Re: [Freeswitch-users] FS 1.0.4 erl configure error

[Michael Jerris]

Are those in the Tarball?

On Sep 15, 2009, at 11:47 PM, Nandy Dagondon   
wrote:


it's working now. the problem? it's the configure script itself.  
some ^M characters somehow crept into the line containing  
ac_config_files.  tks for the tip Andrew!


/nandy

On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon  
 wrote:

is the Erlang source needed in the FS source directory?

/nandy



On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon  
 wrote:
the ./configure script aborts after the last error message. any hint  
where to look for the problem? tks.


/nandy



On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson  
 wrote:

On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
> hi folks, anyone encountered this problem? tks.

I don't think this has anything to do with erlang or the freeswitch
erlang module, it's simply that that module's config checks are run
shortly before the real failure occurs.

Andrew

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[Freeswitch-users] How to Process Invalid extension in FS

[Ahmed Munir]

Hi,

I'm newbie in FS. I want to know how to process invalid extension in FS?
Because I want to prompt the IVR if invalid extension is dialled.


Kindly advice me.

-- 
Regards,

Ahmed Munir
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Re: [Freeswitch-users] How to Process Invalid extension in FS

[Jason White]

Ahmed Munir  wrote:
> I'm newbie in FS. I want to know how to process invalid extension in FS?
> Because I want to prompt the IVR if invalid extension is dialled.

You could write an entry at the end of the dial-plan that matches any
extension and invokes the IVR.


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Re: [Freeswitch-users] FS 1.0.4 erl configure error

[Nandy Dagondon]

hi mike, i download the tarball file to check the configure script. it's
clean. so, there must be an error during my first download or build. - nandy


On Wed, Sep 16, 2009 at 3:54 PM, Michael Jerris  wrote:

> Are those in the Tarball?
>
>
> On Sep 15, 2009, at 11:47 PM, Nandy Dagondon  wrote:
>
> it's working now. the problem? it's the configure script itself. some ^M
> characters somehow crept into the line containing ac_config_files.  tks for
> the tip Andrew!
>
> /nandy
>
> On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon < 
> nandy1...@gmail.com> wrote:
>
>> is the Erlang source needed in the FS source directory?
>>
>> /nandy
>>
>>
>>
>> On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon < 
>> nandy1...@gmail.com> wrote:
>>
>>> the ./configure script aborts after the last error message. any hint
>>> where to look for the problem? tks.
>>>
>>> /nandy
>>>
>>>
>>>
>>> On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson < 
>>> and...@hijacked.us> wrote:
>>>
 On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
 > hi folks, anyone encountered this problem? tks.

 I don't think this has anything to do with erlang or the freeswitch
 erlang module, it's simply that that module's config checks are run
 shortly before the real failure occurs.

 Andrew

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

[Tristan Mahé]

Hi,

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

Regards,

Gled

Diego Viola a écrit :
> 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
> mailto:m...@freeswitch.org>> 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 prime candidate to help answer
> questions, refine the mod_xml_curl wiki documentation, write
> up a tutorial, contribute a working example of a web server &
> database schema, etc. If you are good with a scripting
> language then we could definitely use help with rounding out
> the docs for your favorite language. We could also use code
> samples, so ask for a contrib folder if you have things you
> would like to share. Or how about this: you read something on
> the wiki, it doesn't quite work when you try, so you tinker
> until you figure it out. Now you're in a position to update
> the wiki for everyone else's benefit, too.
>
> As you can see, you don't have to be a FreeSWITCH expert
> before you can help the project. What we really need are
> people who care about the project and want to see it flourish.
> If you are such a person then please contact me off list. Tell
> me what you're good at or where you would like to help.
>
> Many thanks for all of your support!
> -Michael
>
>
>
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[Freeswitch-users] Limit_Hash

[Matt Riddell]

Hi,

Didn't see this one come through before when I posted it, so sending it 
again - apologies if it did come through.

I've moved this discussion to users as it seems my query is moving in 
that direction :)

So, upon looking at limit_hash, it appears to do what I need to do.

My question then becomes, how do I set a hash for an originated call?

It seems that limit_hash is an application rather than a channel 
variable, and so far I've been doing most things without touching the 
dialplan.

So, say I want to originate 9 calls, 3 from 3 customers.

I would like to mark the calls with my_customer_group_1 through 3, and 
then use the limit_hash_usage command to verify the count of channels in 
each group.

I therefore have a few questions:

1. Can I mark a call in the originate statement?

2. How do I use the limit_hash_usage command?

The wiki states:

You can verify the usage of any resource with the limit_hash_usage api call.

limit_hash_usage  

Is realm the same as a SIP realm?

Is id the hash that I have used to mark the call with?

Just making sure :)

-- 
Cheers,

Matt Riddell
Director
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[Freeswitch-users] How can I configure TO and FROM in the invite message

[Tzury Bar Yochay]

Hi,
Currently, the invite message looks as follows

INVITE sip:1...@client_ip:5060 SIP/2.0
Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
Max-Forwards: 69
From: "Extension 1001" ;tag=2rH67Q3aa1rpe
To: 

Is there a way to configure FS so the message will look like this:

INVITE sip:1...@client_ip:5060 SIP/2.0
Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
Max-Forwards: 69
From: "Extension 1001" ;tag=2rH67Q3aa1rpe
To: 

That is, at "From" having the account's domain name (e.g.
sip:1...@example.com) instead of the server's IP address.
and having the same at "To"

thanks
@tzury

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[Freeswitch-users] shared mailbox, mwi

[Frank Carmickle]

Hello

I am trying to set up a shared mailbox per group of extensions where the 
mailboxes are [2-9][2-9][0,2,4,6,8]0 and the phones are [2-9][2-9]\d[1-9].  I 
mostly have it working except I can't seem to figure out how to get mwi for the 
phones in the group.  I've tried a number of things.  I haven't figured out how 
you can have a mailbox that doesn't have a user associated with it.  When I use 
 I get the mwi but I am not able 
to call that user any more.  I have  set 
for each user.

What are my options for making this scenario work?

Thank you
--Frank

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Re: [Freeswitch-users] Limit_Hash

[Rupa Schomaker]

On Wed, Sep 16, 2009 at 5:40 AM, Matt Riddell  wrote:
> My question then becomes, how do I set a hash for an originated call?
>
> It seems that limit_hash is an application rather than a channel
> variable, and so far I've been doing most things without touching the
> dialplan.

Yes, it is a dialplan app.  So, you have to call it from a dialplan.

> So, say I want to originate 9 calls, 3 from 3 customers.
>
> I would like to mark the calls with my_customer_group_1 through 3, and
> then use the limit_hash_usage command to verify the count of channels in
> each group.
>
> I therefore have a few questions:
>
> 1. Can I mark a call in the originate statement?

If you don't want to bounce through the XML Dialplan, you can use an
inline dialplan instead and specify the dialplan on the originate
commandline.  The wiki has an example:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_InlineDialplan#Originate

> 2. How do I use the limit_hash_usage command?
>
> The wiki states:
>
> You can verify the usage of any resource with the limit_hash_usage api call.
>
> limit_hash_usage  
>
> Is realm the same as a SIP realm?
>
> Is id the hash that I have used to mark the call with?
>
> Just making sure :)

realm and id are just arbitrary strings.  They can be useful if you
want to do reporting out of the database (if using regular limit api,
for limit_usage it wouldn't apply).

You can provide your own meaning to realm and id.

In a multi-tenant setup, you might use domain as the realm but that
isn't necessary.

> --
> Cheers,
>
> Matt Riddell
> Director


-- 
-Rupa

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[Freeswitch-users] how to add new user for external profile

[pankaj anand]

hi ,  i m very new to the FreeSwitch..
can any one tell me how to add a new user.
i have already tried creating a new user by creating a
$INSTALL_DIR/conf/directory/default/pankaj.xml :


  

  
  


  
  
  
  
  
  
  
  

  


but when i try to connect it using , the softphone shows  forbidden.
Can anyone tell me where i am making a mistake.

with regards
Pankaj anand
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Re: [Freeswitch-users] reloadxml question

[Brian West]

Yes you're missing a switch_xml_free(xml); some place.

/b

On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote:

> hi,
>
> I've build a custom module for FS and everytihng work well except  
> reloadxml command :P... m'I missing something in my module? ... i  
> used mod_skeleton as a template when i started.
>
>
> When i start the FS without my module reloadxml works fine ... as  
> soon as i include my module within modules.conf.xml and start FS ..  
> it hangs.
> So, it is definitelly up to the custom module ... but what can it be?
>
>
>
> freeswi...@l01freeswitch1>
> freeswi...@l01freeswitch1>
> freeswi...@l01freeswitch1> reloadxml
>
> nothing happens ... i have to kill freeswitch (kill -9) to get the  
> shell.
>
> T.
>


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Re: [Freeswitch-users] how to add new user for external profile

[Tihomir Culjaga]

FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct.


freeswitch.xml:
  




Than, you need to check sip profiles. By default FS will accept
registrations on internal profiles only... so you should enable it on the
external as well.


look at this portion of your adequate sip profile:

   













































Just make sure you use correct IP_ADDRESS:PORT to match the correct profile

vars.xml:

  
  
  
  


  
  
  
  


T.


On Wed, Sep 16, 2009 at 11:29 AM, pankaj anand wrote:

> hi ,  i m very new to the FreeSwitch..
> can any one tell me how to add a new user.
> i have already tried creating a new user by creating a
> $INSTALL_DIR/conf/directory/default/pankaj.xml :
>
> 
>   
> 
>   
>   
> 
> 
>   
>   
>   
>   
>   
>value="$${outbound_caller_name}"/>
>value="$${outbound_caller_id}"/>
>   
> 
>   
> 
>
> but when i try to connect it using , the softphone shows  forbidden.
> Can anyone tell me where i am making a mistake.
>
> with regards
> Pankaj anand
>
>
>
>
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Re: [Freeswitch-users] reloadxml question

[Tihomir Culjaga]

perfect,

thanks.

T.

On Wed, Sep 16, 2009 at 4:05 PM, Brian West  wrote:

> Yes you're missing a switch_xml_free(xml); some place.
>
> /b
>
> On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote:
>
> > hi,
> >
> > I've build a custom module for FS and everytihng work well except
> > reloadxml command :P... m'I missing something in my module? ... i
> > used mod_skeleton as a template when i started.
> >
> >
> > When i start the FS without my module reloadxml works fine ... as
> > soon as i include my module within modules.conf.xml and start FS ..
> > it hangs.
> > So, it is definitelly up to the custom module ... but what can it be?
> >
> >
> >
> > freeswi...@l01freeswitch1>
> > freeswi...@l01freeswitch1>
> > freeswi...@l01freeswitch1> reloadxml
> >
> > nothing happens ... i have to kill freeswitch (kill -9) to get the
> > shell.
> >
> > T.
> >
>
>
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[Freeswitch-users] reloadxml question

[Tihomir Culjaga]

hi,

I've build a custom module for FS and everytihng work well except reloadxml
command :P... m'I missing something in my module? ... i used mod_skeleton as
a template when i started.


When i start the FS without my module reloadxml works fine ... as soon as i
include my module within modules.conf.xml and start FS .. it hangs.
So, it is definitelly up to the custom module ... but what can it be?



freeswi...@l01freeswitch1>
freeswi...@l01freeswitch1>
freeswi...@l01freeswitch1> reloadxml

nothing happens ... i have to kill freeswitch (kill -9) to get the shell.

T.
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[Freeswitch-users] originate command sofia behaviour

[Alberto Escudero]

I will like to update the wiki to spell out clearly the differences
between this three commands
I have a IVR running in 4600 and the FS box has IP address 192.168.46.15

originate sofia/192.168.46.15/1001  4600
originate sofia/internal/1...@192.168.46.15 4600
originate sofia/internal/1001%192.168.46.15 4600

The first originate places a call as a external gateway, not until
registered phone 1001 answers the call is transfer to 4600

The second and third originate command triggers extension 4600 Javascript
IVR although 1001 has not answer

Can anyone clarify me if this is the intended behavior also including the
difference between % and @

/aep




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[Freeswitch-users] memory leak - outbound socket

[Christian Löschenkohl]

hello

version : 1.0.4 std. tarball

- the wiki example for php outbound socket connection leaks memory without the 
async option
- the memory used is never given back
- async isn't that usefull for us - we want to query databases, set variables 
and so on
   no wait statements are possible




  no async 



the script is on the site
http://wiki.freeswitch.org/wiki/PHP_ESL

---

what can i do?
on our production server we use outbound socket connection and the 4 gig of 
memory are
eaten up in less than a day

br

-- 
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenk...@xpirio.com

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Re: [Freeswitch-users] How can I configure TO and FROM in the invite message

[João Mesquita]

Is this what you are looking for?

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

jmesquita

On Wed, Sep 16, 2009 at 10:58 AM, Tzury Bar Yochay  wrote:

> Hi,
> Currently, the invite message looks as follows
>
> INVITE sip:1...@client_ip:5060 SIP/2.0
> Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
> Max-Forwards: 69
> From: "Extension 1001" ;tag=2rH67Q3aa1rpe
> To: 
>
> Is there a way to configure FS so the message will look like this:
>
> INVITE sip:1...@client_ip:5060 SIP/2.0
> Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
> Max-Forwards: 69
> From: "Extension 1001" ;tag=2rH67Q3aa1rpe
> To: 
>
> That is, at "From" having the account's domain name (e.g.
> sip:1...@example.com ) instead of the server's IP
> address.
> and having the same at "To"
>
> thanks
> @tzury
>
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Re: [Freeswitch-users] reloadxml question

[Brian West]

Might I ask what you are working on?Its interesting to hear what  
people are doing with FreeSWITCH.

/b

On Sep 16, 2009, at 9:17 AM, Tihomir Culjaga wrote:

> perfect,
>
> thanks.
>
> T.


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

[Michael Collins]

On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé  wrote:

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

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

Thanks again!
-MC
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Re: [Freeswitch-users] memory leak - outbound socket

[Rupa Schomaker]

Either:

1) Provide a simple self-contained example that demonstrates the leak

or

2) Run your application with FreeSWITCH under valgrind and provide the
final output.  To run freeswitch under valgrind:

http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29

You should not have to run with high load to capture the behavior.
Try with just 5 (in series) and then stop freeswitch.


2009/9/16 Christian Löschenkohl :
> hello
>
> version : 1.0.4 std. tarball
>
> - the wiki example for php outbound socket connection leaks memory without 
> the async option
> - the memory used is never given back
> - async isn't that usefull for us - we want to query databases, set variables 
> and so on
>   no wait statements are possible
>
> 
> 
>  data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>   no async 
> 
> 
> 
>
> the script is on the site
> http://wiki.freeswitch.org/wiki/PHP_ESL
>
> ---
>
> what can i do?
> on our production server we use outbound socket connection and the 4 gig of 
> memory are
> eaten up in less than a day
>
> br
>
> --
> Ing. Christian Löschenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenk...@xpirio.com
>
> ___
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>



-- 
-Rupa

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Re: [Freeswitch-users] A FreeSWITCH Calling Card Application written in Ruby

[Michael Collins]

On Tue, Sep 15, 2009 at 1:46 PM, roberto  wrote:

> Hello,
>
> Someone could tell me what happens to the project, it seems that is no
> longer available in github ?
>
> http://github.com/diego/freeswitch-card/
>
> thanks,
>

I haven't seen Diego Viola on line for a day or two. He can answer this when
he's back online.
-MC
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Re: [Freeswitch-users] How can I configure TO and FROM in the invite message

[Brian West]

Or you setup a gateway and set the from-domain

/b

On Sep 16, 2009, at 10:00 AM, João Mesquita wrote:


Is this what you are looking for?

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

jmesquita

On Wed, Sep 16, 2009 at 10:58 AM, Tzury Bar Yochay > wrote:

Hi,
Currently, the invite message looks as follows

INVITE sip:1...@client_ip:5060 SIP/2.0
Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
Max-Forwards: 69
From: "Extension 1001" ;tag=2rH67Q3aa1rpe
To: 

Is there a way to configure FS so the message will look like this:

INVITE sip:1...@client_ip:5060 SIP/2.0
Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se
Max-Forwards: 69
From: "Extension 1001" ;tag=2rH67Q3aa1rpe
To: 

That is, at "From" having the account's domain name (e.g.
sip:1...@example.com) instead of the server's IP address.
and having the same at "To"

thanks
@tzury

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Re: [Freeswitch-users] memory leak - outbound socket

[Christian Löschenkohl]

as a good fs user - of course i am :-) - i made a jira on this
MODAPP-336 to be precise

i hope this helps to solve my problem

br

On 2009-09-16 17:05, Rupa Schomaker wrote:
> Either:
>
> 1) Provide a simple self-contained example that demonstrates the leak
>
> or
>
> 2) Run your application with FreeSWITCH under valgrind and provide the
> final output.  To run freeswitch under valgrind:
>
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29
>
> You should not have to run with high load to capture the behavior.
> Try with just 5 (in series) and then stop freeswitch.
>
>
> 2009/9/16 Christian Löschenkohl:
>> hello
>>
>> version : 1.0.4 std. tarball
>>
>> - the wiki example for php outbound socket connection leaks memory without 
>> the async option
>> - the memory used is never given back
>> - async isn't that usefull for us - we want to query databases, set 
>> variables and so on
>>no wait statements are possible
>>
>> 
>> 
>> > data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>>    no async 
>> 
>> 
>> 
>>
>> the script is on the site
>> http://wiki.freeswitch.org/wiki/PHP_ESL
>>
>> ---
>>
>> what can i do?
>> on our production server we use outbound socket connection and the 4 gig of 
>> memory are
>> eaten up in less than a day
>>
>> br
>>
>> --
>> Ing. Christian Löschenkohl
>> Technische Leitung, Forschung&  Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation&  Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 (0) 5 77 11 - 1000
>> F  +43 (0) 5 77 11 - 1002
>> E  christian.loeschenk...@xpirio.com
>>
>> ___
>> FreeSWITCH-users mailing list
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>> http://www.freeswitch.org
>>
>
>
>

-- 
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenk...@xpirio.com

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[Freeswitch-users] ATTN: Debian gurus and enthusiasts

[Michael Collins]

If you are a Debian person and have experience creating .debs then read
on...

Frank Carmickle has graciously volunteered to assist with creating a FS deb
config. I've heard that others have been doing something similar or are
interested. If you are such a person then please email Frank and me off list
so that we can organize the volunteers.

Thanks!
-Michael
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Re: [Freeswitch-users] FS Presence Implementation

[Jerry Richards]

I think you're referring to the SIP SIMPLE implementation as the default FS
presence mechanism.  This is fine and I can use that protocol.  The question
I still have regards the plain text content in the body of the SIP MESSAGE
method.  What is the format of this plain text for presence that is
compatible with the FS implementation?
 
Best Regards,
Jerry
 


  _  

From: Anthony Minessale [mailto:anthony.miness...@gmail.com] 
Sent: Tuesday, September 15, 2009 11:53 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Presence Implementation


the default config ships with presence enabled for SIP
if you have a phone that supports it, all you have to do is enable it on the
phone.



On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards 
wrote:



Also, is presence conveyed as any string?  Or is presence a predefined list
of status?

Best Regards,
Jerry



-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Tuesday, September 15, 2009 8:46 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS Presence Implementation

I would like to modify my SIP phone and my gateway to convey/exchange
presence information.  Could someone point me toward the FS presence
documentation?  I've seen bits and pieces.  Also, I think presence can be
communicated via more than one protocol.

Thanks And Best Regards,
Jerry


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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] How to Process Invalid extension in FS

[Michael Collins]

On Wed, Sep 16, 2009 at 1:23 AM, Ahmed Munir wrote:

> Hi,
>
> I'm newbie in FS. I want to know how to process invalid extension in FS?
> Because I want to prompt the IVR if invalid extension is dialled.
>
>
> Kindly advice me.
>
>
Is this an invalid extension that was dialed by a registered SIP phone or by
a caller who has already connected to an IVR? Just curious what your
application is.
-MC
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[Freeswitch-users] faxrx error 13 Unexpected message received

[Travis Stutsman]

In my attempts to receive a fax from a PSTN fax machine, the transaction
fails with error code 13 "Unexpected message received".  Verbose logging
is on for mod_fax. Here is an exerpt:

#
2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  CFR with
final frame tag
2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 84
2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12
2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected
CFR received in state 12
..

==
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not
successful - result (13) Unexpected message received.
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id:
**
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id:
SpanDSP Fax Ident
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution:
8031x3850
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate:
14400
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country:   
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor:
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: 
2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198

==
#

I have successfully received a fax from faxzero.com on this
installation, as have I successfully sent a fax from freeswitch to the
PSTN fax machine in question.

I've been digging, but there doesn't seem to be a whole lot of
information on faxing.  I guess I could use a bit of direction.  Any
input is much appreciated.


Thanks!
-- Travis

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Re: [Freeswitch-users] faxrx error 13 Unexpected message received

[Steve Underwood]

On 09/17/2009 12:08 AM, Travis Stutsman wrote:
> In my attempts to receive a fax from a PSTN fax machine, the transaction
> fails with error code 13 "Unexpected message received".  Verbose logging
> is on for mod_fax. Here is an exerpt:
>
> #
> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  CFR with
> final frame tag
> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 84
> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12
> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected
> CFR received in state 12
> ..
> 
> ==
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not
> successful - result (13) Unexpected message received.
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id:
> **
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id:
> SpanDSP Fax Ident
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages:   0
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution:
> 8031x3850
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate:
> 14400
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country:
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor:
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model:
> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198
> 
> ==
> #
>
> I have successfully received a fax from faxzero.com on this
> installation, as have I successfully sent a fax from freeswitch to the
> PSTN fax machine in question.
>
> I've been digging, but there doesn't seem to be a whole lot of
> information on faxing.  I guess I could use a bit of direction.  Any
> input is much appreciated.
>
You seem to have chopped out all the interesting parts of that log. A 
full log might say something interesting.

Steve


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[Freeswitch-users] session record does not for very short calls

[Frank @ Impact]

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] ATTN: Debian gurus and enthusiasts

[Michael Collins]

FYI,

MikeJ reminded me that we have .deb in tree already, so if you want to help
maintain that then hop on IRC and hook up w/ MikeJ for more info.
-MC

On Wed, Sep 16, 2009 at 8:48 AM, Michael Collins  wrote:

> If you are a Debian person and have experience creating .debs then read
> on...
>
> Frank Carmickle has graciously volunteered to assist with creating a FS deb
> config. I've heard that others have been doing something similar or are
> interested. If you are such a person then please email Frank and me off list
> so that we can organize the volunteers.
>
> Thanks!
> -Michael
>
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Re: [Freeswitch-users] faxrx error 13 Unexpected message received

[Rob Forman]

And make sure verbose is set to true in ./conf/autoload_configs/ 
fax.conf.xml.


On Sep 16, 2009, at 11:50 AM, Steve Underwood wrote:

> On 09/17/2009 12:08 AM, Travis Stutsman wrote:
>> In my attempts to receive a fax from a PSTN fax machine, the  
>> transaction
>> fails with error code 13 "Unexpected message received".  Verbose  
>> logging
>> is on for mod_fax. Here is an exerpt:
>>
>> #
>> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  CFR  
>> with
>> final frame tag
>> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff  
>> 13 84
>> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state  
>> 12
>> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected
>> CFR received in state 12
>> ..
>> = 
>> = 
>> = 
>> =
>> ==
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not
>> successful - result (13) Unexpected message received.
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id:
>> **
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id:
>> SpanDSP Fax Ident
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages:   0
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution:
>> 8031x3850
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate:
>> 14400
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status  
>> on
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country:
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor:
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model:
>> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198
>> = 
>> = 
>> = 
>> =
>> ==
>> #
>>
>> I have successfully received a fax from faxzero.com on this
>> installation, as have I successfully sent a fax from freeswitch to  
>> the
>> PSTN fax machine in question.
>>
>> I've been digging, but there doesn't seem to be a whole lot of
>> information on faxing.  I guess I could use a bit of direction.  Any
>> input is much appreciated.
>>
> You seem to have chopped out all the interesting parts of that log. A
> full log might say something interesting.
>
> Steve
>
>
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Re: [Freeswitch-users] faxrx error 13 Unexpected message received

[Travis Stutsman]

Alrighty.  Here is mod_fax from beginning to end.


#
2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:591 Raw read codec
activation Success L16 2
2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:607 Raw write codec
activation Success L16
2009-09-15 10:41:26.433382 [DEBUG] switch_channel.c:182
sofia/external/***...@**.***.**.*** receive message [AUDIO_SYNC]
2009-09-15 10:41:26.464633 [DEBUG] switch_core_io.c:232
sofia/external/***...@**.***.**.*** receive message
[TRANSCODING_NECESSARY]
2009-09-15 10:41:27.589676 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Carrier up (-2) in state 1
2009-09-15 10:41:27.761558 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Carrier down (-1) in state 1
2009-09-15 10:41:27.792809 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Carrier up (-2) in state 1
2009-09-15 10:41:27.870937 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:28.308454 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:28.355331 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:28.370956 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:28.824099 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:29.27231 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:29.261615 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
status is Abort (-8) in state 1
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete
in phase T30_PHASE_A_CED, state 1
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Starting
answer mode
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from
phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from
state 1 to 17
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident
'SpanDSP Fax Ident'
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI
without final frame tag
2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40
74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete
in phase T30_PHASE_B_TX, state 17
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ...0=
Store and forward Internet fax (T.37): Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    .0..=
Real-time Internet fax (T.38): Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    0...=
3G mobile network: Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. =
V.8 capabilities: Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. =
Preferred octets: 256 octets
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ...0=
Ready to transmit a fax document (polling): Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..1.=
Can receive fax: Set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..=
Supported data signalling rates: V.27 ter, V.29, and V.17
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. =
R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   1... =
2-D coding: Set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..10=
Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    10..=
Recording length: Unlimited
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .111 =
Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   1... =
Extension indicator: Set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..0.=
Compressed/uncompressed mode: Compressed
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    .1..=
Error correction mode (ECM): ECM
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. =
T.6 coding: Set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   1... =
Extension indicator: Set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ...0=
"Field not valid" supported: Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..0.=
Multiple selective polling: Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    .0..=
Polled sub-address: Not set
2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    0...=
T.43 co

Re: [Freeswitch-users] session record does not for very short calls

[João Mesquita]

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

[Diego Toro]

Hi, count on me for testing and answering questions on Windows and spanish 
support.

Diego 
http://lacarretade.blogspot.com/

--- On Wed, 9/16/09, Michael Collins  wrote:


From: Michael Collins 
Subject: Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH 
Subprojects
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, September 16, 2009, 9:56 AM





On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé  wrote:


Hi,

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

Merci!

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

Thanks again!
-MC


-Inline Attachment Follows-


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Re: [Freeswitch-users] reloadxml question

[Tihomir Culjaga]

well, it is a specific module for delivering some sort of services...
Actually, we are trying to build a SIP Application Server based on
freeswitch in core... the server will/should be in charge of delivering
various services e.g.

international call routing (route calls to international destinations
according to src_number, dst_prefix, user_qos_group, asr, acd )
hubbing (route traffic between carriers according to whatever you want)
local number portability (... no need to explain)
location based services (special services e.g. call distribution for
emergency services, local services ... lets say the user dials 101 and he
always gets the closest/preferred store... )
legal intercept (not allowed to talk about :P)
Voice VPNs
Network Announcement
Limited postpaid services
radius authorization (no accounting)
...


The idea is not to route calls through FS... it is going to be used just as
a ticketing server. FS needs to respond with an appropriate SIP message
containing routing decision. This way we make FS stateless as no calls are
going through and in the same time it holds all the services/routing logic
we need.

so, there are 2 main things:

1. all relevant routing parameters are loaded into a DKA memory map
2. all user parameter queries are done towards an OpenLDAP database



remember i was asking questions about performance and how to fine tune the
monster :P... well that's it.

i'm able to run at 400+ CPS on a single server.


we will see how it will finish ... so far, we are at 80% of the project and
I plan to switch some real traffic soon.

T.



On Wed, Sep 16, 2009 at 5:00 PM, Brian West  wrote:

> Might I ask what you are working on?Its interesting to hear what
> people are doing with FreeSWITCH.
>
> /b
>
> On Sep 16, 2009, at 9:17 AM, Tihomir Culjaga wrote:
>
> > perfect,
> >
> > thanks.
> >
> > T.
>
>
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Re: [Freeswitch-users] originate command sofia behaviour

[Michael Collins]

On Wed, Sep 16, 2009 at 7:26 AM, Alberto Escudero  wrote:

>
> I will like to update the wiki to spell out clearly the differences
> between this three commands
> I have a IVR running in 4600 and the FS box has IP address 192.168.46.15
>
> originate sofia/192.168.46.15/1001  4600
> originate sofia/internal/1...@192.168.46.15 4600
> originate sofia/internal/1001%192.168.46.15 4600
>
> The first originate places a call as a external gateway, not until
> registered phone 1001 answers the call is transfer to 4600
>
> The second and third originate command triggers extension 4600 Javascript
> IVR although 1001 has not answer
>
> Can anyone clarify me if this is the intended behavior also including the
> difference between % and @
>

The difference between % and @ is discussed here:
http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
-MC
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Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call?

[email lists]

Hello Mindaugas,

 

It was not really a matter of Freeswitch + mod_radius_cdr not being
good for me, or for what we needed it to do, but rather more of a
resource and time constraint based decision.  If we had 'C'
knowledgeable resources readily available, the Freeswitch and Radius
customizations required could've been completed, providing a more
streamlined setup using Freeswitch alone.  

 

However, due to resource and (in this case more importantly) time
constraints, I shifted to an alternative solution that I could more
quickly implement to meet our immediate needs.  I will eventually
re-visit relying solely on Freeswitch to simplify our setup and
probably ask a few more questions when that time comes.  :)

 

Vladimir

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Mindaugas Kezys
Sent: Wednesday, September 16, 2009 1:29 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS
messages for single call?

 

Can you tell why Freeswitch + mod_radius_cdr was not good for you?

 

Regards,

Mindaugas Kezys

http://www.kolmisoft.com

VoIP Billing and Routing Solutions

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
email lists
Sent: 2009 m. rugsėjo 16 d. 00:53
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS
messages for single call?

 

Thanks to those for the info and help on this issue.  Ultimately ended
up having to use alternative software for the radius piece (not
related to any shortfalls by Freeswitch).

 

Vlad

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Friday, September 11, 2009 11:31 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS
messages for single call?

 

set the variable process_cdr=false on that a_leg first thing in your
dialplan

On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy
 wrote:

It's normal to have to two records for a call - Start and Stop
message.

 From what i see - you have one start and stop for each leg of the
call.

Regards,
AK


email lists wrote:
>
> Hello,
>
>
>
> Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate
> RADIUS messages being generated for individual calls (sample
messages
> for one call below).  Looking at the "Acct-Unique-Session-Id" and
> "Acct-Session-Id" fields, it would appear that perhaps each call leg
> results in a pair of start/stop RADIUS messages; is this the
expected
> behavior?  If so, is there a way to disable RADIUS messaging for
what
> I presume is the "ingress" or A leg of the call?
>
>
>
> Any leads would be appreciated.
>
>
>
> Thanks in advance.
>
>
>
> Vladimir
>
>
>
> Thu Sep 10 10:37:25 2009
>
> Acct-Status-Type = Start
>
> Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004"
>
> User-Name = "8135793256"
>
> Freeswitch-Src = "8135793256"
>
> Freeswitch-CLID = "sipp"
>
> Freeswitch-Dst = "14043297226"
>
> Freeswitch-Dialplan = "XML"
>
> Framed-IP-Address = 50.46.50.55
>
> Freeswitch-Context = "public"
>
> Freeswitch-Source = "mod_sofia"
>
> Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700"
>
> NAS-Port = 0
>
> Acct-Delay-Time = 0
>
> NAS-IP-Address = 1.1.1.1
>
> Acct-Unique-Session-Id = "097c8472ff7bcec7"
>
> Timestamp = 1252604245
>
> Request-Authenticator = Verified
>
>
>
> Thu Sep 10 10:37:25 2009
>
> Acct-Status-Type = Start
>
> Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12"
>
> User-Name = "8135793256"
>
> Freeswitch-Src = "8135793256"
>
> Freeswitch-CLID = "sipp"
>
> Freeswitch-Dst = "14043297...@x.x.x.x"
>
> Freeswitch-Dialplan = "XML"
>
> Framed-IP-Address = 50.46.50.55
>
> Freeswitch-Context = "public"
>
> Freeswitch-Source = "mod_sofia"
>
> Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700"
>
> NAS-Port = 0
>
> Acct-Delay-Time = 0
>
> NAS-IP-Address = 1.1.1.1
>
> Acct-Unique-Session-Id = "53f729e173e8c0a9"
>
> Timestamp = 1252604245
>
> Request-Authenticator = Verified
>
>
>
> Thu Sep 10 10:37:57 2009
>
> Acct-Status-Type = Stop
>
> Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004"
>
> Freeswitch-Hangupcause = Normal-Unspecified
>
> User-Name = "8135793256"
>
> Freeswitch-Src = "8135793256"
>
> Freeswitch-CLID = "sipp"
>
> Freeswitch-Dst = "14043297226"
>
> Freeswitch-Dialplan = "XML"
>
> Framed-IP-Address = 50.46.50.55
>
> Freeswitch-Context = "public"
>
> Freeswitch-Source = "mod_sofia"
>

Re: [Freeswitch-users] originate command sofia behaviour

[Alberto Escudero]

The problem i am facing is the following:

Extension 4600 is a Javascript IVR that starts by session.aswer()

I want to originate a call to leg 1 and then connected to the IVR when the
leg 1 has answered.

If I run

originate sofia/192.168.46.15/1001  4600
call is transfer to extension 4600 *IVR* after 1001 answers the call

If I run
originate sofia/internal/1...@192.168.46.15 4600
the IVR starts BEFORE user 1001 has answered?

What is the best way to:

Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR)
after leg 1 has answered the call?

/aep




-- 
Stopping junk mailers is good for the environment

> On Wed, Sep 16, 2009 at 7:26 AM, Alberto Escudero 
> wrote:
>
>>
>> I will like to update the wiki to spell out clearly the differences
>> between this three commands
>> I have a IVR running in 4600 and the FS box has IP address 192.168.46.15
>>
>> originate sofia/192.168.46.15/1001  4600
>> originate sofia/internal/1...@192.168.46.15 4600
>> originate sofia/internal/1001%192.168.46.15 4600
>>
>> The first originate places a call as a external gateway, not until
>> registered phone 1001 answers the call is transfer to 4600
>>
>> The second and third originate command triggers extension 4600
>> Javascript
>> IVR although 1001 has not answer
>>
>> Can anyone clarify me if this is the intended behavior also including
>> the
>> difference between % and @
>>
>
> The difference between % and @ is discussed here:
> http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
> -MC
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Re: [Freeswitch-users] originate command sofia behaviour

[Michael Collins]

On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero wrote:

> The problem i am facing is the following:
>
> Extension 4600 is a Javascript IVR that starts by session.aswer()
>
> I want to originate a call to leg 1 and then connected to the IVR when the
> leg 1 has answered.
>
> If I run
>
> originate sofia/192.168.46.15/1001  4600
> call is transfer to extension 4600 *IVR* after 1001 answers the call
>
> If I run
> originate sofia/internal/1...@192.168.46.15 4600
> the IVR starts BEFORE user 1001 has answered?
>
> What is the best way to:
>
> Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR)
> after leg 1 has answered the call?
>

You can try ignoring early media to force the A-leg to answer before
anything else happens. Try this and let us know if it does what you want:
originate {ignore_early_media=true} sofia/internal/1...@192.168.46.15 4600

You can probably look at the SIP traces of the two options you've tried
(without ignoring early media) to confirm that you're getting media prior to
answer when doing "originate sofia/internal/1...@192.168.46.15 4600" -
probably in one case you get a 180 and in the other a 183. Check it out and
let us know. :)
-MC
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Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects

[Muhammad Shahzad]

I am also available for FS configuration on various Linux distributions and
Wiki / documentation.

Thank you.


On Wed, Sep 16, 2009 at 11:44 PM, Diego Toro  wrote:

> Hi, count on me for testing and answering questions on Windows and spanish
> support.
>
> Diego
> http://lacarretade.blogspot.com/
>
> --- On *Wed, 9/16/09, Michael Collins * wrote:
>
>
> From: Michael Collins 
> Subject: Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With
> FreeSWITCH Subprojects
> To: freeswitch-users@lists.freeswitch.org
> Date: Wednesday, September 16, 2009, 9:56 AM
>
>
>
>
> On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mahé 
> http://us.mc335.mail.yahoo.com/mc/compose?to=t.m...@telemaque.fr>
> > wrote:
>
>> Hi,
>>
>> Count on me for answering questions on IRC when I'm in, and for
>> subprojects I'm in too as you know ;)
>>
> Merci!
>
> Okay, what's your IRC nick and when are you generally on line? Also, I'm
> pretty sure that you're fluent in French which is good because we need more
> multilingual people out there. Last question: what are your areas of
> expertise? I'd like to keep a list of people and what they're good at so we
> know whom to ask first when questions come up.
>
> Thanks again!
> -MC
>
>
> -Inline Attachment Follows-
>
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-- 
Muhammad Shahzad
---
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Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call?

[Mindaugas Kezys]

Thank you for your answer.

 

Regards,

Mindaugas Kezys

http://www.kolmisoft.com

VoIP Billing and Routing Solutions

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of email
lists
Sent: 2009 m. rugsėjo 16 d. 21:39
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for
single call?

 

Hello Mindaugas,

 

It was not really a matter of Freeswitch + mod_radius_cdr not being good for
me, or for what we needed it to do, but rather more of a resource and time
constraint based decision.  If we had 'C' knowledgeable resources readily
available, the Freeswitch and Radius customizations required could've been
completed, providing a more streamlined setup using Freeswitch alone.  

 

However, due to resource and (in this case more importantly) time
constraints, I shifted to an alternative solution that I could more quickly
implement to meet our immediate needs.  I will eventually re-visit relying
solely on Freeswitch to simplify our setup and probably ask a few more
questions when that time comes.  :)

 

Vladimir

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Mindaugas Kezys
Sent: Wednesday, September 16, 2009 1:29 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for
single call?

 

Can you tell why Freeswitch + mod_radius_cdr was not good for you?

 

Regards,

Mindaugas Kezys

http://www.kolmisoft.com

VoIP Billing and Routing Solutions

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of email
lists
Sent: 2009 m. rugsėjo 16 d. 00:53
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for
single call?

 

Thanks to those for the info and help on this issue.  Ultimately ended up
having to use alternative software for the radius piece (not related to any
shortfalls by Freeswitch).

 

Vlad

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Friday, September 11, 2009 11:31 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for
single call?

 

set the variable process_cdr=false on that a_leg first thing in your
dialplan

On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy
 wrote:

It's normal to have to two records for a call - Start and Stop message.

 From what i see - you have one start and stop for each leg of the call.

Regards,
AK


email lists wrote:
>
> Hello,
>
>
>
> Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate
> RADIUS messages being generated for individual calls (sample messages
> for one call below).  Looking at the "Acct-Unique-Session-Id" and
> "Acct-Session-Id" fields, it would appear that perhaps each call leg
> results in a pair of start/stop RADIUS messages; is this the expected
> behavior?  If so, is there a way to disable RADIUS messaging for what
> I presume is the "ingress" or A leg of the call?
>
>
>
> Any leads would be appreciated.
>
>
>
> Thanks in advance.
>
>
>
> Vladimir
>
>
>
> Thu Sep 10 10:37:25 2009
>
> Acct-Status-Type = Start
>
> Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004"
>
> User-Name = "8135793256"
>
> Freeswitch-Src = "8135793256"
>
> Freeswitch-CLID = "sipp"
>
> Freeswitch-Dst = "14043297226"
>
> Freeswitch-Dialplan = "XML"
>
> Framed-IP-Address = 50.46.50.55
>
> Freeswitch-Context = "public"
>
> Freeswitch-Source = "mod_sofia"
>
> Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700"
>
> NAS-Port = 0
>
> Acct-Delay-Time = 0
>
> NAS-IP-Address = 1.1.1.1
>
> Acct-Unique-Session-Id = "097c8472ff7bcec7"
>
> Timestamp = 1252604245
>
> Request-Authenticator = Verified
>
>
>
> Thu Sep 10 10:37:25 2009
>
> Acct-Status-Type = Start
>
> Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12"
>
> User-Name = "8135793256"
>
> Freeswitch-Src = "8135793256"
>
> Freeswitch-CLID = "sipp"
>
> Freeswitch-Dst = "14043297...@x.x.x.x"
>
> Freeswitch-Dialplan = "XML"
>
> Framed-IP-Address = 50.46.50.55
>
> Freeswitch-Context = "public"
>
> Freeswitch-Source = "mod_sofia"
>
> Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700"
>
> NAS-Port = 0
>
> Acct-Delay-Time = 0
>
> NAS-IP-Address = 1.1.1.1
>
> Acct-Unique-Session-Id = "53f729e173e8c0a9"
>
> Timestamp = 1252604245
>
> Request-Authenticator = Verified
>
>
>
> Thu Sep 10 10:37:57 2009
>
> Acct-Status-Type = Stop
>
> Acct-Session-Id = 

[Freeswitch-users] mod_conference performance

[Роберт Тверитнер]

Hi guys!

I've tested FreeSWITCH conference module performance trying to figure out
maximum number of simultaneous calls my FS box can serve. It took all 100%
of CPU with only 50 calls (in average depending on conference rate) and
"leaking stream handle" messages started appearing.

The environment I was testing in:
OS - Windows Server 2007 SP1 64 Bit
CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz
RAM - 2 GB
FreeSwitch version 1.0.4 (14460)

I've written a test program that used to originate calls once in 5 seconds
from the other box. These calls were routed to particular conference room I
was testing. I had a number of rooms with different rate (8000-32000) and
interval (20,30) settings and with perpetual-sound turned on steraming music
continiously. I've switched off all unnecessary modules, but left logging on
in order to trace what was happening later. Client test softphone used
respective speex codec according to conference room rate.

This is a dialplan I used:





















My questions are:
Do you know any way I can increase my FS conference capacity? What do I have
to tune in FS or in my box?

Best regards, Robert.
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Re: [Freeswitch-users] originate command sofia behaviour

[Alberto Escudero]

Yes, it did work! No we do not need to pay for several  GSM calls to test
a IVR script!

/aep and gmaruzz
-- 
Stopping junk mailers is good for the environment

> On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero
> wrote:
>
>> The problem i am facing is the following:
>>
>> Extension 4600 is a Javascript IVR that starts by session.aswer()
>>
>> I want to originate a call to leg 1 and then connected to the IVR when
>> the
>> leg 1 has answered.
>>
>> If I run
>>
>> originate sofia/192.168.46.15/1001  4600
>> call is transfer to extension 4600 *IVR* after 1001 answers the call
>>
>> If I run
>> originate sofia/internal/1...@192.168.46.15 4600
>> the IVR starts BEFORE user 1001 has answered?
>>
>> What is the best way to:
>>
>> Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR)
>> after leg 1 has answered the call?
>>
>
> You can try ignoring early media to force the A-leg to answer before
> anything else happens. Try this and let us know if it does what you want:
> originate {ignore_early_media=true} sofia/internal/1...@192.168.46.15 4600
>
> You can probably look at the SIP traces of the two options you've tried
> (without ignoring early media) to confirm that you're getting media prior
> to
> answer when doing "originate sofia/internal/1...@192.168.46.15 4600" -
> probably in one case you get a 180 and in the other a 183. Check it out
> and
> let us know. :)
> -MC
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>



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[Freeswitch-users] Vestec Speech Recognition Integration ?

[Gerry Hull]

Has anyone integrated Vestec Speech Recognition with FreeSwitch?  It's
$99/port...http://www.vestec.ca/

They have a C/C++ api, looks pretty simple.   Alas, no MRCP until 2010.
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Re: [Freeswitch-users] mod_conference performance

[João Mesquita]

I would be really interested to replay your test on Linux. Would you be
willing to provide me all the details and relevant files so I can reproduce
the test with a Linux box here?

If yes, contact me offlist and we can work together on this.

Regards,

jmesquita

On Wed, Sep 16, 2009 at 2:56 PM, Роберт Тверитнер wrote:

> Hi guys!
>
> I've tested FreeSWITCH conference module performance trying to figure out
> maximum number of simultaneous calls my FS box can serve. It took all 100%
> of CPU with only 50 calls (in average depending on conference rate) and
> "leaking stream handle" messages started appearing.
>
> The environment I was testing in:
> OS - Windows Server 2007 SP1 64 Bit
> CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz
> RAM - 2 GB
> FreeSwitch version 1.0.4 (14460)
>
> I've written a test program that used to originate calls once in 5 seconds
> from the other box. These calls were routed to particular conference room I
> was testing. I had a number of rooms with different rate (8000-32000) and
> interval (20,30) settings and with perpetual-sound turned on steraming music
> continiously. I've switched off all unnecessary modules, but left logging on
> in order to trace what was happening later. Client test softphone used
> respective speex codec according to conference room rate.
>
> This is a dialplan I used:
> 
>  break="on-true">
> 
> 
>  break="on-true">
> 
> 
>  break="on-true">
> 
> 
>  break="on-true">
> 
> 
>  break="on-true">
> 
> 
>  break="on-true">
> 
> 
> 
>
> My questions are:
> Do you know any way I can increase my FS conference capacity? What do I
> have to tune in FS or in my box?
>
> Best regards, Robert.
>
>
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Re: [Freeswitch-users] mod_conference performance

[Brian West]

I would be very interested in this also.

/b

On Sep 16, 2009, at 4:52 PM, João Mesquita wrote:

> I would be really interested to replay your test on Linux. Would you  
> be willing to provide me all the details and relevant files so I can  
> reproduce the test with a Linux box here?
>
> If yes, contact me offlist and we can work together on this.
>
> Regards,
>
> jmesquita


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Re: [Freeswitch-users] mod_conference performance

[Jay Binks]

A few thing stuck out to me ...
Mainly 50 calls and transcoding speex.

Try it again with g711 and see how you go.

Also not sure windows 7 is going to perform as good as other options,   
could be wrong though .

Jay


On 17/09/2009, at 3:56, Роберт Тверитнер  
 wrote:

> Hi guys!
>
> I've tested FreeSWITCH conference module performance trying to  
> figure out maximum number of simultaneous calls my FS box can serve.  
> It took all 100% of CPU with only 50 calls (in average depending on  
> conference rate) and "leaking stream handle" messages started  
> appearing.
>
> The environment I was testing in:
> OS - Windows Server 2007 SP1 64 Bit
> CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz
> RAM - 2 GB
> FreeSwitch version 1.0.4 (14460)
>
> I've written a test program that used to originate calls once in 5  
> seconds from the other box. These calls were routed to particular  
> conference room I was testing. I had a number of rooms with  
> different rate (8000-32000) and interval (20,30) settings and with  
> perpetual-sound turned on steraming music continiously. I've  
> switched off all unnecessary modules, but left logging on in order  
> to trace what was happening later. Client test softphone used  
> respective speex codec according to conference room rate.
>
> This is a dialplan I used:
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>  data="$...@ultrawideband20"/>
> 
> 
>  data="$...@ultrawideband30"/>
> 
> 
>
> My questions are:
> Do you know any way I can increase my FS conference capacity? What  
> do I have to tune in FS or in my box?
>
> Best regards, Robert.
>
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Re: [Freeswitch-users] mod_conference performance

[Brian West]

Yah speex is a cpu hog (while you can tune it to use less)!  Granted  
it uses less bandwidth but on the server side it doesn't scale very  
well.

/b
On Sep 16, 2009, at 5:11 PM, Jay Binks wrote:

> A few thing stuck out to me ...
> Mainly 50 calls and transcoding speex.
>
> Try it again with g711 and see how you go.
>
> Also not sure windows 7 is going to perform as good as other options,
> could be wrong though .
>
> Jay


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[Freeswitch-users] local stream problem - moh

[Christian Löschenkohl]

hello

since installing the latest trunk 14894 my local streams / moh don't work 
anymore
no config file has changed, the files are in place



show_local_stream outputs

default,/opt/freeswitch/sounds/music/8000
moh/16000,/opt/freeswitch/sounds/music/16000
moh/8000,/opt/freeswitch/sounds/music/8000


console prints

2009-09-17 01:15:01.496277 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.516281 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.516281 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.536282 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.536282 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.556286 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.556286 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.576286 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.576286 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.596283 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.596283 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.616261 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.616261 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.636304 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.636304 [ERR] switch_core_file.c:152 File [(null)] not 
created!
2009-09-17 01:15:01.656273 [WARNING] mod_local_stream.c:318 Unknown source 
local_stream://moh, trying 'default'
2009-09-17 01:15:01.656273 [ERR] switch_core_file.c:152 File [(null)] not 
created!

-
and yes, files are there

ls /opt/freeswitch/sounds/music/8000

danza-espanola-op-37-h-142-xii-arabesca.wav  
partita-no-3-in-e-major-bwv-1006-1-preludio.wav  ponce-preludio-in-e-major.wav  
suite-espanola-op-47-leyenda.wav

-

br

-- 
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenk...@xpirio.com

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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] faxrx error 13 Unexpected message received

[Steve Underwood]

Hi Travis,

That's a pretty weird call. It looks like you have a long delayed echo. 
See below.

On 09/17/2009 01:21 AM, Travis Stutsman wrote:
> Alrighty.  Here is mod_fax from beginning to end.
>
>
> #
> 2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:591 Raw read codec
> activation Success L16 2
> 2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:607 Raw write codec
> activation Success L16
> 2009-09-15 10:41:26.433382 [DEBUG] switch_channel.c:182
> sofia/external/***...@**.***.**.*** receive message [AUDIO_SYNC]
> 2009-09-15 10:41:26.464633 [DEBUG] switch_core_io.c:232
> sofia/external/***...@**.***.**.*** receive message
> [TRANSCODING_NECESSARY]
> 2009-09-15 10:41:27.589676 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Carrier up (-2) in state 1
> 2009-09-15 10:41:27.761558 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Carrier down (-1) in state 1
> 2009-09-15 10:41:27.792809 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Carrier up (-2) in state 1
> 2009-09-15 10:41:27.870937 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:28.308454 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:28.355331 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:28.370956 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:28.824099 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:29.27231 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:29.261615 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal
> status is Abort (-8) in state 1
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete
> in phase T30_PHASE_A_CED, state 1
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Starting
> answer mode
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from
> phase T30_PHASE_A_CED to T30_PHASE_B_TX
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from
> state 1 to 17
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident
> 'SpanDSP Fax Ident'
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI
> without final frame tag
> 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40
> 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete
> in phase T30_PHASE_B_TX, state 17
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ...0=
> Store and forward Internet fax (T.37): Not set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    .0..=
> Real-time Internet fax (T.38): Not set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    0...=
> 3G mobile network: Not set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. =
> V.8 capabilities: Not set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. =
> Preferred octets: 256 octets
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ...0=
> Ready to transmit a fax document (polling): Not set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..1.=
> Can receive fax: Set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..=
> Supported data signalling rates: V.27 ter, V.29, and V.17
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. =
> R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   1... =
> 2-D coding: Set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..10=
> Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    10..=
> Recording length: Unlimited
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .111 =
> Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   1... =
> Extension indicator: Set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ..0.=
> Compressed/uncompressed mode: Compressed
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    .1..=
> Error correction mode (ECM): ECM
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. =
> T.6 coding: Set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30   1... =
> Extension indicator: Set
> 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30    ...0=

Re: [Freeswitch-users] session record does not for very short calls

[Nandy Dagondon]

this makes sense. a workaround would be to provide an optional variable to
delete recording file if it's less than N seconds. otherwise, it defaults to
a preset duration.

/nandy


On Thu, Sep 17, 2009 at 7:46 AM, Seven Du  wrote:

> 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?
> >
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > ___
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> ___
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Re: [Freeswitch-users] local stream problem - moh

[Anthony Minessale]

Update again, already fixed

On Sep 16, 2009 6:25 PM, "Christian Löschenkohl" <
christian.loeschenk...@xpirio.com> wrote:

hello

since installing the latest trunk 14894 my local streams / moh don't work
anymore
no config file has changed, the files are in place



show_local_stream outputs

default,/opt/freeswitch/sounds/music/8000
moh/16000,/opt/freeswitch/sounds/music/16000
moh/8000,/opt/freeswitch/sounds/music/8000


console prints

2009-09-17 01:15:01.496277 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.516281 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.516281 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.536282 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.536282 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.556286 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.556286 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.576286 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.576286 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.596283 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.596283 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.616261 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.616261 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.636304 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.636304 [ERR] switch_core_file.c:152 File [(null)] not
created!
2009-09-17 01:15:01.656273 [WARNING] mod_local_stream.c:318 Unknown source
local_stream://moh, trying 'default'
2009-09-17 01:15:01.656273 [ERR] switch_core_file.c:152 File [(null)] not
created!

-
and yes, files are there

ls /opt/freeswitch/sounds/music/8000

danza-espanola-op-37-h-142-xii-arabesca.wav
 partita-no-3-in-e-major-bwv-1006-1-preludio.wav
 ponce-preludio-in-e-major.wav  suite-espanola-op-47-leyenda.wav

-

br

--
Ing. Christian Löschenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenk...@xpirio.com

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Re: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.)

[Karl Vesterling]

Folks;

I give credit where credit is due, and I thank Brian K. West

What For:
This was found to be a compounded problem.  (Cisco was part of it...   
But the real problem was the linux kernel...)
Suffice it to say, without the kernel bug, the cisco bug wouldn't have  
been easily found.


What Kernel Bug:
It's a kernel bug that corrupted the sqlite database.
This caused Freeswitch to refuse the phones registration request.
This in turn caused the phones to re-register.
Problem was, with 10 phones, 6 lines each, perpetually registering on  
a 100Mbps LAN, well, you can imagine the overhead.
This created severe latency with Freeswitch, and manifested as dropped  
calls, one way audio, and the more phones you had, the worse the  
problem was.


Workaround for problem was to use a ramdisk (tmpfs)  for the database  
- (Big Thanks to bkw!)


So far, it's been 24 hours, and all systems are nice and stable.  (no  
negative reports yet (fingers crossed)).


Brian (and Folks);
If this is stable through Friday (and there's no reason to think it  
won't be),  I will take the time to document the problem, basic  
configuration,  and the workaround for the problem on the Wiki this  
weekend.


Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3231 x0

On Sep 14, 2009, at 9:22 AM, Brian West wrote:


HAHA I couldn't have said this better!

/b

On Sep 14, 2009, at 8:17 AM, Anthony Minessale wrote:


The first hint was when the firmware rev began with the letters POS



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Re: [Freeswitch-users] FS Presence Implementation

[Anthony Minessale]

we support both application/dialog-info+xml (snom maybe a few others, I
can't keep track) and pidf used by polycom and eyebeam



On Wed, Sep 16, 2009 at 10:50 AM, Jerry Richards  wrote:

>  I think you're referring to the SIP SIMPLE implementation as the default
> FS presence mechanism.  This is fine and I can use that protocol.  The
> question I still have regards the plain text content in the body of the SIP
> MESSAGE method.  What is the format of this plain text for presence that is
> compatible with the FS implementation?
>
> Best Regards,
> Jerry
>
>
>  --
> *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
> *Sent:* Tuesday, September 15, 2009 11:53 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] FS Presence Implementation
>
> the default config ships with presence enabled for SIP
> if you have a phone that supports it, all you have to do is enable it on
> the phone.
>
>
> On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards <
> jerry.richa...@teotech.com> wrote:
>
>>
>> Also, is presence conveyed as any string?  Or is presence a predefined
>> list
>> of status?
>>
>> Best Regards,
>> Jerry
>>
>>
>> -Original Message-
>> From: Jerry Richards [mailto:jerry.richa...@teotech.com]
>> Sent: Tuesday, September 15, 2009 8:46 AM
>> To: 'freeswitch-users@lists.freeswitch.org'
>> Subject: FS Presence Implementation
>>
>> I would like to modify my SIP phone and my gateway to convey/exchange
>> presence information.  Could someone point me toward the FS presence
>> documentation?  I've seen bits and pieces.  Also, I think presence can be
>> communicated via more than one protocol.
>>
>> Thanks And Best Regards,
>> Jerry
>>
>>
>> ___
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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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>
>


-- 
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] A real puzzler for you guys... (perpetual registration causes denial of service.)

[Jason White]

Karl Vesterling  wrote:
 
> What Kernel Bug:
> It's a kernel bug that corrupted the sqlite database.
> This caused Freeswitch to refuse the phones registration request.

Please take this up with your Linux distribution as a bug report related to
the kernel, and persist with it until it's sorted out.

The more that users do this, the more kernel bugs will get fixed.

We're all responsible to some extent for the quality of our free/open-source
operating systems.


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[Freeswitch-users] Compile error

[Luis M. Zuccolo]

Hi:

Since svn version 13523 to current I get this error:

make[5]: swig: Command not found
make[5]: *** [mod_lua_wrap.cpp] Error 127
make[4]: *** [all] Error 1
make[3]: *** [mod_lua-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 + FreeSWITCH Build Complete ---+
 + FreeSWITCH has been successfully built.  +
 + Install by running:  +
 +  +
 +   make install   +
 +--+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

Was there any change from that version?
Thanks in advance

__
Correo Yahoo!
Espacio para todos tus mensajes, antivirus y antispam ¡gratis! 
¡Abrí tu cuenta ya! - http://correo.yahoo.com.ar


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Re: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.)

[Tuyan Özipek]

Which distro is this?

/tyn

On Thu, Sep 17, 2009 at 12:07 AM, Jason White  wrote:
> Karl Vesterling  wrote:
>
>> What Kernel Bug:
>> It's a kernel bug that corrupted the sqlite database.
>> This caused Freeswitch to refuse the phones registration request.
>
> Please take this up with your Linux distribution as a bug report related to
> the kernel, and persist with it until it's sorted out.
>
> The more that users do this, the more kernel bugs will get fixed.
>
> We're all responsible to some extent for the quality of our free/open-source
> operating systems.
>
>
> ___
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Re: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.)

[Gabriel Gunderson]

On Wed, Sep 16, 2009 at 9:10 PM, Karl Vesterling  wrote:
> It's a kernel bug that corrupted the sqlite database.
> This caused Freeswitch to refuse the phones registration request.
> This in turn caused the phones to re-register.
> Problem was, with 10 phones, 6 lines each, perpetually registering on a
> 100Mbps LAN, well, you can imagine the overhead.
> This created severe latency with Freeswitch, and manifested as dropped
> calls, one way audio, and the more phones you had, the worse the problem
> was.

> Workaround for problem was to use a ramdisk (tmpfs)  for the database - (Big
> Thanks to bkw!)

The data in the sqlite db doesn't need to survive a reboot?  Any
benefit if it does?

Also, I've read that the ramdisk isn't that different than what the
kernel already does to keep things in memory (yielding very little
gain).

Thoughts on this?

Gabe

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Re: [Freeswitch-users] Compile error

[Frank Carmickle]

On Thu, Sep 17, Luis M. Zuccolo wrote:
> Hi:
> 
> Since svn version 13523 to current I get this error:
> 
> make[5]: swig: Command not found

You must install swig.  If your on debian apt-get install swig.  If your not 
see http://www.swig.org/

HTH
--FC

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