocoa dev. Unable to check in code right now,
> > will send the diff to you offlist for now.
> >
> > 0) I'm not familiar with Mac dev, just tried my best
> > 1) It doesn't work yet, but should be able to compile, sure you
> > already have the Skype framework in place :)
> > 2) if run the skype delegate from a threat, then cannot get event
> > callback. e.g. mac_client.c works but mac_client2.c doesn't. Since
> > skypiax is running in a thread, we need to figure out this first.
> > 3) it uses Carbon, since I think we only need to low level api, no
> > need to bother the complicate of Cocoa.
> > 4) strsep shows some warning on compile, haven't figured out why
> > 5) perhaps you should only add one interface in skypiax.conf.xml
> > 6) do you want to run multi-instances like on Linux?
> > 7) I really not sure if it will work or not :)
> >
> > Let me know if it helps. I bet you can make it work. Also code will be
> > in my branch soon.
> >
> > 7.
> >
> >
> > On Aug 9, 2009, at 11:34 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?
> &

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

[Seven Du]

On Aug 9, 2009, at 11:52 PM, Giovanni Maruzzelli wrote:
> Seven,
> thanks a lot for your effort, please let your stuff be available,
> maybe Ivan can make use of it!
>
svn diff http://svn.freeswitch.org/svn/freeswitch/branches/seven -r  
14473
:14475

When this done I think it's better to split codes into

skypiax_protocol.c
skypiax_protocol_mac.c
skypiax_protocol_linux.c
skypiax_protocol_windows.c

:)

7.
> 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

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

[Seven Du]

Ivan,

Good to know you are a cocoa dev. Unable to check in code right now,  
will send the diff to you offlist for now.

0) I'm not familiar with Mac dev, just tried my best
1) It doesn't work yet, but should be able to compile, sure you  
already have the Skype framework in place :)
2) if run the skype delegate from a threat, then cannot get event  
callback. e.g. mac_client.c works but mac_client2.c doesn't. Since  
skypiax is running in a thread, we need to figure out this first.
3) it uses Carbon, since I think we only need to low level api, no  
need to bother the complicate of Cocoa.
4) strsep shows some warning on compile, haven't figured out why
5) perhaps you should only add one interface in skypiax.conf.xml
6) do you want to run multi-instances like on Linux?
7) I really not sure if it will work or not :)

Let me know if it helps. I bet you can make it work. Also code will be  
in my branch soon.

7.


On Aug 9, 2009, at 11:34 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 -lcr

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

[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
>> 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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Re: [Freeswitch-users] Softphone control

[Seven Du]

You can run FreeSWITCH as a softphone and control it.
http://wiki.freeswitch.org/wiki/Freeswitch_softphone

2009/8/7 Artem Vasiliev 

> Hi
>
> I have FreeSwitch and external application, which communicates to it via
> event socket - listens for events for certain number and gives some
> commands.
> Is it possible for this application to control client softphones, for
> example, make them answer or hold, using the event socket or other
> FreeSwitch capabilities?
>
>
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Re: [Freeswitch-users] A few questions about lua

[Seven Du]

Sorry it's a typo. I read the code, it works not like in event socket. So,
only works with one event.
either
EventConsumer("all")
or
EventConsumer("CUSTOM", "lua::stop");


Thank you.

2009/8/7 Raffaele P. Guidi 

> >> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
> idea to this?
>
> isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
> code, too?
>
> On Thu, Aug 6, 2009 at 17:52, Seven Du  wrote:
>
>> for e in (function() return con:pop(1) end) do
>>
>> btw, the script works.
>>
>> Thanks.
>> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
>> > Hi
>> > I dont know about events so much but I cannot see variable "e" is
>> > setting
>> >
>> > event_name = e:getHeader("Event-Name") or ""
>> > event_subclass = e:getHeader("Event-Subclass") or ""
>> >
>> > regurds
>> > Eli
>> >
>> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> >> ALL-
>> >>
>> >>
>> >> I have a few questions when scripting lua. According to wiki, it is
>> >> possible to run looping forever lua scripts through start-up config
>> >> or luarun.
>> >>
>> >>
>> >> 1) Will the lua script stop when unload mod_lua? I experienced core
>> >> dump when unload mod_lua while there was a running lua script.
>> >> Reported on jira.
>> >>
>> >>
>> >> 2) How to stop a forever running lua script?  I stop it by
>> >> listening a CUSTOM event fired elsewhere. See code below. Is there
>> >> any standard way like luastop ?
>> >>
>> >>
>> >> 3) Any way to show how many running lua scripts? luashow ?
>> >>
>> >>
>> >> 4) It seems cannot get the lua script name in a lua script, I made
>> >> a patch to jira by assign it to the argv[0].
>> >>
>> >>
>> >> 5) Seems that only EventConsumer("all") working.
>> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
>> >> work. Any idea to this?
>> >>
>> >>
>> >> Thanks a lot.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> code example:
>> >>
>> >>
>> >> con = freeswitch.EventConsumer("all");
>> >>
>> >>
>> >> argv[0] = "test.lua"
>> >>
>> >> freeswitch.consoleLog("info", " Lua Script [" .. argv[0] .. "]
>> >> Starting =\n");
>> >>
>> >> local all_events = 0
>> >>
>> >> for e in (function() return con:pop(1) end) do
>> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>> >>all_events = all_events + 1;
>> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>> >>
>> >> event_name = e:getHeader("Event-Name") or ""
>> >> event_subclass = e:getHeader("Event-Subclass") or ""
>> >>
>> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>> >>  freeswitch.consoleLog("info", "-lua Script [" .. argv[0] ..
>> >> "]---Exiting--\n")
>> >>  break
>> >> end
>> >>
>> >>
>> >> end
>> >>
>> >>
>> > ___
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>> > FreeSWITCH-users@lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://lists.freeswitch.org/mailman/options/freeswitch-users
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Re: [Freeswitch-users] A few questions about lua

[Seven Du]

for e in (function() return con:pop(1) end) do

btw, the script works.

Thanks.
On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> Hi
> I dont know about events so much but I cannot see variable "e" is  
> setting
>
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
>
> regurds
> Eli
>
> On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> ALL-
>>
>>
>> I have a few questions when scripting lua. According to wiki, it is  
>> possible to run looping forever lua scripts through start-up config  
>> or luarun.
>>
>>
>> 1) Will the lua script stop when unload mod_lua? I experienced core  
>> dump when unload mod_lua while there was a running lua script.  
>> Reported on jira.
>>
>>
>> 2) How to stop a forever running lua script?  I stop it by  
>> listening a CUSTOM event fired elsewhere. See code below. Is there  
>> any standard way like luastop ?
>>
>>
>> 3) Any way to show how many running lua scripts? luashow ?
>>
>>
>> 4) It seems cannot get the lua script name in a lua script, I made  
>> a patch to jira by assign it to the argv[0].
>>
>>
>> 5) Seems that only EventConsumer("all") working.  
>> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to  
>> work. Any idea to this?
>>
>>
>> Thanks a lot.
>>
>>
>>
>>
>>
>> code example:
>>
>>
>> con = freeswitch.EventConsumer("all");
>>
>>
>> argv[0] = "test.lua"
>>
>> freeswitch.consoleLog("info", " Lua Script [" .. argv[0] .. "]  
>> Starting =\n");
>>
>> local all_events = 0
>>
>> for e in (function() return con:pop(1) end) do
>>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>>all_events = all_events + 1;
>> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>>
>> event_name = e:getHeader("Event-Name") or ""
>> event_subclass = e:getHeader("Event-Subclass") or ""
>>
>> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>>  freeswitch.consoleLog("info", "-lua Script [" .. argv[0] ..  
>> "]---Exiting--\n")
>>  break
>> end
>>
>>
>> end
>>
>>
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[Freeswitch-users] A few questions about lua

[Seven Du]

ALL-
I have a few questions when scripting lua. According to wiki, it is possible
to run looping forever lua scripts through start-up config or luarun.

1) Will the lua script stop when unload mod_lua? I experienced core dump
when unload mod_lua while there was a running lua script. Reported on jira.

2) How to stop a forever running lua script?  I stop it by listening a
CUSTOM event fired elsewhere. See code below. Is there any standard way like
luastop ?

3) Any way to show how many running lua scripts? luashow ?

4) It seems cannot get the lua script name in a lua script, I made a patch
to jira by assign it to the argv[0].

5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

Thanks a lot.



code example:

con = freeswitch.EventConsumer("all");

argv[0] = "test.lua"

freeswitch.consoleLog("info", " Lua Script [" .. argv[0] .. "] Starting
=\n");

local all_events = 0


for e in (function() return con:pop(1) end) do
  -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
   all_events = all_events + 1;
freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
 event_name = e:getHeader("Event-Name") or ""
event_subclass = e:getHeader("Event-Subclass") or ""
 if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
  freeswitch.consoleLog("info", "-lua Script [" .. argv[0] ..
"]---Exiting--\n")
  break
end

end
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Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to configure?)

[Seven Du]

mod_easyroute?

2009/8/6 Vladimir Rodionov 

> Hi, everybody
>
> This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
> assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
> VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
> possible in FS? If yes, how everything should be configuered? Dialplan, sip
> gateway? One more question: suppose it is doeable as I hope then how can I
> get in my script CalleeID (not a CallerID)? Basicaly,
>
> I want to acomplish the following:
>
> 1. Avoid re-configuring FS every time I got new bunch of DIDs
> assigned/released from/to my Voip provider.
> 2. Have a way of extracting CalleeID in my script.
>
> TIA,
>
> Vladimir Rodionov
>
>
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Re: [Freeswitch-users] How to distinguish between the different type of call rejections from Javascript?

[Seven Du]

variable_originate_disposition

On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> Hi,
>
> Say i originate a call to a mobile phone and the call fails. There  
> are many possible reasons: congestion, user busy, call rejected by  
> user, etc. Is there a way i can get the failure code from Javascript?
> Thanks,
> Max.
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Re: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing announcement

[Seven Du]

I think you can check the loopback endpoint or inline dialplan.

On Aug 4, 2009, at 3:23 AM, Michael Frager wrote:
> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to  
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement  
> feature that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/1551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the  
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
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Re: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story

[Seven Du]

And I  added this on the wiki page:
mod_conference and mod_fifo: We also use FreeSWITCH in our office
environment as a PBX for call center and customer service connected with
VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
naturally and just made sales process, business logic and world wide
conference much more simpler and easier.

:)

7

2009/8/5 Jonathan Palley 

> Diego -   Already done.  See the bottom of the page (we linked to another
> page because of its length)!
>
> :)
> Jonathan Palley
> Idapted Ltd.
>
>
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[Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story

[Seven Du]

Hello All -   In the spirit of ClueCon (which we are missing this year, but
hopefully not next), we wanted to document our "FreeSWITCH Story".  We've
posted it to the wiki(
http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
it is copied below.

Thank you all and enjoy a good conference!

Seven Du (seven)
Jonathan Palley (jpalley_idapted)
Idapted Ltd.


*How FreeSWITCH has created hundreds of job opportunities and changed
lives. *

We want to share our experience working with FreeSWITCH.  FreeSWITCH has
been a key enabler of our business.  We hope this story can be a small way
to say a very big THANK YOU ALL.

"Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
really allowed us to do just that.

What We Do:
We are not a telephony business; we are an educational technology and
service business. In Asia (China, in our case) students must pass English
examinations to study or work abroad and gain new experiences.  However,
there is limited access to native English speakers and the access students
can gain is typically very expensive.  At the same time, in the U.S., there
are many professionals looking for work-at-home opportunities - people who
need jobs and would create great teachers.  Through our technology and
content we empower these people to be effective English teachers.  Does it
work?  Yes.  The majority of our students are getting test scores that many
failed for years to get.  Just hours ago one student called one of our sales
agents crying with joy.  And for our teachers, they are now working in an
industry that was previously unavailable to those living in the U.S.
http://www.idapted.com

Why FreeSWITCH Enables This:
FreeSWITCH has been a key enabler of our business.  Recording calls,
controlling routing, integrating with various web-based interfaces, enabling
multiple endpoints - these are all key features of what we must do.  Most
importantly, setting up various servers and routes to mitigate cross-Pacific
and country-specific network challenges is key.  Doing what we are doing
with commercial solutions would have made the business unworkable.

Our Experiences with FreeSWITCH:
We started using FreeSWITCH as our VoIP Platform in April 2008, after
receiving unsatisfactory results with other open source solutions.  It took
one day of reading through the FreeSWITCH source code to know, "this is it.
 This is the VoIP platform we build our business on".  It took a few days of
working with the extremely competent and focused community to re-affirm this
commitment.

Our Setup:
Our teachers use a custom software that integrates a VoIP client with our
web based platform. Students connect to our teachers "on-demand".  Simply
put, on a web-based comet interface the student enters a phone number (or a
skype name or a gtalk account) and our platform bridges the best available
trainer and the student.  At the same time a web-based interface is being
updated.

The challenge for us is the connection between teachers and students over a
cross-continent network. For example, we experienced problems earlier this
year when a Asis-Pacific communication fiber broken... So, we've learned to
setup multi servers in multiple datacenters for redundancy.

We run multi instances of FreeSWITCH so we can always use the cutting edge
and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
only loads mod_dingaling. Here is one beauty of FS: We just had to create
different conf dirs (/usr/local/freeswitch, /usr/local/skype,
/usr/local/gtalk etc). This allows us to run the same code base over
different configurations, and call skype and gtalk accounts just like a
normal PSTN gateway (sofia/gateway/pstn/ or sofia/gateway/skype/ or
sofia/gateway/gtalk/ ). More important, if one FS (say FS-skype) behaves
abnormally or crashes, we can easily change to another FS-skype server (we
run other servers located in various places in China and HK for
redundancy).

FS --|
 |---PSTN gateways
 |--- FS-skype
 |--- FS-gtalk
 |--- FS-skype2
 |--- more ...



COMMUNITY:

The community's commitment cannot be undervalued.  The insightful, modular
design of FreeSWITCH allows anyone to contribute, whereever their skills
lie.  It also allows us to easily make modifications to the underlying code
to suit our specific use-cases  We want to highlight a few key people and
modules 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 tea

Re: [Freeswitch-users] Missing mod_curl

[Seven Du]

On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> I've gotten 1.0.4pre9 , but i can not see it :(
> -- afshin
>

In trunk.

>
> On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris   
> wrote:
> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali 
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
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Re: [Freeswitch-users] telnet to event socket

[Seven Du]

On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> Hi,
>
> I'm taking a close look at event socket on FS 1.0.3. Configuration is
> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> fsconsole.pl work but I was not able to telnet to port 8021. As I've
> done that I received somewhat like:
> #>auth/request
>
> I typed in: auth ClueCon
>
followed by two Enters (\n\n).

> After some seconds I've got the message 'connection close by foreign  
> host'
>
> Any ideas?
>
> Thank you
> Hoai-Anh
>
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Re: [Freeswitch-users] how to enable ESL for ruby?

[Seven Du]

Hi Brian,

Sorry responding late. I still cannot get this work, can you take a look?

http://pastebin.freeswitch.org/9877

Everything works fine on Linux but not on my MAC. I have the default ruby
framework and port install on /opt/local/bin/ruby, however, even I changed
the Makefile to use the default ruby framework, it just doesn't work though
the ESL.so compiled successfully. Where I'm wrong?

Thanks.



2009/5/27 Brian West 

>
> On May 26, 2009, at 11:20 AM, dujinfang wrote:
>
> Thanks Brain. Got ESL.so, however on my Mac it is #include  instead
> of .
>
>
> Actually since we do -framework Ruby it should be ruby/ruby but I think the
> line above the -framework Ruby should be removed since you're doing i tthe
> Mac way.
>
> /b
>
>
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Re: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped

[Seven Du]

Thanks for the great work.

Just want you know that 20 channels with the same username works well on my
server. And echo() works without any problem.

An updated version of Round Robin hunt and a minor bug posted on jira.

Thanks again.

2009/7/27 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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[Freeswitch-users] FreeSWITCH-Air, Another GUI?

[Seven Du]

ALL,

I know you guys more prefer a CLI version of softphone to a GUI  
version. But I still would like to share this:

http://wiki.freeswitch.org/wiki/FsAir

And feel free to give me feedbacks.

I'v only played a few days of ActionScript, it's highly appreciated if  
someone can give me help on the following problems.


1) Bounce the icon on Mac on incoming call.
2) Show a small window on incoming call.
3) Is it possible to block read/write a socket? I want to implement  
sendRecv() in ActionScript like in the C version of ESL.

Thanks.

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Re: [Freeswitch-users] ERROR in Sofia internal profile

[Seven Du]

chances are the tcp/udp port (5060?) already used by other software,  
are you running softphone on the same computer?

On Jul 11, 2009, at 4:29 PM, velusamy velu wrote:
> Dear Friends,
>  When I reload the mod_sofia I have got the following error.
>
> "2009-07-11 13:19:32 [ERR] sofia.c:739 sofia_profile_thread_run()  
> Error Creating SIP UA for profile: internal"
>
> Please any one explain about this error and please give any  
> suggestions to solve this problem..
>
> Thanks in Advance..
>
> Regards,
> K.Velusamy
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Re: [Freeswitch-users] mod_fifo: Selection of consumers in a certain order.

[Seven Du]

On Jul 7, 2009, at 11:25 PM, Luis F Urrea wrote:
> Hi all,
>
> We are in the need of a certain application using mod fifo.  
> Basically we are doing the following as described in the wiki:
> 
>
>lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1...@$$ 
> {domain}
>lag="5">{call_timeout=30,fifo_member_wait=nowait}user/1...@$$ 
> {domain}
>
> 
>
> 
>   
>
>
>   
>   
>   
>
>   
>   
>   
>
>   
> 
>
> Things work fine, but we have noticed that the consumers are  
> selected in kind of a load balancing basis, so that if member 1009
> answered the last call then the next call goes to 1008 even if 1009  
> is available.
>
>
> We would like to know if there is a way to select the next consumer  
> available in a different fashion such as in strict order setting a  
> preference
> for each member.
No. Maybe you'd like to patch it or add a wishlist to jira or add a  
bounty.

But, depending on how many agents you have, a '|' separated dialstring  
might do the trick:

>  lag="5">{leg_timeout=30,fifo_member_wait=nowait}user/1009|user/1008| 
> user/1007...

>
>
> Say
> member 1 preference=1
> member 2 preference=2
>
>
> so that only if member 1 is on the phone the call rolls to member 2.
>
> Hope it makes sense.
>
> All input on how to achieve this is appreciated.
>
> Regards,
>
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Re: [Freeswitch-users] Jingle (Mod_Dingaling Dialplan Extention

[Seven Du]

for outbound:
 
   
 
   
 

for inbount:
you should be able to set context and extension in your dingaling  
profile, and set the dialplan accordingly. as always you can press F8  
on console to see dialplan information

On Jul 2, 2009, at 10:19 PM, Meftah Tayeb wrote:
> hello,
> i asked about mod_dingaling usage befor
> finally, i configured my XMPP acount with my FS, connected to it and  
> is
> ready / Online
> now i need to call / recev call from / to GTalk (Jingle)
> cool anyone give me a sample Dialplan Extention?
> thanks
>
>
> __ Information from ESET NOD32 Antivirus, version of virus  
> signature database 4209 (20090702) __
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
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Re: [Freeswitch-users] how to build on MAC using /opt/local/include/tiffio.h

[Seven Du]

Thanks. I installed libtiff from source to /usr/local/ and now it works.

On Jun 30, 2009, at 11:49 PM, Michael Jerris wrote:

> I think that detection is not working on mac right due to it looking
> in default search paths.  I am in process of fixing this to use in
> tree libtiff soon so this should fix this issue.
>
> Mike
>
> On Jun 29, 2009, at 11:08 PM, seven wrote:
>
>> Hi,
>>
>> I'm on the latest svn 14041, and I have tiff installed with "port
>> install tiff", how can I tell FS to find libtiff at /opt/local/
>> include?
>>
>> on Linux the package should be libtiff-dev of libtiff-devel, but I
>> think tiff is the equivalent on Mac.
>>
>> previous versions build ok on my Mac, possible to include the libtiff
>> in trunk like other libs?
>>
>> checking tiffio.h usability... no
>> checking tiffio.h presence... no
>> checking for tiffio.h... no
>> checking pthread.h usability... yes
>> checking pthread.h presence... yes
>> checking for pthread.h... yes
>> checking X11/X.h usability... yes
>> checking X11/X.h presence... yes
>> checking for X11/X.h... yes
>> checking for libxml/xmlmemory.h... checking libxml/xmlmemory.h
>> usability... yes
>> checking libxml/xmlmemory.h presence... yes
>> checking for libxml/xmlmemory.h... yes
>> checking libxml/parser.h usability... yes
>> checking libxml/parser.h presence... yes
>> checking for libxml/parser.h... yes
>> checking libxml/xinclude.h usability... yes
>> checking libxml/xinclude.h presence... yes
>> checking for libxml/xinclude.h... yes
>> checking FL/Fl.H usability... no
>> checking FL/Fl.H presence... no
>> checking for FL/Fl.H... no
>> checking FL/Fl_Overlay_Window.H usability... no
>> checking FL/Fl_Overlay_Window.H presence... no
>> checking for FL/Fl_Overlay_Window.H... no
>> checking FL/Fl_Light_Button.H usability... no
>> checking FL/Fl_Light_Button.H presence... no
>> checking for FL/Fl_Light_Button.H... no
>> checking FL/fl_draw.H usability... no
>> checking FL/fl_draw.H presence... no
>> checking for FL/fl_draw.H... no
>> checking FL/Fl_Cartesian.H usability... no
>> checking FL/Fl_Cartesian.H presence... no
>> checking for FL/Fl_Cartesian.H... no
>> checking FL/Fl_Audio_Meter.H usability... no
>> checking FL/Fl_Audio_Meter.H presence... no
>> checking for FL/Fl_Audio_Meter.H... no
>> checking for cos in -lm... yes
>> checking for library containing sinf... none required
>> checking for library containing cosf... none required
>> checking for library containing tanf... none required
>> checking for library containing asinf... none required
>> checking for library containing acosf... none required
>> checking for library containing atanf... none required
>> checking for library containing atan2f... none required
>> checking for library containing ceilf... none required
>> checking for library containing floorf... none required
>> checking for library containing powf... none required
>> checking for library containing expf... none required
>> checking for library containing logf... none required
>> checking for library containing log10f... none required
>> checking for TIFFOpen in -ltiff... no
>> configure: error: "Can't build without libtiff (does your system
>> require a libtiff-devel package?)"
>> configure: error: ./configure.gnu failed for libs/spandsp
>>
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Re: [Freeswitch-users] mod_dingaling no audio

[Seven Du]

> 3)   is not  
> really required at least for my working setup behind the NAT router.
>

ok, this param is originally added for another problem 
http://jira.freeswitch.org/browse/MODENDP-198 
. But I think it might be useful for this.

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Re: [Freeswitch-users] How to change outbound_caller_name=FreeSWITCH and number=000000000

[Seven Du]

depending how do you make out going call.

On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote:

>
> Actually the extension_caller_id=Extension 1001 and
> extension_caller_number=1001 is set as Harmeet says but the same issue
> FreeSwitch the caller name and the number is 000 i just want  
> 1001 the
> caller number and the id
>
> Edmar
>
> Edward Q. wrote:
>>
>> Sorry Edmar
>>
>> I missundertood you .. I thought you wanted to change the number  
>> showing
>> once you were going out not the 1001.xml file.
>> In this case Harmeet is right. There you have those values to to  
>> make the
>> changes.
>> My bad.
>> Ed
>>
>> On Tue, Jun 23, 2009 at 12:46 AM, Harmeet Singh 
>> wrote:
>>
>>> In my case the 1001 resides in -
>>> /usr/local/freeswitch/conf/directory/default/1001.xml
>>>
>>> And you set the Caller Name and ID by  adding -
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Jun 23, 2009 at 12:26 AM, Edward Q.   
>>> wrote:
>>>
 Edmar

 Eso esta en freeswitch/conf/vars.xml  en ese archivo.
 If i am not mistaken and anyone welcome to correct me i just told  
 Edmar
 this is set in freeswitch/conf/vars.xml ... file
 Ed


 On Tue, Jun 23, 2009 at 12:02 AM, Edmar Cruz
 wrote:

>
> When I calling an outbound extension it appears:
>
> name is FreeSWITCH and number is 0
>
> How can i change it depends on the user who is calling?
>
> Sample 1001->64521223
>
> I just want the name 1001 to appear not FreeSWITCH same as the  
> number
>
> Thanks
>
>
>
> --
> View this message in context:
> http://www.nabble.com/How-to-change-outbound_caller_name%3DFreeSWITCH-and-number%3D0-tp24158823p24158823.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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>>>
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>
> -- 
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>
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Re: [Freeswitch-users] How can I join two freeswitch on two servers?

[Seven Du]

comment lines in the user directory do the trick:

   
   
   
   


On Jun 17, 2009, at 12:26 PM, Edmar Cruz wrote:

>
> If FS A has an account 8011105 does FS B also nid to register  
> 8011105? Yes it
> working on a gateway but the username of the gateway was shown on my
> softphone and also it nids a password for the gateway... is there an  
> option
> to view the caller name and number of the FS A gateway to FS B?
>
>
>
>
> Brian West-3 wrote:
>>
>> COPY paste fail :)
>>
>>
>>
>> something like that as per the example.
>>
>> /b
>>
>> On Jun 16, 2009, at 11:02 PM, Edmar Cruz wrote:
>>
>>>
>>> How can sofia profile can call ACL?
>>> Can you give me an example?
>>> Like this?
>>>
>>> I put this on external profile
>>>
>>> "/>
>>> "/>
>>>
>>>
>>> Brian West-3 wrote:

 Now you have to tell the sofia profile to use that ACL

 /b

 On Jun 16, 2009, at 10:03 PM, Edmar Cruz wrote:

> How can i turn off authentication? This is my acl.conf.xml on
> 192.168.0.105
>
> 
> 
> 
>
> On 192.168.0.4
>
> 
> 
> 


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>>>
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>>>
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Re: [Freeswitch-users] Zoiper reject freeswitch calls

[Seven Du]

What's wrong of the contact string? 639(snom) works but 637(zoiper)  
doesn't.


"user" >


"seven" >



On Jun 16, 2009, at 8:43 PM, Brian West wrote:

Ok i'll have to se what I can do about reproducing this issue now  
that I have more info on how its happening.


/b

On Jun 16, 2009, at 7:40 AM, dujinfang wrote:


Almost caught you on IRC Mike.

Our server is in a NAT'd network and all agents registered in the  
same LAN. I can remotely register by using the public IP and the  
contact string is right.


Call-ID:ZTZhMGJkZTE0NzNjZTlmZTkxYmU5NWRlZTU1MzJlYTE.
User:   6...@192.168.1.16
Contact:"user" >

Agent:  Zoiper rev.1809

So it's like only happens on our LAN and where there's a fs_path  
present.


Just curious, why agents registered on a local LAN has param  
fs_nat=yes; (default internal profile, port 5060) ?


Seems our time doesn't match, I'm generally available in office  
9:00AM-6:00PM GMT+0800, so will try to catch you tomorrow.


Thank you.


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