Re: [Freeswitch-users] spidermonkey problems

[Michael Collins]

FYI, I opened http://jira.freeswitch.org/browse/MODLANG-97 on this issue.
-MC

On Fri, Jan 16, 2009 at 9:03 AM, Michael Jerris  wrote:
> All long running non js code should be wrapped in the suspend/resume gc
> stuff.  For example:
> cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE);
> cb_state.saveDepth = JS_SuspendRequest(cx);
> args.input_callback = dtmf_func;
> args.buf = bp;
> args.buflen = len;
> switch_ivr_sleep(jss->session, ms, sync, &args);
> JS_ResumeRequest(cx, cb_state.saveDepth);
> I think this is your issue.  Can you please file a bug on jira for this
> issue (even better with a patch)
> Mike
>
>
> On Jan 16, 2009, at 5:54 AM, Jonas Gauffin wrote:
>
> I've found the problem. one js thread wait in socket.read
> (mod_spidermonkey_socket) on data.
> That caller have hangup, which means that the garbage collector waits on it
> to close.
>
> All new javascript sessions waits in JS_AWAIT_GC_DONE for the garbage
> collector to be done before proceeding (which means that all new javascript
> calls don't do anything after being launched).
> My server will not send anything until an agent gets free or the session
> hangs up (detects it through the event socket). And the event socket will
> not send that the session has been hangup until the socket have received
> anything (and the script can exit). So it's kind of deadlock between my
> server and the spidermonkey_socket.
> Is it possible to add an option to socket.read to make it abort if the
> session have been closed? I know that I wrote mod_spidermonkey_socket from
> the start, but I can't figure out how to do it.
> Will new sessions always wait on old ones to be garbage collected properly?
> For instance, what happens if a script have a lenghty post process after
> caller have hang up?
> On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin 
> wrote:
>>
>> I've got a loop, but the first thing checked in each iteration is if
>> session.ready() returns false (and in that case exit the loop).
>> I do create sessions in the script: create, try to originate to a
>> destination and then finally bridge together the caller and the new session.
>> I'll try to give you more details during the day.
>> On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale
>>  wrote:
>>>
>>> do you have any loops in your code that might not check for
>>> session.ready() in a exit when its not true.
>>>
>>> The symptoms you posted would be consistent with held readlocks so if you
>>> got a gcore (or windows equiv) of the process you might be able to see what
>>> threads where doing what to hang on to the read lock.
>>>
>>> also are you creating sessions in the script then executing app with
>>> them, beware of this because the thread of the script is used to execute
>>> apps on a session created that way and not the session thread.
>>>
>>>
>>>
>>>
>>> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin 
>>> wrote:

 Hello
 I got problems with hanging spidermonkey sessions and need some advice
 on how to debug them.
 I've made a javascript queue application that uses
 mod_spidermonkey_socket. It works fine for a while,
 but after some calls I noticed that calls didnt get transferred to
 agents. The reason was that earlier
 calls had not been terminated properly.
 freeswi...@test1> hupall
 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147
 switch_core_session_hupall() Giving up with 8 sessions remaining
 API CALL [hupall()] output:
 +OK hangup all channels with cause MANAGER_REQUEST

 freeswi...@test1> show calls
 API CALL [show(calls)] output:
 0 total.

 As you can see, 8 sessions are alive, but none of them are listed as
 calls. What kind of logs should I turn on to see what is happening with
 those sessions?
 Thanks,
   Jonas
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>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_miness...@hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> 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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>>
>
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Re: [Freeswitch-users] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH

[Michael Collins]

Andrew,

On behalf of the OSS telephony community, and particularly the
FreeSWITCH community, many thanks to you! We appreciate all the stuff
you've given back. Thanks for showing the true Open Source spirit!
-MC

On Fri, Jan 16, 2009 at 3:35 PM, Andrew Thompson  wrote:
> I'd like to announce the first beta release of a cross-platform ruby/tk
> GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's
> not particularly fancy, but I needed a cross platform softphone with
> good voice quality that was debuggable and didn't have a ton of
> features to confuse the users. I couldn't find one so we built one.
>
> I've got some sparse documentation up at:
>
> http://opencsm.org/wiki/index.php/Spice_SoftPhone
>
> And you can download it from http://opencsm.org/download . It's under
> the MPL and I've been cleared to re-licence my other FreeSWITCH related
> projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris
> and OSX (it used to work on linux, I assume it still does).
>
> Comments/complaints/bugreports welcome. It's definitely still got some
> rough spots (I don't think it'll run without a controlling terminal, for
> example), but we're going to be polishing it up and hopefully putting it
> in production here in the next few weeks to replace a very buggy
> closed-source phone we've had to endure far too long.
>
> Please download it if you're interested, the download count helps us
> continue working on this kind of stuff :)
>
> Andrew - opencsm.org
>
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[Freeswitch-users] [ANN] Spice SoftPhone, a softphone GUI for FreeSWITCH

[Andrew Thompson]

I'd like to announce the first beta release of a cross-platform ruby/tk
GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's
not particularly fancy, but I needed a cross platform softphone with
good voice quality that was debuggable and didn't have a ton of
features to confuse the users. I couldn't find one so we built one.

I've got some sparse documentation up at:

http://opencsm.org/wiki/index.php/Spice_SoftPhone

And you can download it from http://opencsm.org/download . It's under
the MPL and I've been cleared to re-licence my other FreeSWITCH related
projects under the MPL too. I've tested it on Windows, FreeBSD, Solaris
and OSX (it used to work on linux, I assume it still does).

Comments/complaints/bugreports welcome. It's definitely still got some
rough spots (I don't think it'll run without a controlling terminal, for
example), but we're going to be polishing it up and hopefully putting it
in production here in the next few weeks to replace a very buggy
closed-source phone we've had to endure far too long.

Please download it if you're interested, the download count helps us
continue working on this kind of stuff :)

Andrew - opencsm.org

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Re: [Freeswitch-users] Dialing Out Problem via Gateway

[Brian West]

I would suspect the NAT wasn't punching holes or lied.  :)

/b

On Jan 16, 2009, at 3:57 PM, Will Smith wrote:

Well, if NAT involved, why did I get through after I put the call on  
hold and take the call back. I am getting the SIP trace, hope that  
will show something.

Thank you all


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Re: [Freeswitch-users] Dialing Out Problem via Gateway

[Will Smith]

Well, if NAT involved, why did I get through after I put the call on hold and 
take the call back. I am getting the SIP trace, hope that will show something.
Thank you all

--- On Fri, 1/16/09, Brian West  wrote:

From: Brian West 
Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway
To: freeswitch-users@lists.freeswitch.org
Date: Friday, January 16, 2009, 1:41 PM


NAT involved?


/b



On Jan 16, 2009, at 3:30 PM, Will Smith wrote:







Thank you Brian,
 
The problem is very simple, I or the other party cannot hear each other when I 
first dial and the other party picks up the phone. We hear the phone ring, the 
other end picks up the phone says something, but I cannot hear - nothing, even 
static. Same thing happen on my end, I say something, and the other end do  not 
hear a thing. When I  put the call on hold,  the other end can hear music on 
hold. When I take the call back, now we can talk. Something does not go through 
when the other end picks up the call.
This is the extension in the dialplan/default.xml
 







    
  
  
  
  
 
   

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Re: [Freeswitch-users] Dialing Out Problem via Gateway

[Brian West]

NAT involved?

/b

On Jan 16, 2009, at 3:30 PM, Will Smith wrote:


Thank you Brian,

The problem is very simple, I or the other party cannot hear each  
other when I first dial and the other party picks up the phone. We  
hear the phone ring, the other end picks up the phone says  
something, but I cannot hear - nothing, even static. Same thing  
happen on my end, I say something, and the other end do  not hear a  
thing. When I  put the call on hold,  the other end can hear music  
on hold. When I take the call back, now we can talk. Something does  
not go through when the other end picks up the call.

This is the extension in the dialplan/default.xml



  data="effective_caller_id_number=1222333"/>
  
  
  
 
   



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Re: [Freeswitch-users] Vmware voice quality

[jeff sacksteder]

This is a known issue with all virtualization solutions. The realtime
clocks inside the Virtual Machine jitter quite a bit which causes
havoc with the udp media streams. I have never heard of someone using
any VoIP product inside a VM and being happy with the result. To the
best of my knowledge, all the current VM products suffer from this -
Hyper-V, Xen and VMware, VMWare being the worst.

Xen shares access to the system clock with paravirtualized  guests
directly, so you might find that to be somewhat better than VMWare.

More here - http://www.vmware.com/pdf/vmware_timekeeping.pdf

The performance of the server is irrelevant. Google for UDP jitter in
combination with VMWare.

On Thu, Jan 15, 2009 at 12:08 PM, Remko Kloosterman
 wrote:
> Hello Ken, hello all,
>
> I just read about the FreeSWITCH VMware applicance. I'm curious about your
> experiences with the audio quality on VMWare, so here's a new thread.
>
> I've installed freeswitch on VMware Server for Windows. The IVR audio always
> plays choppy, while the server itself has no performance issues. The same
> poor voice quality also goes for Asterisk or Yate, even on a very fast
> VMware ESX system.
>
> Did you experience the same and/or do you have pointers on how to
> troubleshoot and fix this?
>
> Thanks,
>
> Remko
>
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Re: [Freeswitch-users] Dialing Out Problem via Gateway

[Will Smith]

Thank you Brian,
 
The problem is very simple, I or the other party cannot hear each other when I 
first dial and the other party picks up the phone. We hear the phone ring, the 
other end picks up the phone says something, but I cannot hear - nothing, even 
static. Same thing happen on my end, I say something, and the other end do  not 
hear a thing. When I  put the call on hold,  the other end can hear music on 
hold. When I take the call back, now we can talk. Something does not go through 
when the other end picks up the call. 
This is the extension in the dialplan/default.xml
 



if (typeof YAHOO == "undefined") {
 var YAHOO = {};
}
YAHOO.Shortcuts = YAHOO.Shortcuts || {};
YAHOO.Shortcuts.hasSensitiveText = false;
YAHOO.Shortcuts.sensitivityType = [];
YAHOO.Shortcuts.doUlt = false;
YAHOO.Shortcuts.location = "us";
YAHOO.Shortcuts.document_id = 0;
YAHOO.Shortcuts.document_type = "";
YAHOO.Shortcuts.document_title = "t";
YAHOO.Shortcuts.document_publish_date = "";
YAHOO.Shortcuts.document_author = "willbel...@yahoo.com";
YAHOO.Shortcuts.document_url = "";
YAHOO.Shortcuts.document_tags = "";
YAHOO.Shortcuts.document_language = "english";
YAHOO.Shortcuts.annotationSet = {
"lw_1232141334_0": {
"text": "9054516117",
"extended": 0,
"startchar": 391,
"endchar": 400,
"start": 391,
"end": 400,
"extendedFrom": "",
"predictedCategory": "",
"predictionProbability": "0",
"weight": 1,
"relScore": 0,
"type": ["shortcuts:/us/instance/identifier/fedex_tracking"],
"category": ["IDENTIFIER"],
"wikiId": "",
"relatedWikiIds": [],
"relatedEntities": [],
"showOnClick": [],
"context": "",
"metaData": {
"verified": "false",
"visible": "true"
}
 }
};
YAHOO.Shortcuts.headerID = "a9059b1f35336b4363b0c75035b61d07";







    
  
  
  
  
 
   

--- On Fri, 1/16/09, Brian West  wrote:

From: Brian West 
Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway
To: freeswitch-users@lists.freeswitch.org
Date: Friday, January 16, 2009, 1:13 PM


Can you detail your problem a bit more? 


/b



On Jan 16, 2009, at 3:09 PM, Will Smith wrote:






Hi,
I got a strange problem that I don't really understand, and I hope that you 
could give me some hint how to fix that:
 
When I dial out through a gateway that is defined in the 
sip_profiles/external , (The xml file is simple as below. )  I cannot talk or 
hear from the other end. But when I put the line on hold, two ends can hear 
music, and when open the line again, this time 2 ends can hear and talk. Is 
there any where that I can fix this problem? Thank you
 

  
    
    
    
    
    
    
  


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Re: [Freeswitch-users] Dialing Out Problem via Gateway

[Michael Collins]

A SIP trace would be extremely helpful.
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Enabling_SIP.2FSofia_Tracing
-MC

On Fri, Jan 16, 2009 at 1:13 PM, Brian West  wrote:
> Can you detail your problem a bit more?
> /b
> On Jan 16, 2009, at 3:09 PM, Will Smith wrote:
>
> Hi,
> I got a strange problem that I don't really understand, and I hope that you
> could give me some hint how to fix that:
>
> When I dial out through a gateway that is defined in the
> sip_profiles/external , (The xml file is simple as below. )  I cannot talk
> or hear from the other end. But when I put the line on hold, two ends can
> hear music, and when open the line again, this time 2 ends can hear and
> talk. Is there any where that I can fix this problem? Thank you
>
> 
>   
> 
> 
> 
> 
> 
> 
>   
> 
>
>
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Re: [Freeswitch-users] Dialing Out Problem via Gateway

[Brian West]

Can you detail your problem a bit more?

/b

On Jan 16, 2009, at 3:09 PM, Will Smith wrote:


Hi,
I got a strange problem that I don't really understand, and I hope  
that you could give me some hint how to fix that:


When I dial out through a gateway that is defined in the  
sip_profiles/external , (The xml file is simple as below. )  I  
cannot talk or hear from the other end. But when I put the line on  
hold, two ends can hear music, and when open the line again, this  
time 2 ends can hear and talk. Is there any where that I can fix  
this problem? Thank you



  






  




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[Freeswitch-users] Dialing Out Problem via Gateway

[Will Smith]

Hi,
I got a strange problem that I don't really understand, and I hope that you 
could give me some hint how to fix that:
 
When I dial out through a gateway that is defined in the 
sip_profiles/external , (The xml file is simple as below. )  I cannot talk or 
hear from the other end. But when I put the line on hold, two ends can hear 
music, and when open the line again, this time 2 ends can hear and talk. Is 
there any where that I can fix this problem? Thank you
 

  
    
    
    
    
    
    
  



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Re: [Freeswitch-users] Starting FS on OSX (10.4.11 PPC)

[Ivan C Myrvold]

I have chown the  freeswitch directory to my user imyrvold, therefore  
I put it in ~/Library/LaunchDaemons.
Do you run freeswitch as root, as you put it in /System/library/ 
LaunchDaemons? That directory should be reserved anyway for Apple's  
system tools.
A better idea would be to put it in /Library/LaunchDaemons if you run  
it as root (but you should in my opinion run it as a normal user, as I  
have (almost) always done).

Ivan

Den 16. jan.. 2009 kl. 20:12 skrev Martin Joseph:

>
> On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote:
>
>> I haven't tried using launchctl for FreeSWITCH. But when I saw your
>> post, I tried it out. I have no problem getting it to work:
>>
>> I make a file "org.freeswitch.freeswitch.plist" and save it to ~/
>> Library/LaunchAgents with the following content:
>>
>> 
>> > "http://www.apple.com/DTDs/PropertyList-1.0.dtd
>> ">
>> 
>> 
>>  KeepAlive
>>  
>>  Label
>>  org.freeswitch.freeswitch
>>  Program
>>  /usr/local/freeswitch/bin/freeswitch
>>  RunAtLoad
>>  
>>  ServiceIPC
>>  
>> 
>> 
>>
>> Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/
>> org.freeswitch.freeswitch.plist"
>>
>> If you do the same command, but unload instead of load, it should  
>> stop
>> freeswitch.
>>
>> Does this work for you?
>
> Huh,
>
> I have been trying to do something similar, but putting the file in /
> System/Library/LaunchDaemons.
>
> Strangely, I tried your file right now (in LaunchDaemons) and
> freeswitch is started, but it doesn't respond to my devices trying to
> register?
>
> Weird.
>
> If I kill it and start it manually, it immediately responds and my
> devices register.
>
> Ideas?
> Thanks,
>
> Marty
>>
>>
>> Ivan
>>
>> Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph:
>>
>>> Hello again FreeSwitchers,
>>>
>>> I have built the 1.02 on 10.4.11(OSX) and had no problems with that.
>>>
>>> I have never been able to build from the SVN, but that is another
>>> story.
>>>
>>> Now that I have migrated to 1.02 I was wondering if I can get some
>>> help on a long standing issue I have with starting FS at boot.
>>>
>>> I am hoping to use Launchd which is the standard on OSX 10.4 and I
>>> attempted to cobble together a script, but haven't had great  
>>> results.
>>>
>>> I did search for wiki entries on this, but haven't found any help
>>> with
>>> it.
>>>
>>> Ideas?
>>> Thanks,
>>> Marty
>>>
>>>
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>>
>>
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Re: [Freeswitch-users] Starting FS on OSX

[Martin Joseph]

On Jan 16, 2009, at 5:45 AM, Shido Xavier wrote:

> Please specify Intel or PPC.

Very good point and I had had that thought as well.

I am on PPC 10.4.11.

Thanks for any help or ideas.
Marty

>
>
> -Greg M.
>
>
> On Fri, Jan 16, 2009 at 1:29 AM, Martin Joseph  
>  wrote:
>>
>> On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote:
>>
>>> Your build issue is with your autotools install, I have seen  
>>> issues if
>>> you have ever installed any of the autotools from macports or fink.
>> I have never used Fink or Macports so that isn't it.  In fact the
>> supposed statements made to the effect that FS will build from SVN
>> fine on 10.4 with the latest available apple dev tools is quite wrong
>> in my experience. I setup a virgin 10.4 and updated everything and  
>> had
>> many complaints from FS about tool versions.
>>>
>>> If you want to build from svn you can run bootstrap on another box  
>>> (a
>>> linux box perhaps) and then tar up that dir and move it to your mac.
>> Huh, interesting.
>>>
>>> We pre-bootstrap the release tarballs which is why that is building
>>> fine for you.
>> Right, Thanks for all your efforts and an outstanding platform!
>>
>> Marty
>>
>>
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Re: [Freeswitch-users] Starting FS on OSX (10.4.11 PPC)

[Martin Joseph]

On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote:

> I haven't tried using launchctl for FreeSWITCH. But when I saw your
> post, I tried it out. I have no problem getting it to work:
>
> I make a file "org.freeswitch.freeswitch.plist" and save it to ~/
> Library/LaunchAgents with the following content:
>
> 
>  "http://www.apple.com/DTDs/PropertyList-1.0.dtd
> ">
> 
> 
>   KeepAlive
>   
>   Label
>   org.freeswitch.freeswitch
>   Program
>   /usr/local/freeswitch/bin/freeswitch
>   RunAtLoad
>   
>   ServiceIPC
>   
> 
> 
>
> Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/
> org.freeswitch.freeswitch.plist"
>
> If you do the same command, but unload instead of load, it should stop
> freeswitch.
>
> Does this work for you?

Huh,

I have been trying to do something similar, but putting the file in / 
System/Library/LaunchDaemons.

Strangely, I tried your file right now (in LaunchDaemons) and  
freeswitch is started, but it doesn't respond to my devices trying to  
register?

Weird.

If I kill it and start it manually, it immediately responds and my  
devices register.

Ideas?
Thanks,

Marty
>
>
> Ivan
>
> Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph:
>
>> Hello again FreeSwitchers,
>>
>> I have built the 1.02 on 10.4.11(OSX) and had no problems with that.
>>
>> I have never been able to build from the SVN, but that is another
>> story.
>>
>> Now that I have migrated to 1.02 I was wondering if I can get some
>> help on a long standing issue I have with starting FS at boot.
>>
>> I am hoping to use Launchd which is the standard on OSX 10.4 and I
>> attempted to cobble together a script, but haven't had great results.
>>
>> I did search for wiki entries on this, but haven't found any help  
>> with
>> it.
>>
>> Ideas?
>> Thanks,
>> Marty
>>
>>
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Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality

[Anthony Minessale]

Yes it's hard to trust virtualized stuff because you have no idea what they
skimp on in terms of realtime access.

I won't endorse using FS on a VM as i have not done it very extensively
beyond openvz but I can
point out a few reasons why it has a fighting chance.

FS uses a timer architecture designed to amplify the work of one timer
thread into every timer open
by FS.  This single thread uses the monotonic clock on the system to try and
perfrom a 1ms accurate loop.

This single loop updates a soft value for current epoch time and microsecond
epoch time with the goal
of (again) being as close as possible to being accurate to 1 ms.

The timer loop also has a global matrix to all of the timing intervals being
subscribed to by a timer open by FS.
The loop will tick a counter in each unique timing interval (10ms, 20ms,
60ms etc) and fire a conditional broadcast to all of the timers who are
blocking for a tick.  This is not perfectly accurate but close enough to end
up plus or minus 2ms in resulting rtp traces.

So as long as the VM will expose the syscall down to the real monotonic
clock rather than doing it's own soft timing technique you have a better
chance for success.

The other issue with VM is with vmware, the bridged networking mode seems to
send 2 of every RTP
packet to the channel resulting in garbled audio from the obvious timing
issue introduced from too many packets.

Anyway evaluating FS with a VM is a good way to get acquainted but, with all
the money saved choosing FS, many can afford to buy it a nice 8 core box for
it to live on and still have money left over to support the project or
ClueCon 2009 this august ;)


On Fri, Jan 16, 2009 at 2:17 AM, Kristian Kielhofner <
kristian.kielhof...@gmail.com> wrote:

> Speaking of networking...
>
> After timing that's the next "achilles heel" of RTP handing with
> virtualization.
>
> Very, very few of these platforms were designed to handle massive
> numbers of very small RTP packets.  Everything from interrupt handling
> on the actual ethernet adapter to getting the data into userspace
> within the virtual instance is worrisome:
>
> http://www.xen.org/files/xensummit_4/NetworkIO_Santos.pdf
> http://forum.openvz.org/index.php?t=msg&goto=11619&;
>
> Interestingly enough the Xen paper makes it out to be really bad yet
> the OpenVZ post praises Xen's performance.  Without any real testing,
> who knows?  I just know that scaling 50pps per RTP stream (20ms
> packetization) can be difficult enough on native hardware, let alone
> [virtualization technology du jour].
>
> On Thu, Jan 15, 2009 at 5:02 PM, Remko Kloosterman
>  wrote:
> > Lot's of experience and suggestions here. Thanks.
> >
> > I believe it should be theoretically possible to have blip-free RTP
> > streaming through the appliance. Most windows ethernet drivers allow for
> > QoS packet scheduling. If the VMware network bridge driver honors this
> > and syncs the buffers at 20ms frames (or whatever frame size applies)
> > you should be able to schale up a bit and maintain low jitter.
> >
> > Anyone knows how the VMware network bridge exactly works?
> >
> >
> > -Oorspronkelijk bericht-
> > Van: freeswitch-users-boun...@lists.freeswitch.org
> > [mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Gregory
> > Boehnlein
> > Verzonden: donderdag 15 januari 2009 21:37
> > Aan: freeswitch-users@lists.freeswitch.org
> > Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality
> >
> >> To the contrary, we have had quite good results in virtualized
> >> environments and you don't really need timing that is that accurate to
> >
> >> make it work.
> >
> > If you don't handle RTP, I'm sure it is amazing. However, if you have to
> > do voicemail, stream audio from the server or do any kind of actual
> > time/latency/jitter sensitive processing, I don't care how much you tune
> > your hypervisor, it's never going to scale.
> >
> >> We work quite well on amazon EC2 for example.  There are 2 issues I
> >> know about with vmware, 1 is you need to set a setting on the host to
> >> extend somewhat sane clocks being available, the second is I have seen
> >
> >> issues with the bridged network adapter actually doubling up all
> >> packets causing very strange issues, I suggest not using bridged
> >> networking if you experience this.
> >
> > I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on
> > Vmware Server or Workstation?
> >
> >
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> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Re: [Freeswitch-users] Sending SMS to SIPtoGSM gateway

[Imthiyaz Ahmed]

Hi

This is what I found in the manual

SMS and MMS
The GSM and CDMA modules can send and receive SMS text messages and
MMS multimedia messages to or from a mobile phone. These are
transmitted over the Ethernet port as SIP MESSAGE messages to ensure
compatibility with a variety of different SIP PBXs. You determine for
each module the destination of received SMS or MMS messages and you
can program that SMS and MMS messages are always sent to a person who
sent an SMS or MMS to that phone number within a specified time.

The GS8 modular gateway can send and receive text messages in 7 bit
mode (160 characters per message) or in Unicode (70 characters per
message). It can send and receive multimedia messages comprising
images and rich text that are limited to 30 KB. The sending or
receiving client is responsible for formatting or interpreting the
message to OMA specifications.

Thanks

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Re: [Freeswitch-users] Outbound call - choose profile

[Anthony Minessale]

all gateways are properties of a profile
so even in the parse domain way of handling gateways whatever profile parsed
the domain
will be the owner of all the gateways discovered from the domain.

FYI,

The internal profile has nothing to do with users or anything else.
You are confusing the default examples for absolute fact.

On Fri, Jan 16, 2009 at 8:53 AM, Apostolos Pantsiopoulos wrote:

> Raymond Chandler wrote:
> > Apostolos Pantsiopoulos wrote:
> >> When I am using the following method to place a call from the dialplan :
> >>
> >> sofia/gateway//
> >>
> >> how do I tell FS which profile to use (as in the
> >> sofia// method?)
> >>
> >> I am asking that because all my calls to my declared  use
> >> the 5080 port,
> >> and I want them to use the 5060 port. Is there a way to configure a
> >>  to use
> >> a specific profile when making outbound calls?
> >>
> >>
> > the gateway should always use the profile from which it was included.
> > so, for instance, if you include gw1 from internal.xml, then gw1
> > should always use the internal profile
> >
> > -Ray
> > ___
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> >
> I am using the directory to define my gateways. Specifically, because
> this certain gateway
> is related to a specific user I am including this gateway in the user's
> xml file (in the dieractory).
> Then I use :
>
> 
>
> in my internal profile to let FS "parse" the gateways that I have
> declared in the directory.
>
> So one would expect that the gateways declared in the user's file would
> belong to the internal
> profile. Yet, when I am using the sofia/gateway//
> notation to send a call
> through this gateway the SIP packets get send from the 5080 port (which
> is my external profile's port)
>
> --
> ---
> Apostolos Pantsiopoulos
> Kinetix Tele.com R & D
> email: r...@kinetix.gr
> ---
>
>
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Re: [Freeswitch-users] spidermonkey problems

[Michael Jerris]

All long running non js code should be wrapped in the suspend/resume  
gc stuff.  For example:


cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE);
cb_state.saveDepth = JS_SuspendRequest(cx);
args.input_callback = dtmf_func;
args.buf = bp;
args.buflen = len;
switch_ivr_sleep(jss->session, ms, sync, &args);
JS_ResumeRequest(cx, cb_state.saveDepth);

I think this is your issue.  Can you please file a bug on jira for  
this issue (even better with a patch)


Mike



On Jan 16, 2009, at 5:54 AM, Jonas Gauffin wrote:

I've found the problem. one js thread wait in socket.read  
(mod_spidermonkey_socket) on data.
That caller have hangup, which means that the garbage collector  
waits on it to close.


All new javascript sessions waits in JS_AWAIT_GC_DONE for the  
garbage collector to be done before proceeding (which means that all  
new javascript calls don't do anything after being launched).


My server will not send anything until an agent gets free or the  
session hangs up (detects it through the event socket). And the  
event socket will not send that the session has been hangup until  
the socket have received anything (and the script can exit). So it's  
kind of deadlock between my server and the spidermonkey_socket.


Is it possible to add an option to socket.read to make it abort if  
the session have been closed? I know that I wrote  
mod_spidermonkey_socket from the start, but I can't figure out how  
to do it.


Will new sessions always wait on old ones to be garbage collected  
properly? For instance, what happens if a script have a lenghty post  
process after caller have hang up?


On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin > wrote:
I've got a loop, but the first thing checked in each iteration is if  
session.ready() returns false (and in that case exit the loop).


I do create sessions in the script: create, try to originate to a  
destination and then finally bridge together the caller and the new  
session.


I'll try to give you more details during the day.

On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale > wrote:
do you have any loops in your code that might not check for  
session.ready() in a exit when its not true.


The symptoms you posted would be consistent with held readlocks so  
if you got a gcore (or windows equiv) of the process you might be  
able to see what threads where doing what to hang on to the read lock.


also are you creating sessions in the script then executing app with  
them, beware of this because the thread of the script is used to  
execute apps on a session created that way and not the session thread.





On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin > wrote:

Hello

I got problems with hanging spidermonkey sessions and need some  
advice on how to debug them.


I've made a javascript queue application that uses  
mod_spidermonkey_socket. It works fine for a while,
but after some calls I noticed that calls didnt get transferred to  
agents. The reason was that earlier

calls had not been terminated properly.

freeswi...@test1> hupall
2009-01-15 12:15:04 [CRIT] switch_core_session.c:147  
switch_core_session_hupall() Giving up with 8 sessions remaining

API CALL [hupall()] output:
+OK hangup all channels with cause MANAGER_REQUEST


freeswi...@test1> show calls
API CALL [show(calls)] output:

0 total.


As you can see, 8 sessions are alive, but none of them are listed as  
calls. What kind of logs should I turn on to see what is happening  
with those sessions?


Thanks,
  Jonas

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Re: [Freeswitch-users] Starting FS on OSX

[Shido Xavier]

Please specify Intel or PPC.

-Greg M.


On Fri, Jan 16, 2009 at 1:29 AM, Martin Joseph  wrote:
>
> On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote:
>
>> Your build issue is with your autotools install, I have seen issues if
>> you have ever installed any of the autotools from macports or fink.
> I have never used Fink or Macports so that isn't it.  In fact the
> supposed statements made to the effect that FS will build from SVN
> fine on 10.4 with the latest available apple dev tools is quite wrong
> in my experience. I setup a virgin 10.4 and updated everything and had
> many complaints from FS about tool versions.
>>
>> If you want to build from svn you can run bootstrap on another box (a
>> linux box perhaps) and then tar up that dir and move it to your mac.
> Huh, interesting.
>>
>> We pre-bootstrap the release tarballs which is why that is building
>> fine for you.
> Right, Thanks for all your efforts and an outstanding platform!
>
> Marty
>
>
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Re: [Freeswitch-users] Starting FS on OSX

[Ivan C Myrvold]

I haven't tried using launchctl for FreeSWITCH. But when I saw your  
post, I tried it out. I have no problem getting it to work:

I make a file "org.freeswitch.freeswitch.plist" and save it to ~/ 
Library/LaunchAgents with the following content:


http://www.apple.com/DTDs/PropertyList-1.0.dtd 
">


KeepAlive

Label
org.freeswitch.freeswitch
Program
/usr/local/freeswitch/bin/freeswitch
RunAtLoad

ServiceIPC




Then in Terminal.app, I do a "launchctl load ~/Library/LaunchAgents/ 
org.freeswitch.freeswitch.plist"

If you do the same command, but unload instead of load, it should stop  
freeswitch.

Does this work for you?

Ivan

Den 15. jan.. 2009 kl. 22:54 skrev Martin Joseph:

> Hello again FreeSwitchers,
>
> I have built the 1.02 on 10.4.11(OSX) and had no problems with that.
>
> I have never been able to build from the SVN, but that is another  
> story.
>
> Now that I have migrated to 1.02 I was wondering if I can get some
> help on a long standing issue I have with starting FS at boot.
>
> I am hoping to use Launchd which is the standard on OSX 10.4 and I
> attempted to cobble together a script, but haven't had great results.
>
> I did search for wiki entries on this, but haven't found any help with
> it.
>
> Ideas?
> Thanks,
> Marty
>
>
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Re: [Freeswitch-users] Channel variable 'call_timeout'.

[Michael S Collins]

Tamas,

The channel variable won't work for you if you can't ignore early  
media. Your best bet is to use the variable execute_on_answer to  
transfer an answered call to a new extension. Then you could just  
sleep for 15sec and then check the value of endpoint_disposition.

What is the application that you are working on?


Sent from my iPhone

On Jan 16, 2009, at 5:17 AM, Tamas Cseke   
wrote:

> Hello,
>
> It seems originate_timeout isn't take effect when we got early media.
>
> Our carrier is sending the ring tone in early media, so if I try
>  data="{originate_timeout=15}sofia/gateway/mygw/whatsoever"/>
> timeout isn't occur after 15s
>
> however with:
> 
>
> timeout is OK. but we don't get early media, I think it would be  
> nice to
> not ignore it
>
> I found ring_ready and return_ring_ready could you please tell me what
> do they mean?
> So I'd like to make the timer expire only on 200 OK. and don't ignore
> early media.
>
> Thank you,
> Tomi
>
>
> Anthony Minessale írta:
>> call_timeout is only used if you are bridging 2 channels where one  
>> or both
>> of them is still unanswered.
>>
>> what you want to use is originate_timeout and forget about  
>> call_timeout
>>
>> you also have
>> leg_timeout and leg_progress_timeout both to be set in the {}
>> that do the timeout from the perspective of the new channel leg  
>> instead of
>> the caller leg.
>>
>>
>>
>> On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke > >wrote:
>>
>>
>>> Hello,
>>>
>>> I have the same problem,
>>>
>>> I don't understand the difference between
>>>
>>> progress_timeout
>>> originate_timeout
>>> call_timeout.
>>>
>>> I log timelimit_sec in switch_ivr_originate function and it seems,
>>> if I set call_timeout then, timelimit_sec will be this value
>>> if I set originate_timeout then timelimit_sec will be this value.  
>>> maybe
>>> this is for backward compat?
>>>
>>> originate_timeout as in the wiki:
>>> "Determines how long FreeSwitch is going to wait for a response from
>>> the invite message sent to the gateway. "
>>>
>>> I quess this would be an 100 reply.
>>>
>>> But how could I set a timeout for 200? I mean timeout for an answer.
>>> which variable would control this?
>>> I quess it was call_timeout previosly.
>>> Please explain me this timeout variables
>>>
>>> Thanks,
>>> Tamas
>>>
>>> Michael Collins írta:
>>>
 FYI, it is on the channel variables page but it's in a crazy  
 place under
 "unknown functionality" which is silly.
 http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout

 Anyway, I've got wiki cleaning on my to-do list and I'll start in  
 earnest
 next month when I have some time...

 -MC

 On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins >>> >

>>> wrote:
>>>

>> I used "call-timeout" because that's how I understood it from  
>> the Wiki.
>> (?)
>>
>>
>>
> Yep, that's all that there is on the wiki. Unfortunately the  
> channel
> variables page is one of many in need of some attention. I will  
> add
> "originate_timeout" right away. The only question remaining is  
> what, if
> anything, does call_timeout do? That channel variable is in the  
> source
>
>>> code
>>>
> but I don't know exactly what it does.
>
> -MC
>
>
>
>
>> --
>> View this message in context:
>>
>>
>>> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.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] Outbound call - choose profile

[Apostolos Pantsiopoulos]

Raymond Chandler wrote:
> Apostolos Pantsiopoulos wrote:
>> When I am using the following method to place a call from the dialplan :
>>
>> sofia/gateway//
>>
>> how do I tell FS which profile to use (as in the 
>> sofia// method?)
>>
>> I am asking that because all my calls to my declared  use 
>> the 5080 port,
>> and I want them to use the 5060 port. Is there a way to configure a 
>>  to use
>> a specific profile when making outbound calls?
>>
>>   
> the gateway should always use the profile from which it was included.
> so, for instance, if you include gw1 from internal.xml, then gw1 
> should always use the internal profile
>
> -Ray
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>   
I am using the directory to define my gateways. Specifically, because 
this certain gateway
is related to a specific user I am including this gateway in the user's 
xml file (in the dieractory).
Then I use :



in my internal profile to let FS "parse" the gateways that I have 
declared in the directory.

So one would expect that the gateways declared in the user's file would 
belong to the internal
profile. Yet, when I am using the sofia/gateway// 
notation to send a call
through this gateway the SIP packets get send from the 5080 port (which 
is my external profile's port)

-- 
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr
--- 


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Re: [Freeswitch-users] FS doesn't maitain PRI- D-channel state right

[Anthony Minessale]

Please do not report bugs on the mailing list.
It's very hard to keep track of them this way.
Please file all bugs to jira so we will not lose track of them.

http://jira.freeswitch.org


On Fri, Jan 16, 2009 at 1:25 AM, Helmut Kuper wrote:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
>
> I found, that FS doesn't maintain D-Channel's state correctly.
>
> I have a PRI with disabled layer 2 and 3 on TDM side. When FS starts up
> I get this on console:
>
>
> 2009-01-16 08:16:10 [DEBUG] ozmod_isdn.c:1441 zap_isdn_run() ISDN thread
> starting.
> 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Starting trunk 0xae411008
> (sapi: 0, tei: 0, mode: PTP TE)
> 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending SABME
> 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() Sending frame
> - - Q.921 Packet [Outgoing] ---
>SAPI: 0, TEI: 0, C/R: Command (0)
>
>Type: U Frame (SABME)
>  P/F: 1
>
>Q.921 state: "TEI Assigned" (4) [flags: ]
> - --
>
> 2009-01-16 08:16:10 [CONSOLE] switch_loadable_module.c:857
> switch_loadable_module_load_file() Successfully Loaded [mod_openzap]
> 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:141
> switch_loadable_module_process() Adding Endpoint 'openzap'
> 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:239
> switch_loadable_module_process() Adding Application 'disable_ec'
> 2009-01-16 08:16:10 [NOTICE] switch_loadable_module.c:259
> switch_loadable_module_process() Adding API Function 'oz'
> 2009-01-16 08:16:10 [DEBUG] Span:0 Q.921() New packet received (3 bytes)
> - - Q.921 Packet [Incoming] ---
>SAPI: 0, TEI: 0, C/R: Response (0)
>
>Type: U Frame (DM (Disconnected Mode))
>  P/F: 1
>
>Q.921 state: "TEI Assigned" (4) [flags: ]
> - --
>
> So TDM side tells FS that the link is down:
>
> but when I check d-channel's state FS tells me that it is UP:
>
> freeswi...@ippbx-prod-node0> oz dump 1 16
> API CALL [oz(dump 1 16)] output:
> span_id: 1
> chan_id: 16
> physical_span_id: 1
> physical_chan_id: 16
> type: DQ921
> state: UP
> last_state: DOWN
> cid_date:
> cid_name:
> cid_num:
> ani:
> aniII:
> dnis:
> rdnis:
> cause: NONE
>
> When I do an outbound call FS throws no error or warning that the link
> isn't up and gives the call a try.
>
>
> regards
> helmut
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.9 (MingW32)
>
> iEYEARECAAYFAklwNnsACgkQ4tZeNddg3dzjKQCbBDU/SSOyKbD2JGcJFOJZDyBQ
> nI0An3CFp9HIuTB0cQWT0iJ1Rlx1+yGk
> =ycwj
> -END PGP SIGNATURE-
>
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>



-- 
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Re: [Freeswitch-users] Outbound call - choose profile

[Raymond Chandler]

Apostolos Pantsiopoulos wrote:

When I am using the following method to place a call from the dialplan :

sofia/gateway//

how do I tell FS which profile to use (as in the 
sofia// method?)


I am asking that because all my calls to my declared  use the 
5080 port,
and I want them to use the 5060 port. Is there a way to configure a 
 to use

a specific profile when making outbound calls?

  

the gateway should always use the profile from which it was included.
so, for instance, if you include gw1 from internal.xml, then gw1 should 
always use the internal profile


-Ray
begin:vcard
fn:intralanman
n:Chandler;Raymond
adr:;;630 Cooks Rd.;Farmville;VA;23901;United States
email;internet:intralan...@freeswitch.org
tel;cell:+1.434.315.4132
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Re: [Freeswitch-users] Channel variable 'call_timeout'.

[Tamas Cseke]

Hello,

It seems originate_timeout isn't take effect when we got early media.

Our carrier is sending the ring tone in early media, so if I try

timeout isn't occur after 15s

however with:


timeout is OK. but we don't get early media, I think it would be nice to 
not ignore it

I found ring_ready and return_ring_ready could you please tell me what 
do they mean?
So I'd like to make the timer expire only on 200 OK. and don't ignore 
early media.

Thank you,
Tomi


Anthony Minessale írta:
> call_timeout is only used if you are bridging 2 channels where one or both
> of them is still unanswered.
>
> what you want to use is originate_timeout and forget about call_timeout
>
> you also have
> leg_timeout and leg_progress_timeout both to be set in the {}
> that do the timeout from the perspective of the new channel leg instead of
> the caller leg.
>
>
>
> On Fri, Dec 5, 2008 at 10:34 AM, Tamas Cseke wrote:
>
>   
>> Hello,
>>
>> I have the same problem,
>>
>> I don't understand the difference between
>>
>> progress_timeout
>> originate_timeout
>> call_timeout.
>>
>> I log timelimit_sec in switch_ivr_originate function and it seems,
>> if I set call_timeout then, timelimit_sec will be this value
>> if I set originate_timeout then timelimit_sec will be this value. maybe
>> this is for backward compat?
>>
>> originate_timeout as in the wiki:
>>  "Determines how long FreeSwitch is going to wait for a response from
>> the invite message sent to the gateway. "
>>
>> I quess this would be an 100 reply.
>>
>> But how could I set a timeout for 200? I mean timeout for an answer.
>> which variable would control this?
>> I quess it was call_timeout previosly.
>> Please explain me this timeout variables
>>
>> Thanks,
>> Tamas
>>
>> Michael Collins írta:
>> 
>>> FYI, it is on the channel variables page but it's in a crazy place under
>>> "unknown functionality" which is silly.
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>>>
>>> Anyway, I've got wiki cleaning on my to-do list and I'll start in earnest
>>> next month when I have some time...
>>>
>>> -MC
>>>
>>> On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins 
>>>   
>> wrote:
>> 
>>>   
> I used "call-timeout" because that's how I understood it from the Wiki.
>  (?)
>
>
>   
 Yep, that's all that there is on the wiki. Unfortunately the channel
 variables page is one of many in need of some attention. I will add
 "originate_timeout" right away. The only question remaining is what, if
 anything, does call_timeout do? That channel variable is in the source
 
>> code
>> 
 but I don't know exactly what it does.

 -MC



 
> --
> View this message in context:
>
>   
>> http://www.nabble.com/Channel-variable-%27call_timeout%27.-tp20677406p20689832.html
>> 
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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>
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>>>
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[Freeswitch-users] Outbound call - choose profile

[Apostolos Pantsiopoulos]

When I am using the following method to place a call from the dialplan :

sofia/gateway//

how do I tell FS which profile to use (as in the 
sofia// method?)

I am asking that because all my calls to my declared  use the 
5080 port,
and I want them to use the 5060 port. Is there a way to configure a 
 to use
a specific profile when making outbound calls?

-- 
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr
--- 


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Re: [Freeswitch-users] spidermonkey problems

[Jonas Gauffin]

I've found the problem. one js thread wait in socket.read
(mod_spidermonkey_socket) on data.
That caller have hangup, which means that the garbage collector waits on it
to close.

All new javascript sessions waits in JS_AWAIT_GC_DONE for the garbage
collector to be done before proceeding (which means that all new javascript
calls don't do anything after being launched).

My server will not send anything until an agent gets free or the session
hangs up (detects it through the event socket). And the event socket will
not send that the session has been hangup until the socket have received
anything (and the script can exit). So it's kind of deadlock between my
server and the spidermonkey_socket.

Is it possible to add an option to socket.read to make it abort if the
session have been closed? I know that I wrote mod_spidermonkey_socket from
the start, but I can't figure out how to do it.

Will new sessions always wait on old ones to be garbage collected properly?
For instance, what happens if a script have a lenghty post process after
caller have hang up?

On Fri, Jan 16, 2009 at 9:38 AM, Jonas Gauffin wrote:

> I've got a loop, but the first thing checked in each iteration is if
> session.ready() returns false (and in that case exit the loop).
> I do create sessions in the script: create, try to originate to a
> destination and then finally bridge together the caller and the new session.
>
> I'll try to give you more details during the day.
>
> On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> do you have any loops in your code that might not check for
>> session.ready() in a exit when its not true.
>>
>> The symptoms you posted would be consistent with held readlocks so if you
>> got a gcore (or windows equiv) of the process you might be able to see what
>> threads where doing what to hang on to the read lock.
>>
>> also are you creating sessions in the script then executing app with them,
>> beware of this because the thread of the script is used to execute apps on a
>> session created that way and not the session thread.
>>
>>
>>
>>
>> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin 
>> wrote:
>>
>>> Hello
>>> I got problems with hanging spidermonkey sessions and need some advice on
>>> how to debug them.
>>>
>>> I've made a javascript queue application that uses
>>> mod_spidermonkey_socket. It works fine for a while,
>>> but after some calls I noticed that calls didnt get transferred to
>>> agents. The reason was that earlier
>>> calls had not been terminated properly.
>>>
>>> freeswi...@test1> hupall
>>> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147
>>> switch_core_session_hupall() Giving up with 8 sessions remaining
>>> API CALL [hupall()] output:
>>> +OK hangup all channels with cause MANAGER_REQUEST
>>>
>>>
>>> freeswi...@test1> show calls
>>> API CALL [show(calls)] output:
>>>
>>> 0 total.
>>>
>>>
>>> As you can see, 8 sessions are alive, but none of them are listed as
>>> calls. What kind of logs should I turn on to see what is happening with
>>> those sessions?
>>>
>>> Thanks,
>>>   Jonas
>>>
>>> ___
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>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>
>> AIM: anthm
>> MSN:anthony_miness...@hotmail.com 
>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
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Re: [Freeswitch-users] mod_opal first unsuccessful test

[Apostolos Pantsiopoulos]

Any news regarding this issue?

Apostolos Pantsiopoulos wrote:

I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and
FS is on xxx.xxx.xxx.212

Robert Jongbloed wrote:


Can you send me a WireShark capture?

 


Robert Jongbloed

OPAL/OpenH323/PTLib Architect and Co-founder.

 


*From:* Apostolos Pantsiopoulos [mailto:r...@kinetix.gr]
*Sent:* Tuesday, 13 January 2009 11:05 PM
*To:* freeswitch-users@lists.freeswitch.org
*Cc:* Robert Jongbloed
*Subject:* Re: [Freeswitch-users] mod_opal first unsuccessful test

 


I also tried using Ekiga - which is OPAL based - and got the same
behavior. No audio - although I can see RTP packets.

Apostolos Pantsiopoulos wrote:

Hi,

Yes, openPhone is working with my soundcard. I am using it
every day for testing purposes. I use the 1.8.1 version. Is there a newer
version that uses OPAL? I didn't know that. Where can I get it from?

Robert Jongbloed wrote:

Hi guys,

 

I was using the OpenPhone that you build with OPAL for my testing. So 
that is identical (I think) to you.


 


I have not (yet) do any third party client testing.

 


Two ALERTING messages are fine, perfectly legal and OPAL can handle it.

 

You say you can see the RTP packets flowing so that implies that the 
mod_opal is actually working, so let's look somewhere else. Have you 
confirmed that OpenPhone is using the sound card correctly? Made a 
call between two machines JUST using OpenPhone for example?


 

 


Robert Jongbloed

OPAL/OpenH323/PTLib Architect and Co-founder.

 


*From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
*Sent:* Tuesday, 13 January 2009 6:20 AM
*To:* Robert Jongbloed
*Subject:* Fwd: [Freeswitch-users] mod_opal first unsuccessful test

 


Heh,
what client are you using in your tests that are working?


-- Forwarded message --
From: *Apostolos Pantsiopoulos* >

Date: Mon, Jan 12, 2009 at 9:51 AM
Subject: [Freeswitch-users] mod_opal first unsuccessful test
To: freeswitch-users@lists.freeswitch.org 




Hi,

   I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

   When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when 
I am

doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

   In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

   Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr 
---


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--
---
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email: r...@kinetix.gr 
--- 
 




  
 
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--
---
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: r...@kinetix.gr 
--- 



--
---
Apostolos

Re: [Freeswitch-users] spidermonkey problems

[Jonas Gauffin]

I've got a loop, but the first thing checked in each iteration is if
session.ready() returns false (and in that case exit the loop).
I do create sessions in the script: create, try to originate to a
destination and then finally bridge together the caller and the new session.

I'll try to give you more details during the day.

On Fri, Jan 16, 2009 at 12:48 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> do you have any loops in your code that might not check for session.ready()
> in a exit when its not true.
>
> The symptoms you posted would be consistent with held readlocks so if you
> got a gcore (or windows equiv) of the process you might be able to see what
> threads where doing what to hang on to the read lock.
>
> also are you creating sessions in the script then executing app with them,
> beware of this because the thread of the script is used to execute apps on a
> session created that way and not the session thread.
>
>
>
>
> On Thu, Jan 15, 2009 at 5:20 AM, Jonas Gauffin wrote:
>
>> Hello
>> I got problems with hanging spidermonkey sessions and need some advice on
>> how to debug them.
>>
>> I've made a javascript queue application that uses
>> mod_spidermonkey_socket. It works fine for a while,
>> but after some calls I noticed that calls didnt get transferred to agents.
>> The reason was that earlier
>> calls had not been terminated properly.
>>
>> freeswi...@test1> hupall
>> 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147
>> switch_core_session_hupall() Giving up with 8 sessions remaining
>> API CALL [hupall()] output:
>> +OK hangup all channels with cause MANAGER_REQUEST
>>
>>
>> freeswi...@test1> show calls
>> API CALL [show(calls)] output:
>>
>> 0 total.
>>
>>
>> As you can see, 8 sessions are alive, but none of them are listed as
>> calls. What kind of logs should I turn on to see what is happening with
>> those sessions?
>>
>> Thanks,
>>   Jonas
>>
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>
>
> --
> Anthony Minessale II
>
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Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality

[Kristian Kielhofner]

Speaking of networking...

After timing that's the next "achilles heel" of RTP handing with virtualization.

Very, very few of these platforms were designed to handle massive
numbers of very small RTP packets.  Everything from interrupt handling
on the actual ethernet adapter to getting the data into userspace
within the virtual instance is worrisome:

http://www.xen.org/files/xensummit_4/NetworkIO_Santos.pdf
http://forum.openvz.org/index.php?t=msg&goto=11619&;

Interestingly enough the Xen paper makes it out to be really bad yet
the OpenVZ post praises Xen's performance.  Without any real testing,
who knows?  I just know that scaling 50pps per RTP stream (20ms
packetization) can be difficult enough on native hardware, let alone
[virtualization technology du jour].

On Thu, Jan 15, 2009 at 5:02 PM, Remko Kloosterman
 wrote:
> Lot's of experience and suggestions here. Thanks.
>
> I believe it should be theoretically possible to have blip-free RTP
> streaming through the appliance. Most windows ethernet drivers allow for
> QoS packet scheduling. If the VMware network bridge driver honors this
> and syncs the buffers at 20ms frames (or whatever frame size applies)
> you should be able to schale up a bit and maintain low jitter.
>
> Anyone knows how the VMware network bridge exactly works?
>
>
> -Oorspronkelijk bericht-
> Van: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Gregory
> Boehnlein
> Verzonden: donderdag 15 januari 2009 21:37
> Aan: freeswitch-users@lists.freeswitch.org
> Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality
>
>> To the contrary, we have had quite good results in virtualized
>> environments and you don't really need timing that is that accurate to
>
>> make it work.
>
> If you don't handle RTP, I'm sure it is amazing. However, if you have to
> do voicemail, stream audio from the server or do any kind of actual
> time/latency/jitter sensitive processing, I don't care how much you tune
> your hypervisor, it's never going to scale.
>
>> We work quite well on amazon EC2 for example.  There are 2 issues I
>> know about with vmware, 1 is you need to set a setting on the host to
>> extend somewhat sane clocks being available, the second is I have seen
>
>> issues with the bridged network adapter actually doubling up all
>> packets causing very strange issues, I suggest not using bridged
>> networking if you experience this.
>
> I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on
> Vmware Server or Workstation?
>
>
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-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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