Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Saurabh Aggarwal]

Ok, my bad. Ethereal for some reason was showing only the first fragment 
(ethereal bug?).
 
But, now it seems I have hit another problem - it seems that the SIP invites 
(which are fragmented) are being dropped by the firewall in between us and the 
SIP provider. Is it possible to shrink the size of the SIP invite so that it 
fits in a single packet? Any optional stuff in the SIP invite that is sent, 
that can be thrown away?
 
-Saurabh



From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 18 Nov 2008 07:34:11 
+Subject: [Freeswitch-users] SIP Invite IP fragmentation issue

I am having an *odd* issue, which i am not sure freeswitch is to be blamed for. 
Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, 
but when I look at the TCP dump (on the same machine as freeswitch), I see that 
only the first packet of the fragment is captured. Is freeswitch trying to do 
its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? 
I created a small program to send UDP packets of 2000 bytes, and also tried 
with ping -s 2000, and both were successful, so am leaning towards blaming 
Freeswitch. Any suggestions? -Saurabh 



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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Iñaki Baz Castillo]

2008/11/18 Saurabh Aggarwal <[EMAIL PROTECTED]>:
> Ok, my bad. Ethereal for some reason was showing only the first fragment
> (ethereal bug?).
>
> But, now it seems I have hit another problem - it seems that the SIP invites
> (which are fragmented) are being dropped by the firewall in between us and
> the SIP provider. Is it possible to shrink the size of the SIP invite so
> that it fits in a single packet? Any optional stuff in the SIP invite that
> is sent, that can be thrown away?

Welcome to the reason for which IETF is moving to SIP TCP/SCTP ;)

-- 
Iñaki Baz Castillo
<[EMAIL PROTECTED]>
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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Saurabh Aggarwal]

enabling compact headers - what is that?
 
-Saurabh



Date: Tue, 18 Nov 2008 04:29:28 -0600From: [EMAIL PROTECTED]: [EMAIL 
PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueIts not 
really possible other then enabling compact headers or by getting rid of codecs 
that you don’t actually want to use... Another thing you could do is get your 
broken ISP to fix their firewall... It is not correct to just drop fragmented 
packets just because they are fragmented.. This is something that will happen 
on a regular basis on the internet as not everything has an MTU of 1500



From: Saurabh Aggarwal <[EMAIL PROTECTED]>Reply-To: 
Date: Tue, 18 Nov 2008 10:19:55 +To: 
Subject: Re: [Freeswitch-users] SIP 
Invite IP fragmentation issueOk, my bad. Ethereal for some reason was showing 
only the first fragment (ethereal bug?). But, now it seems I have hit another 
problem - it seems that the SIP invites (which are fragmented) are being 
dropped by the firewall in between us and the SIP provider. Is it possible to 
shrink the size of the SIP invite so that it fits in a single packet? Any 
optional stuff in the SIP invite that is sent, that can be thrown away? -Saurabh



From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 18 Nov 2008 07:34:11 
+Subject: [Freeswitch-users] SIP Invite IP fragmentation issueI am having 
an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, 
the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I 
look at the TCP dump (on the same machine as freeswitch), I see that only the 
first packet of the fragment is captured. Is freeswitch trying to do its own IP 
fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a 
small program to send UDP packets of 2000 bytes, and also tried with ping -s 
2000, and both were successful, so am leaning towards blaming Freeswitch. Any 
suggestions? -Saurabh 



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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Iñaki Baz Castillo]

2008/11/18 Saurabh Aggarwal <[EMAIL PROTECTED]>:
> enabling compact headers - what is that?

SIP allows compact headers names for a few heades:

 From = f
 To = t
 Via = v
 ...


-- 
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<[EMAIL PROTECTED]>
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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Ken Rice]

Its not really possible other then enabling compact headers or by getting
rid of codecs that you don¹t actually want to use... Another thing you could
do is get your broken ISP to fix their firewall... It is not correct to just
drop fragmented packets just because they are fragmented.. This is something
that will happen on a regular basis on the internet as not everything has an
MTU of 1500



From: Saurabh Aggarwal <[EMAIL PROTECTED]>
Reply-To: 
Date: Tue, 18 Nov 2008 10:19:55 +
To: 
Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue

Ok, my bad. Ethereal for some reason was showing only the first fragment
(ethereal bug?).
 
But, now it seems I have hit another problem - it seems that the SIP invites
(which are fragmented) are being dropped by the firewall in between us and
the SIP provider. Is it possible to shrink the size of the SIP invite so
that it fits in a single packet? Any optional stuff in the SIP invite that
is sent, that can be thrown away?
 
-Saurabh



From: [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Date: Tue, 18 Nov 2008 07:34:11 +
Subject: [Freeswitch-users] SIP Invite IP fragmentation issue

I am having an *odd* issue, which i am not sure freeswitch is to be blamed
for.
 
Sometimes, the SIP invites are bigger than 1500 bytes causing IP
fragmentation, but when I look at the TCP dump (on the same machine as
freeswitch), I see that only the first packet of the fragment is captured.
Is freeswitch trying to do its own IP fragmentation or is it relying on
underlying linux (kernel 2.6.18)?
 
I created a small program to send UDP packets of 2000 bytes, and also tried
with ping -s 2000, and both were successful, so am leaning towards blaming
Freeswitch.
 
Any suggestions?
 
-Saurabh
 





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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Jonas Gauffin]

The rfc also describes why:

SIP provides a mechanism to represent common header field names in an
   abbreviated form.  This may be useful when messages would otherwise
   become too large to be carried on the transport available to it
   (exceeding the maximum transmission unit (MTU) when using UDP, for
   example).  These compact forms are defined in Section 20.  A compact
   form MAY be substituted for the longer form of a header field name at
   any time without changing the semantics of the message.  A header
   field name MAY appear in both long and short forms within the same
   message.  Implementations MUST accept both the long and short forms
   of each header name.


On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo <[EMAIL PROTECTED]> wrote:

> 2008/11/18 Saurabh Aggarwal <[EMAIL PROTECTED]>:
> > enabling compact headers - what is that?
>
> SIP allows compact headers names for a few heades:
>
>  From = f
>  To = t
>  Via = v
>  ...
>
>
> --
> Iñaki Baz Castillo
> <[EMAIL PROTECTED]>
> ___
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Re: [Freeswitch-users] DTMF

[Baskar]

Hi,

I am working on Dtmf signal on barging

I have added these line to default.xml

   
 
   
   
   
   
 
   

   
 
   
   
 
   

i eavesdrop on extension 1003 by dialing 881003

1003 is connected to mobile no

but when i dial 881003 it get hangup

i have attached the full default.xml separately.barging is not working
through default.xml

And another thing DTMF signal is *not working* through* even socket api
command *

I tried in conference also when we manually done in softphone it work . when
i press the # button it hangup and * for mute  etc. it works fine but when i
pass these through event socket it does not work why?

if i pass the DTMF digits in api  command it receive a beep sound but the
process is not done.

*i tried i n conference the output is :*

api conference 3001 dtmf 15 #

Content-Type: api/response
Content-Length: 16

OK sent # to 15

15 is member id
# is to hangup
but it does not get hangup through api command
*
In barging also same problem *

api uuid_send_dtmf f7666a65-5fc4-4199-bb6c-95a5e22d4515 "3210*"

Content-Type: api/response
Content-Length: 14

-ERR no reply

I get the beep sound and there is no process done

what should be done please guide me.

-- 
Warm Regards,
N.Baskar


  
   
   
   
   
   
 
   
	   
   
   
   
	   
	   
   
   
	   
	   
	   
	   
		
		
		
		
		
		
		
		
		
	
		
		
  


   
  
  
  
	  
  
  
  
  
  
   
  
 
   
   
  
		
		
		
  


   
 
   
   
   
   
 
   


  


  


  

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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Saurabh Aggarwal]

Thanks, how do I "enable" this in freeswitch? Can this be done through the SIP 
configuration file?
 
-Saurabh



Date: Tue, 18 Nov 2008 12:05:18 +0100From: [EMAIL PROTECTED]: [EMAIL 
PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueThe rfc 
also describes why:SIP provides a mechanism to represent common header field 
names in an   abbreviated form.  This may be useful when messages would 
otherwise   become too large to be carried on the transport available to it   
(exceeding the maximum transmission unit (MTU) when using UDP, for   example).  
These compact forms are defined in Section 20.  A compact   form MAY be 
substituted for the longer form of a header field name at   any time without 
changing the semantics of the message.  A header   field name MAY appear in 
both long and short forms within the same   message.  Implementations MUST 
accept both the long and short forms   of each header name.
On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo <[EMAIL PROTECTED]> wrote:
2008/11/18 Saurabh Aggarwal <[EMAIL PROTECTED]>:> enabling compact headers - 
what is that?SIP allows compact headers names for a few heades: From = f To = t 
Via = v ...--Iñaki Baz Castillo<[EMAIL 
PROTECTED]>___Freeswitch-users 
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[Freeswitch-users] Problems with DTMF and * on inbound leg

[Dennis]

i am using socket outbound and if an inbound call comes in, i answer
the call and play a soundfile for the caller.

if the caller presses the dtmf key * the playback is stopped and i
receive an channel_execute_complete playback for this file. this only
happens with the * - the other keys do nothing (as i whish).

i there somewhere a default setting for this, which i can turn off? i
looked through all conf-files, but could not find enything.

for my tests i use the default 1000 user as inbound. in the
ivr.conf.xml i found something with menu-top and digits *. i commented
this line out, but this didn't change anything.

thanks for your help.

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[Freeswitch-users] Listen to a file, while recording?

[Dennis]

hi,

i would like to be able to listen to conversations, while they are
ongoing. this should not happen over a phone. i would like to be able
to have a link or something in my admin-area, where i can click, if i
want to listen to a conversation.

i thought about to start a record with socket inbound on a specific
uuid and while the recording is done, i would like to play the file.
but i have the feeling, that this is not possible, while the file is
not finished.

is there a way to stream the audio over the web?


thanks for your help.

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Re: [Freeswitch-users] DTMF

[Brian West]

These aren't inserting 1003 as the caller_id_number are they?

/b

On Nov 18, 2008, at 5:19 AM, Baskar wrote:

>
>
>
>
>


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[Freeswitch-users] Unstable att_xfer

[x y]

Hy!

I have tested several times so far the att_xfer function of the freeswitch, and 
I've found it unstable. I'm
using a similar code to the example at the freeswitch wiki, and sometimes a 
call isnt transfered.
The att_xfer works fine in the cases as I figured out from the log, when the 
switch_ivr_bridge comes
back with a "ending bridge by request from read function" message. When it 
comes back with a 
"request from write function", the two channels dont get bridged together: they 
cant hear each other 
form the other side, and they dont even realize that the other hangs up.
I have pasted some logs about this a week ago into paste bin, but if you wish, 
I could paste new logs
with the newest fs.
Could it has something with silence, because I think there are more failures 
when I dont talk into
the phones?
Im using latest fs, and sip phones without built in transfer.

Cheers,
Viktor






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Re: [Freeswitch-users] Unstable att_xfer

[Brian West]

What device are you using?

/b

On Nov 18, 2008, at 8:03 AM, x y wrote:


Im using latest fs, and sip phones without built in transfer.


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Re: [Freeswitch-users] Unstable att_xfer

[x y]

Datavox Ip-300, and X-Lite softphone for testing.





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[Freeswitch-users] RFC 4028 - SIP Session Timers

[Iñaki Baz Castillo]

Hi, I've read that FS supports/implements Session Timers to monitorice
both legs of a call. How to enable it? I mean:

  alice --- FS  bob

- alice calls bob vía FS
- FS calls bob.
- bob answers (sends 200 OK).
- "bypass_media" mode, no RTP through FS.
- FS establishes a SIP dialog with alice and other one with bob.
- From this moment FS starts sending periodically in-dialog
INVITE/UPDATE to both legs in order to check if each SIP dialog is
still alive in both endpoints.
- In case alice crashes (looses dialog info), alice will reply "481
Call/Transaction doesn't exist" when the in-dialog INVITE/UPDATE
arrives from FS, so FS will understand that alice has ended the dialog
(or has crashed) and sends a BYE to bob.

Is it possible with FS? how to enable it?


-- 
Iñaki Baz Castillo
<[EMAIL PROTECTED]>
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Re: [Freeswitch-users] Unstable att_xfer

[Brian West]

Now I need to see the console logs with debug set.  "console loglevel  
debug" or press F8, Also document the exact scenario and every step  
you take, every key you press.


/b

On Nov 18, 2008, at 9:04 AM, x y wrote:


Datavox Ip-300


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Re: [Freeswitch-users] Javascript: stream, speak, stream - cepstral cut offs 2nd stream

[Birgit Arkesteijn]

Hi,

Thanks, Anthony, for your reply.
mSession.sleep(100);
does indeed do the trick!

Cheers, Birgit

On 17/11/08 19:51, Anthony Minessale wrote:
> you never want to msleep during a js running on a call
> you should use session.sleep(500);
> msleep blocks the whole thread and thus the audio.
> 
> 
> On Mon, Nov 17, 2008 at 1:05 PM, Birgit Arkesteijn <[EMAIL PROTECTED]>wrote:
> 
>> Hi,
>>
>> In javascript I doing the following:
>>
>> var consumer_name = "Birgit Arkesteijn";
>> var endpoint_url = 'sofia/gateway/westhawk/0662';
>> var mSession = new Session("{ignore_early_media=true,originate_timeout=8}"
>> + endpoint_url);
>> var ready = mSession.ready();
>> // log("merchant answered: " + ready);
>> if (ready == true)
>> {
>> mSession.streamFile("westhawk/lead_waiting.wav", dummy);
>> mSession.speak("cepstral", "Diane", "" +
>> consumer_name);
>> // msleep(500);
>> mSession.streamFile("westhawk/consumer_hungup.wav",
>> dummy);
>> exit();
>> }
>> function dummy(session, type, data, arg){}
>>
>>
>> No matter what I try, cepstral somehow cuts off the beginning two
>> seconds of the second streamFile().
>> Adding a msleep(500) in between 'speak' and 'streamFile' makes it only
>> worse.
>>
>> I'm running
>> FreeSWITCH Version 1.0.trunk (597:10325M) Started.
>> and
>> Cepstral_Diane_x86-64-linux_5.1.0/
>> on Linux Suse 10.0 x86_64
>>
>> Any ideas?
>>
>> Thanks, Birgit

-- 
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
-- Company no: 1769350
-- Registered Office:
-- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
-- tel.: +44 (0)161 237 0660
-- http://www.westhawk.co.uk>

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[Freeswitch-users] Javascript: record ringing of session

[Birgit Arkesteijn]

Hi,

In Javascript, I do the following:

var mSession = new 
Session("{ignore_early_media=true,originate_timeout=8}" + endpoint_url);
mSession.execute("record_session", recordfile);

This works very well, however, the client would like the rings to be 
recorded as well.

I've tried using various channel variables, but 'new Session' only 
returns after the call is answered.

I tried calling
apiExecute("record", );
before 'new Session', but that returns an error.

Is is possible to record the rings (in javascript)?
And if so, how would I do that?

Thanks, Birgit

-- 
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
-- Company no: 1769350
-- Registered Office:
-- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
-- tel.: +44 (0)161 237 0660
-- http://www.westhawk.co.uk>

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Re: [Freeswitch-users] Listen to a file, while recording?

[Anthony Minessale]

you can use the eavesdrop dialplan app from a new call to spy on an in
progress session
it takes the uuid of the channel you want to listen to as the arg.


On Tue, Nov 18, 2008 at 6:28 AM, Dennis <[EMAIL PROTECTED]> wrote:

> hi,
>
> i would like to be able to listen to conversations, while they are
> ongoing. this should not happen over a phone. i would like to be able
> to have a link or something in my admin-area, where i can click, if i
> want to listen to a conversation.
>
> i thought about to start a record with socket inbound on a specific
> uuid and while the recording is done, i would like to play the file.
> but i have the feeling, that this is not possible, while the file is
> not finished.
>
> is there a way to stream the audio over the web?
>
>
> thanks for your help.
>
> ___
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Re: [Freeswitch-users] Problems with DTMF and * on inbound leg

[Anthony Minessale]

set the variable playback_terminators=none before you execute playback.


On Tue, Nov 18, 2008 at 6:21 AM, Dennis <[EMAIL PROTECTED]> wrote:

> i am using socket outbound and if an inbound call comes in, i answer
> the call and play a soundfile for the caller.
>
> if the caller presses the dtmf key * the playback is stopped and i
> receive an channel_execute_complete playback for this file. this only
> happens with the * - the other keys do nothing (as i whish).
>
> i there somewhere a default setting for this, which i can turn off? i
> looked through all conf-files, but could not find enything.
>
> for my tests i use the default 1000 user as inbound. in the
> ivr.conf.xml i found something with menu-top and digits *. i commented
> this line out, but this didn't change anything.
>
> thanks for your help.
>
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Re: [Freeswitch-users] Unstable att_xfer

[x y]

Devices: IP-300 with 1007 registered
 IP-300 with 1009 registered
 X-Lite with 1011 registered

Situation where att_xfer success:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to 
execute my extension)
-1007 ringing, get the phone
-press *1 on 1007, then dial 1009 when the dialing wav is starting to play, 
while X-Lite plays hold music
-1009 (the other IP-300) ringing, get the phone
-1009 and 1007 bridged together, they can talk, then I hang up 1007
-1009 and 1011 bridged together, they can talk
-few secs later, I hang up 1009
-1011 hangs up too

Situation where att_xfer fails:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to 
execute my extension)

-1007 ringing, get the phone

-press *1 on 1007, then dial 1009 when the dialing wav is starting to play, 
while X-Lite plays hold music

-1009 (the other IP-300) ringing, get the phone

-1009 and 1007 bridged together, they can talk, then I hang up 1007 
different from 
here-

-1009 silent, but not on hold, nor on hung up, 1011 stops playing waiting 
music, being silent too

-few secs later, I hang up 1009

-1011 doesnt react, so I hang up 1011 too



When the att_xfer went failure, I didnt make any noise. When it went ok, I 
spoke into all phones.
I dont know if it counts.

Logst added to pastebin.

Cheers,
Viktor





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Re: [Freeswitch-users] SIP Invite IP fragmentation issue

[Anthony Minessale]

the only reliable answer is use TCP.

The RFC is daft in this matter.
They say when it's bigger than mtu to automatically use TCP instead.
And timeout for 10 seconds then fall back to UDP.

Its mandatory in SIP to support both TCP and UDP up to 64k per packet.
As you can see, since barely anything will do this right, your best bet is
to only use TCP when you have this kind of traffic.


On Tue, Nov 18, 2008 at 5:52 AM, Saurabh Aggarwal <
[EMAIL PROTECTED]> wrote:

>  Thanks, how do I "enable" this in freeswitch? Can this be done through the
> SIP configuration file?
>
> -Saurabh
>
>
> --
>
> Date: Tue, 18 Nov 2008 12:05:18 +0100
> From: [EMAIL PROTECTED]
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue
>
> The rfc also describes why:
>
> SIP provides a mechanism to represent common header field names in an
>abbreviated form.  This may be useful when messages would otherwise
>become too large to be carried on the transport available to it
>(exceeding the maximum transmission unit (MTU) when using UDP, for
>example).  These compact forms are defined in Section 20.  A compact
>form MAY be substituted for the longer form of a header field name at
>any time without changing the semantics of the message.  A header
>field name MAY appear in both long and short forms within the same
>message.  Implementations MUST accept both the long and short forms
>of each header name.
>
>
>
> On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo <[EMAIL PROTECTED]>wrote:
>
> 2008/11/18 Saurabh Aggarwal <[EMAIL PROTECTED]>:
> > enabling compact headers - what is that?
>
> SIP allows compact headers names for a few heades:
>
>  From = f
>  To = t
>  Via = v
>  ...
>
>
> --
> Iñaki Baz Castillo
> <[EMAIL PROTECTED]>
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Re: [Freeswitch-users] DTMF

[Cesar Cepeda]

Hi Baskar,

 

I'm not sure why you get the "-ERR no reply" answer since that answer is
never given by the function itself.

 

Nevertheless the uuid_send_dtmf function sends the dtmf_data TO the uuid you
enter, for what you want to achieve you need to "simulate" that the channel
that is eavesdropping presses the dtmf digits, instead of receiving them.

 

I've just uploaded a patch to jira MODAPP-168 which modifies uuid_send_dtmf
function to receive a third parameter, so that the new syntax is
"uuid_send_dtmf   [to|from|both]" that allows you to decide
if the dtmf data is sent "to" or "from" the uuid channel.

 

For example to achieve a three-party call after eavesdropping you would need
to call:

 

uuid_send_dtmf  3 from

 

Bye.

 

Cesar Cepeda.

 

De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Baskar
Enviado el: Monday, November 17, 2008 5:59 AM
Para: freeswitch-users@lists.freeswitch.org
Asunto: Re: [Freeswitch-users] DTMF

 

Hi,

i have tried it before itself first i pass one digit

api uuid_send_dtmf c08f77be-fbed-44c3-a2a7-8650d88b0e33 2


output:
Content-Type: api/response
Content-Length: 14

-ERR no reply

Then i passed all the values in the barging

api uuid_send_dtmf baf82956-111d-4cd8-9568-47010ac8bd20 2130*

output:
Content-Type: api/response
Content-Length: 14

-ERR no reply

Here i get only the beep sound for each values and error message as a output

Thanks for the reply and correct me where i am work

Thanks in advance.
-- 
Warm Regards,
N.Baskar

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Re: [Freeswitch-users] Unstable att_xfer

[Brian West]

You missed one thing. the console log with debug.

/b

On Nov 18, 2008, at 9:57 AM, x y wrote:


Devices: IP-300 with 1007 registered
 IP-300 with 1009 registered
 X-Lite with 1011 registered

Situation where att_xfer success:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix,  
assuring to execute my extension)

-1007 ringing, get the phone
-press *1 on 1007, then dial 1009 when the dialing wav is starting  
to play, while X-Lite plays hold music

-1009 (the other IP-300) ringing, get the phone
-1009 and 1007 bridged together, they can talk, then I hang up 1007
-1009 and 1011 bridged together, they can talk
-few secs later, I hang up 1009
-1011 hangs up too

Situation where att_xfer fails:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix,  
assuring to execute my extension)

-1007 ringing, get the phone
-press *1 on 1007, then dial 1009 when the dialing wav is starting  
to play, while X-Lite plays hold music

-1009 (the other IP-300) ringing, get the phone
-1009 and 1007 bridged together, they can talk, then I hang up 1007
different from  
here-
-1009 silent, but not on hold, nor on hung up, 1011 stops playing  
waiting music, being silent too

-few secs later, I hang up 1009
-1011 doesnt react, so I hang up 1011 too

When the att_xfer went failure, I didnt make any noise. When it went  
ok, I spoke into all phones.

I dont know if it counts.

Logst added to pastebin.

Cheers,
Viktor


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Re: [Freeswitch-users] Unstable att_xfer

[x y]

And btw, sorry for my english... thinked and tought, in one 
sentenceif only my english teacher would saw this :)





Hirdetés (x)  

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Re: [Freeswitch-users] Unstable att_xfer

[x y]

I tought u thinked of these:

http://pastebin.freeswitch.org/6179
http://pastebin.freeswitch.org/6178

If not, please correct me .

Viktor





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Re: [Freeswitch-users] Javascript: record ringing of session

[Michael S Collins]

Birgit,

I'm pretty sure that ringing is early media and you've got  
ignore_early_media set to true. If you can work without ignoring early  
media then you'll get rings. However, there's probably a reason you're  
ignoring early media so be sure to test thoroughly without ignoring  
early media just to make sure something else doesn't break. Actually I  
just noticed the call timeout you have. IIRC that does require you to  
ignore early media. Is there a way you can work without that call  
timeout? Let me know. Also, I might have an alternate solution but I  
can't give you more info until I get to my desk in an hour or so.

-MC

Sent from my iPhone

On Nov 18, 2008, at 7:40 AM, Birgit Arkesteijn <[EMAIL PROTECTED]>  
wrote:

> Hi,
>
> In Javascript, I do the following:
>
> var mSession = new
> Session("{ignore_early_media=true,originate_timeout=8}" +  
> endpoint_url);
> mSession.execute("record_session", recordfile);
>
> This works very well, however, the client would like the rings to be
> recorded as well.
>
> I've tried using various channel variables, but 'new Session' only
> returns after the call is answered.
>
> I tried calling
> apiExecute("record", );
> before 'new Session', but that returns an error.
>
> Is is possible to record the rings (in javascript)?
> And if so, how would I do that?
>
> Thanks, Birgit
>
> -- 
> -- Birgit Arkesteijn, [EMAIL PROTECTED],
> -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
> -- Company no: 1769350
> -- Registered Office:
> -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
> -- tel.: +44 (0)161 237 0660
> -- http://www.westhawk.co.uk>
>
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Re: [Freeswitch-users] Javascript: record ringing of session

[Birgit Arkesteijn]

Hi Michael,

Thanks for your reply.

I tried a variety of options:
var mSession = new Session(endpoint_url);
var mSession = new Session("{ignore_early_media=false}" +  endpoint_url);

both only return after the session is answered (originated?).

The customer wants
"the number of seconds we wait for each user to pick up"
configurable, hence I use "originate_timeout". I noticed (by try and 
error) that this is the timeout that does the trick. The other timeout 
variables behaved differently.

I'm currently trying to see how I can do recording of the rings. If that 
clashes with this neat "originate_timeout" variable, I'll have to see 
what my options are.

I did read the http://wiki.freeswitch.org/wiki/Channel_Variables page, 
but to be honest, even with descriptions I don't really understand the 
implications of using these variables, sorry.


Thanks, Birgit

On 18/11/08 17:03, Michael S Collins wrote:
> Birgit,
> 
> I'm pretty sure that ringing is early media and you've got  
> ignore_early_media set to true. If you can work without ignoring early  
> media then you'll get rings. However, there's probably a reason you're  
> ignoring early media so be sure to test thoroughly without ignoring  
> early media just to make sure something else doesn't break. Actually I  
> just noticed the call timeout you have. IIRC that does require you to  
> ignore early media. Is there a way you can work without that call  
> timeout? Let me know. Also, I might have an alternate solution but I  
> can't give you more info until I get to my desk in an hour or so.
> 
> -MC
> 
> Sent from my iPhone
> 
> On Nov 18, 2008, at 7:40 AM, Birgit Arkesteijn <[EMAIL PROTECTED]>  
> wrote:
> 
>> Hi,
>>
>> In Javascript, I do the following:
>>
>> var mSession = new
>> Session("{ignore_early_media=true,originate_timeout=8}" +  
>> endpoint_url);
>> mSession.execute("record_session", recordfile);
>>
>> This works very well, however, the client would like the rings to be
>> recorded as well.
>>
>> I've tried using various channel variables, but 'new Session' only
>> returns after the call is answered.
>>
>> I tried calling
>> apiExecute("record", );
>> before 'new Session', but that returns an error.
>>
>> Is is possible to record the rings (in javascript)?
>> And if so, how would I do that?
>>
>> Thanks, Birgit


-- 
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
-- Company no: 1769350
-- Registered Office:
-- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
-- tel.: +44 (0)161 237 0660
-- http://www.westhawk.co.uk>

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Re: [Freeswitch-users] Javascript: record ringing of session

[Michael S Collins]

Birgit,

I'm almost to my office. I will give you more info soon. I have not  
used js in this capacity so we will have to do some experimenting.

-MC

Sent from my iPhone

On Nov 18, 2008, at 9:51 AM, Birgit Arkesteijn <[EMAIL PROTECTED]>  
wrote:

> Hi Michael,
>
> Thanks for your reply.
>
> I tried a variety of options:
> var mSession = new Session(endpoint_url);
> var mSession = new Session("{ignore_early_media=false}" +   
> endpoint_url);
>
> both only return after the session is answered (originated?).
>
> The customer wants
> "the number of seconds we wait for each user to pick up"
> configurable, hence I use "originate_timeout". I noticed (by try and
> error) that this is the timeout that does the trick. The other timeout
> variables behaved differently.
>
> I'm currently trying to see how I can do recording of the rings. If  
> that
> clashes with this neat "originate_timeout" variable, I'll have to see
> what my options are.
>
> I did read the http://wiki.freeswitch.org/wiki/Channel_Variables page,
> but to be honest, even with descriptions I don't really understand the
> implications of using these variables, sorry.
>
>
> Thanks, Birgit
>
> On 18/11/08 17:03, Michael S Collins wrote:
>> Birgit,
>>
>> I'm pretty sure that ringing is early media and you've got
>> ignore_early_media set to true. If you can work without ignoring  
>> early
>> media then you'll get rings. However, there's probably a reason  
>> you're
>> ignoring early media so be sure to test thoroughly without ignoring
>> early media just to make sure something else doesn't break.  
>> Actually I
>> just noticed the call timeout you have. IIRC that does require you to
>> ignore early media. Is there a way you can work without that call
>> timeout? Let me know. Also, I might have an alternate solution but I
>> can't give you more info until I get to my desk in an hour or so.
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Nov 18, 2008, at 7:40 AM, Birgit Arkesteijn  
>> <[EMAIL PROTECTED]>
>> wrote:
>>
>>> Hi,
>>>
>>> In Javascript, I do the following:
>>>
>>> var mSession = new
>>> Session("{ignore_early_media=true,originate_timeout=8}" +
>>> endpoint_url);
>>> mSession.execute("record_session", recordfile);
>>>
>>> This works very well, however, the client would like the rings to be
>>> recorded as well.
>>>
>>> I've tried using various channel variables, but 'new Session' only
>>> returns after the call is answered.
>>>
>>> I tried calling
>>> apiExecute("record", );
>>> before 'new Session', but that returns an error.
>>>
>>> Is is possible to record the rings (in javascript)?
>>> And if so, how would I do that?
>>>
>>> Thanks, Birgit
>
>
> -- 
> -- Birgit Arkesteijn, [EMAIL PROTECTED],
> -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
> -- Company no: 1769350
> -- Registered Office:
> -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
> -- tel.: +44 (0)161 237 0660
> -- http://www.westhawk.co.uk>
>
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[Freeswitch-users] accountcode and user id not present in CDRs

[[EMAIL PROTECTED]]

I am using acls (cidr) to accept incoming calls from a gateway that
I do not want to register in my FS box.

I have this gateway configured in a xml file : 
freeswitch/conf/directory/default/gateway1.xml


  

  


  
  
  
  

  


I have the corresponding cidr in my ACL in acl.conf.xml.

I am able to make a call from that gateway to my FS but in my CDRs (both 
xml or cdr_csv)
the accountcode or user id is not present.

Any help on how to define an endpoint (originating) and use some 
attribute (like account_code or user id)
for billing purposes?


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[Freeswitch-users] Call answered by human or answerphone

[Birgit Arkesteijn]

Hi,

Our customer has different requirements for the situation where a person 
answered a call or a answerphone.

I found a solution on:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
This solution requires a person to press a DTMF.

However, I'd much rather do it the other way round, but didn't find 
anything when searching.

I heard mentioned 'wait for silence', but I don't see how waiting for 
silence will give me information on human VS machine detection.

Any ideas or solutions out there?
Our implementation is done in Javascript, so ideally the solution should 
be tied in.

Thanks, Birgit

-- 
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK
-- Company no: 1769350
-- Registered Office:
-- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK.
-- tel.: +44 (0)161 237 0660
-- http://www.westhawk.co.uk>

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Re: [Freeswitch-users] accountcode and user id not present in CDRs

[Brian West]

Add ${accuntcode} to the CDR template in cdr.conf.xml... the template  
can include any variables from the session.

/b

On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:

> Any help on how to define an endpoint (originating) and use some
> attribute (like account_code or user id)
> for billing purposes?


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Re: [Freeswitch-users] accountcode and user id not present in CDRs

[[EMAIL PROTECTED]]

The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the 
field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs included 
all of the session variables.




Brian West wrote:
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template  
can include any variables from the session.


/b

On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:

  

Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing purposes?




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Re: [Freeswitch-users] accountcode and user id not present in CDRs

[Brian West]

I'm going to guess that this is an inbound call to the user.  Which  
means the variables aren't set inbound to the user.

/b

On Nov 18, 2008, at 1:40 PM, [EMAIL PROTECTED] wrote:

> The ${accountcode} variable IS set in the cdr_csv.xml conf file yet  
> the field is empty after the call.
> Shouldn't it also show in the xml cdr? I thought the XML CDRs  
> included all of the session variables.


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Re: [Freeswitch-users] Relative timeout in Session.collectInput?

[mszlazak]

 Yup, I tried different settings for dft_min and dft_confirm before but did not 
understand what they meant. 
Back then I guessed that "dft" meant "discrete Fourier transform" not "default" 
and these variables were cut off frequencies of a filter but that did not quite 
make sense.
For instance, in the "yes/no" recognition part of the demo, one variable was 
set lower than the other but not in other places. 

Thanks for clearing up their meanings.

I've played with values for those variables more based on the output and it 
does help.

However, I'm and was getting recognition of "strange patterns" that all start 
with  in superscript like:


?
h
p?o??
h\|
?

What's going on here and what can I do about it?

Thanks.
Mark.


 


 

-Original Message-
From: Anthony Minessale <[EMAIL PROTECTED]>
To: freeswitch-users@lists.freeswitch.org
Sent: Mon, 17 Nov 2008 2:39 pm
Subject: Re: [Freeswitch-users] Relative timeout in Session.collectInput?









maybe its the detection of talk stop.
are you pasing between your toppings.
try saying them all very fast with no pause.
It may be the pause between utterances that is catching you.
the config has params.


threshold (default 400): higher the number louder you have to talk to be 
considered "talking" 
silence-hits (default 35): number of hits below threshold before detecting 
"stop talking"



On Mon, Nov 17, 2008 at 4:23 PM,  <[EMAIL PROTECTED]> wrote:




I'm suspecting that one of my pauses (or something else), as I'm saying my 
topping choices, is being interpreted as the end of my choices. 




I've tried adjusting the timeout argument value of collectInput but that 
doesn't seem to do anything. 



So, I'm left wondering what's causing my choices to be truncated and thus 
collectInput handles only a part of the toppings ordered. 



It doesn't seem like the speech recognizer is only recognizing one topping at a 
time as an event that's caught by collectInput since I have gotten recognition 
of more than one topping but usually not that many more.




Nickolay Shmyrev at CMU Sphinx Help Forum thought there maybe some timeout 
involved because passing silence to the recognizer wasn't a good idea and 
suggested a relative timeout. 




So what's being passed to the recognizer or is there something else going on? 



Mark.






 -Original Message-From: Anthony Minessale <[EMAIL PROTECTED]>



To: freeswitch-users@lists.freeswitch.org

Sent: Sun, 16 Nov 2008 8:40 am

Subject: Re: [Freeswitch-users] Relative timeout in Session.collectInput?

















The speech detector never stops working.? The collect input just pauses to wait 
for some input.

It can catch the event any time a file is playing or that collect input is 
called.







On Sat, Nov 15, 2008 at 1:19 PM,  <[EMAIL PROTECTED]> wrote:



Session.collectInput looks like it has a fixed timeout setting which blocks the 
channel until the timeout expires.







I've been informed that appears to be the case from a brief examination of how 
speech is collected.
SWITCH_DECLARE(switch_status_t) switch_ivr_collect_digits_callback?


seems to use a fixed timeout to cancel recording. ??







Is it possible to set-up a "relative" timeout which is based on the time from 
when the last valid sound was recognized as in choosing toppings from a list of 
20 toppings in the pizza demo? This situation involves an indeterminate number 
of valid sounds (toppings) and thus is unlike the dtmf example give in? 
http://wiki.freeswitch.org/wiki/Session_collectInput where a counter in the 
callback is also used to determine when to unblock the channel.







Mark.





? 










Instant access to the latest & most popular FREE games while you browse with 
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-- 

Anthony Minessale II



FreeSWITCH http://www.freeswitch.org/

ClueCon http://www.cluecon.com/




AIM: anthm

MSN:[EMAIL PROTECTED]

GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]


IRC: irc.freenode.net #freeswitch



FreeSWITCH Developer Conference

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Re: [Freeswitch-users] accountcode and user id not present in CDRs

[Anthony Minessale]

you have to manually set the var on the channel in your dialplan.


On Tue, Nov 18, 2008 at 1:40 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

>  The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the
> field is empty after the call.
> Shouldn't it also show in the xml cdr? I thought the XML CDRs included all
> of the session variables.
>
>
>
>
> Brian West wrote:
>
> Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
> can include any variables from the session.
>
> /b
>
> On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
>
>
>
>  Any help on how to define an endpoint (originating) and use some
> attribute (like account_code or user id)
> for billing purposes?
>
>
>  ___
> Freeswitch-users mailing [EMAIL 
> PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]>
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]>
iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
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Re: [Freeswitch-users] RFC 4028 - SIP Session Timers

[Iñaki Baz Castillo]

El Martes, 18 de Noviembre de 2008, Iñaki Baz Castillo escribió:
> Hi, I've read that FS supports/implements Session Timers to monitorice
> both legs of a call. How to enable it? I mean:
>
>   alice --- FS  bob
>
> - alice calls bob vía FS
> - FS calls bob.
> - bob answers (sends 200 OK).
> - "bypass_media" mode, no RTP through FS.
> - FS establishes a SIP dialog with alice and other one with bob.
> - From this moment FS starts sending periodically in-dialog
> INVITE/UPDATE to both legs in order to check if each SIP dialog is
> still alive in both endpoints.
> - In case alice crashes (looses dialog info), alice will reply "481
> Call/Transaction doesn't exist" when the in-dialog INVITE/UPDATE
> arrives from FS, so FS will understand that alice has ended the dialog
> (or has crashed) and sends a BYE to bob.
>
> Is it possible with FS? how to enable it?


I've found those options in Sofia profiles:

   
   

They seem to be related to SIP Session Timers (nothing related to RTP), am I 
right?

Thanks.



-- 
Iñaki Baz Castillo

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Re: [Freeswitch-users] RFC 4028 - SIP Session Timers

[Brian West]

yes but RTP timers are in there too.

/b

On Nov 18, 2008, at 5:26 PM, Iñaki Baz Castillo wrote:

>
> They seem to be related to SIP Session Timers (nothing related to  
> RTP), am I
> right?


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Re: [Freeswitch-users] RFC 4028 - SIP Session Timers

[Iñaki Baz Castillo]

El Miércoles, 19 de Noviembre de 2008, Brian West escribió:
> yes but RTP timers are in there too.

Well, but I expect that RTP timers parameters are the following:

   
   
   
   


While SIP Session Timers parameters are those:

   
   


Am I right? Thanks a lot.

-- 
Iñaki Baz Castillo

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[Freeswitch-users] Anthony is all heart ...

[Gonzalo Servat]

Hi All,

We all like to be thanked and when someone does something for me, I like to
show my gratitude towards them and I think what Anthony did for me today
deserves a public show of gratitude. It goes like this:

I have a single port FXO card in my home machine running FS and I needed to
modify the tone_detect app so that it would not incorrectly detect busy
tones during phone calls. If I call in and hang up on the remote end, FS
would detect the busy tone and hang up correctly. The problem was that
tone_detect would act as soon as it detected the requested tone (in my case,
busy tone) so I might be in the middle of a phone call it would sometimes
incorrectly detect a single busy tone and drop the call. I tried to ignore
this at first but it started happening more often so I spoke to Anthony
about the possibility of modifying tone_detect to have a "hit count" during
a time window, so that it would have to detect X number of busy tones before
hanging up the call.
As you can all see, this is probably a minor issue on Anthony's priority
list since there are many more important issues to fix or things to do on FS
than fix tone_detect. Even so, he spent a good 6 hours on this issue and
kept hacking at it until he got it fixed.

Anthony, once again, fixing tone_detect really helped me but what I
appreciate more than anything is the willingness to help and invest so much
of your time on this .. so thank you once again!
- Gonzalo (aka znoG)

PS: I will soon update the wiki page
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect   with
details on how to use the hit count
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[Freeswitch-users] i need help regarding freeswitch

[Faisal Maqsoodi]

Dear all,
   Hi
    I ve just started reading on freeswitch. >From where should i 
start? Basically, i am am a graduate electronic engineer. Can i be a successful 
developer of freeswitch? Its my first experience of any telephonic system.
Faisal rehman, Pakistan



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Re: [Freeswitch-users] i need help regarding freeswitch

[Wasim Baig]

On Wed, Nov 19, 2008 at 11:07 AM, Faisal Maqsoodi
<[EMAIL PROTECTED]>wrote:

Faisal:

Welcome to the wonderful word of open source telephony.

I ve just started reading on freeswitch. From where should i
> start?


http://wiki.freeswitch.org/wiki/Main_Page
http://wiki.freeswitch.org/wiki/Documentation

are two good places to start ...


> Basically, i am am a graduate electronic engineer. Can i be a successful
> developer of freeswitch?


Absolutely.


> Its my first experience of any telephonic system.


Be prepared to read a lot, and then ask questions.
When you get an answer,  do document it for others on the wiki.

-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as
you scope creep, so shall we reap ...
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Re: [Freeswitch-users] i need help regarding freeswitch

[Faisal Maqsoodi]

Thanks very much sir u r so nice n cooperative.
--- On Tue, 11/18/08, Wasim Baig <[EMAIL PROTECTED]> wrote:
From: Wasim Baig <[EMAIL PROTECTED]>
Subject: Re: [Freeswitch-users] i need help regarding freeswitch
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, November 18, 2008, 10:36 PM

On Wed, Nov 19, 2008 at 11:07 AM, Faisal Maqsoodi <[EMAIL PROTECTED]> wrote:

Faisal:



Welcome to the wonderful word of open source telephony.


 
    I ve just started reading on freeswitch. >From where should i 
start? 
http://wiki.freeswitch.org/wiki/Main_Page

http://wiki.freeswitch.org/wiki/Documentation

are two good places to start ...
 

Basically, i am am a graduate electronic engineer. Can i be a successful 
developer of freeswitch? 
Absolutely.
 

Its my first experience of any telephonic system.
Be prepared to read a lot, and then ask questions. 
When you get an answer,  do document it for others on the wiki.


-- 
wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as 
you scope creep, so shall we reap ... 

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