[Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Dennis
hi,

we are fighting with two flaws in fs and would be happy, if they could
be fixed. we are using socket outbound.

1.) hangup a call in ringing state:
this worked in one of the last fs versions, but suddenly does not work anymore.

let's say, we have an inbound call and do 3 originates to different
targets. all 3 targets are in ringing state. the target, which answers
first, will be bridged with the inbound call, the other two (still
ringing) targets should be hung up.

we do not want fs to hang up the other two originates automatically.
we want to hang up the other two originates by sending the hangups. we
set hangup_after_bridge=false and park_after_bridge=true.
we do sendmsg hangup uuid. but the originates are first hung up, when
they are answered. when they are in ringing state, the hangup will do
nothing (anymore).

as i said, it worked before, so i assume, that something has changed
in the latest trunks.



2.) sendmsg uuid *whatever* can cause to excute the command on the wrong uuid:

let's say, we have an inbound call and an outbound call - at least we
thing we have it ;-).
now we do for example sendmsg outbound_uuid hangup to hangup the
outbound call. but, if the uuid of the outbound call does not exist
(because there was a problem or something), the inbound will be hung
up instead.
the same happens with all sent messages to an uuid, which does not
exist. if we want to do a playback for the same outbound, the inbound
will hear it, if the outbound_uuid does not exist.

perhaps this is a feature, but i think that it would be nicer and more
reliable, if the sendmsg is only executed on the given uuid. if the
given uuid does not exist, nothing should happen or even nicer, an
event with an error should be sent to the socket.


thanks
dennis

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[Freeswitch-users] Two nice to have features in fs

2008-12-08 Thread Dennis
hi,

we are using socket outbound and found out, that there are two
(perhaps) small things, we would like to see in fs.

1.) if we try to playback a soundfile, which does not exist, we do not
get an error or something. in the cli of fs we can see the error, but
we do not get anything about it over the socket.

we get an execute event for the playback and directly after it an
execute-complete.
wouldn't it be nicer, if one could get a notice about an error, like:
execute-complete-error or a variable, which is set with an error, if
an error occours?


2.) the session heartbeat event is an absolutely great feature in fs.
but sometimes we would like to get the session heartbeat every 5
seconds. at the moment the allowed minimum is every 10 seconds. a
smaller setting, like every 5 seconds, will result in a session
heartbeat of every 60 seconds.

to help us out, we edit the switch_core_session.c at line 899 and
change the seconds  10 to seconds  5.

because one has to set enable_heartbeat_events=5 manually, i do not
think, that there is a risk, that others, who do not want the
heartbeat to come that often, will be negatively affected by this
change.

might it be possible, to do the same changes to the default code?


thanks
dennis

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Re: [Freeswitch-users] Read app ignores custom variable when called via socket interface

2008-12-08 Thread Jan Kubr
OK my bad. The variable is set (I can see it in the Freeswitch console
when I use the info app), but they are only not send to me via the
socket interface. I get the variable_* variables only in the
beginning (after calling connect), but not in the events. How do I
enable this?
Thanks,
Jan

On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
 That seems unlikely.
 You sure about that?

 The var param is in the middle of the data which is passed as 1 giant string
 to the same exact app execution code.
 I don't see how it could differentiate

 did you try executing the info app right after to see all the vars.

 I'm not saying i don't believe you but it seems fishy.



 On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr [EMAIL PROTECTED] wrote:

 Hi,
 I checked out the current trunk (rev 10641) and found out that the
 read app ignores the varname parameter, it always puts the result in
 the DTMF-Digit variable. I'm calling it via the socket interface:

 sendmsg
 call-command: execute
 execute-app-name: read
 execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 1 #
 event-lock:true


 In the XML dialplan it works fine:

 action application=read data=1 1
 sounds/en/us/callie/conference/8000/conf-pin.wav res 1 #/
 action application=phrase data=spell,${res}/


 I have been using the socket call above successfully in the 1.0.1 release.

 Any ideas? Thanks,

 Jan Kubr

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Re: [Freeswitch-users] conference configured to call automatically the attended does not work

2008-12-08 Thread Carole O.

I have found my mistake.

In the dialplan I have written action
application=conference_set_auto_outcall data=sofia/internal/.../ instead
of action application=conference_set_auto_outcall data=user/.../ 

Sorry for the post,
Carole


Carole O. wrote:
 
 Hello,
 
 I have got some problems for the configuration of a simple conference
 which should be established by calling an extension and automatically
 inviting 2 people.
 
 Actually, this is based on the default configuration of Freeswitch
 (extension 0911). I have changed it a little:
 
 extension name=conference_test
 condition field=destination_number expression=^0911$
 !--  These parameters effect the outcalls made once you join 
   -- 
   action application=set
 data=conference_auto_outcall_caller_id_name=telephoneX / 
   action application=set
 data=conference_auto_outcall_id_number=0911 / 
   action application=set data=conference_auto_outcall_timeout=60
 / 
   action application=set data=conference_auto_outcall_flags=none
 / 
  
 !--  list of devices called  -- 
   action application=conference_set_auto_outcall
 data=sofia/internal/[EMAIL PROTECTED] / 
   action application=conference_set_auto_outcall
 data=sofia/internal/[EMAIL PROTECTED] / 
 
   action application=conference data=[EMAIL PROTECTED] / 
 /condition
  /extension
 
 
 I have attached a file with the console errors. There are some errors (moh
 errors) but since these were also present for room conference and it did
 not prevent it for working, I guess this is not the fundamental reason for
 the previous problem.
 
 I have an additional question. I have installed freeswitch from
 opensuse.org, there is a simple one-click installation but I am not sure
 this was a good idea, it seems to be light isn't?
 
 Thanks for your help,
 Carole
  http://www.nabble.com/file/p20856465/error_console.txt error_console.txt 
 

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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Michael Jerris
Can you please file bugs on http://jira.freeswitch.org with full sip  
trace and FreeSWITCH debug output of these cases.


On Dec 8, 2008, at 4:28 AM, Dennis wrote:

 hi,

 we are fighting with two flaws in fs and would be happy, if they could
 be fixed. we are using socket outbound.

 1.) hangup a call in ringing state:
 this worked in one of the last fs versions, but suddenly does not  
 work anymore.

 let's say, we have an inbound call and do 3 originates to different
 targets. all 3 targets are in ringing state. the target, which answers
 first, will be bridged with the inbound call, the other two (still
 ringing) targets should be hung up.

 we do not want fs to hang up the other two originates automatically.
 we want to hang up the other two originates by sending the hangups. we
 set hangup_after_bridge=false and park_after_bridge=true.
 we do sendmsg hangup uuid. but the originates are first hung up, when
 they are answered. when they are in ringing state, the hangup will do
 nothing (anymore).

 as i said, it worked before, so i assume, that something has changed
 in the latest trunks.



 2.) sendmsg uuid *whatever* can cause to excute the command on the  
 wrong uuid:

 let's say, we have an inbound call and an outbound call - at least we
 thing we have it ;-).
 now we do for example sendmsg outbound_uuid hangup to hangup the
 outbound call. but, if the uuid of the outbound call does not exist
 (because there was a problem or something), the inbound will be hung
 up instead.
 the same happens with all sent messages to an uuid, which does not
 exist. if we want to do a playback for the same outbound, the inbound
 will hear it, if the outbound_uuid does not exist.

 perhaps this is a feature, but i think that it would be nicer and more
 reliable, if the sendmsg is only executed on the given uuid. if the
 given uuid does not exist, nothing should happen or even nicer, an
 event with an error should be sent to the socket.


 thanks
 dennis

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Re: [Freeswitch-users] Two nice to have features in fs

2008-12-08 Thread Michael Jerris

On Dec 8, 2008, at 4:47 AM, Dennis wrote:

 at the moment the allowed minimum is every 10 seconds. a
 smaller setting, like every 5 seconds, will result in a session
 heartbeat of every 60 seconds.

Huh?

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Re: [Freeswitch-users] Proto specific hangup cause issue

2008-12-08 Thread Anthony Minessale
sip_hangup_disposition will be set to recv_bye on the side that was hungup.


On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos [EMAIL 
PROTECTED]wrote:

  Any updates about the which side hanged up potential variable?

 Michael S Collins wrote:

 Makes sense. I will look into this.
 -MC


 On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED]
 wrote:

  I am sending this second email to request/suggest/enquire about something
 relevant :

 Wouldn't it be useful to know which end of a specific call leg send the
 protocol
 specific hangup cause? Otherwise it would be difficult to understand what
 really happened.



 Michael S Collins wrote:

 I will do some research on this and let you know what I find out.
 Question: are these internal calls or pstn or ?? Just curious about
 your environment.

 Thanks,
 MC



 On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos  [EMAIL 
 PROTECTED][EMAIL PROTECTED]
 wrote:



  The proto_specific_hangup_cause is missing on one of the two
 call legs. When the caller hangs up it is missing from the a-leg CDR.
 When the callee hangs up it is missing from the b-leg CDR. Is this
 nornal?

 And another question : what piece of info could inform me about who
 hanged up?


 --
 ---
 Apostolos Pantsiopoulos
 Kinetix Tele.com R  D
 email: [EMAIL PROTECTED]
 ---


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ClueCon http://www.cluecon.com/

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Re: [Freeswitch-users] Proto specific hangup cause issue

2008-12-08 Thread Apostolos Pantsiopoulos

Any updates about the which side hanged up potential variable?

Michael S Collins wrote:
Makes sense. I will look into this. 
-MC



On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


I am sending this second email to request/suggest/enquire about 
something relevant :


Wouldn't it be useful to know which end of a specific call leg send 
the protocol
specific hangup cause? Otherwise it would be difficult to understand 
what really happened.




Michael S Collins wrote:
I will do some research on this and let you know what I find out.  
Question: are these internal calls or pstn or ?? Just curious about  
your environment.


Thanks,
MC



On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  
wrote:


  

The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this  
nornal?


And another question : what piece of info could inform me about who
hanged up?


--
---
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Kinetix Tele.com R  D
email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
---


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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Anthony Minessale
#2 was because when you sendmsg with no uuid on an outbound socket it
defaults to the session who called you.
I changed to code to make a distinction between not supplying a uuid and
supplying an invalid uuid.

#1 seems hard to believe.  Please provide a console trace of the channel
*ignoring* the hangup command.


On Mon, Dec 8, 2008 at 3:28 AM, Dennis [EMAIL PROTECTED] wrote:

 hi,

 we are fighting with two flaws in fs and would be happy, if they could
 be fixed. we are using socket outbound.

 1.) hangup a call in ringing state:
 this worked in one of the last fs versions, but suddenly does not work
 anymore.

 let's say, we have an inbound call and do 3 originates to different
 targets. all 3 targets are in ringing state. the target, which answers
 first, will be bridged with the inbound call, the other two (still
 ringing) targets should be hung up.

 we do not want fs to hang up the other two originates automatically.
 we want to hang up the other two originates by sending the hangups. we
 set hangup_after_bridge=false and park_after_bridge=true.
 we do sendmsg hangup uuid. but the originates are first hung up, when
 they are answered. when they are in ringing state, the hangup will do
 nothing (anymore).

 as i said, it worked before, so i assume, that something has changed
 in the latest trunks.



 2.) sendmsg uuid *whatever* can cause to excute the command on the wrong
 uuid:

 let's say, we have an inbound call and an outbound call - at least we
 thing we have it ;-).
 now we do for example sendmsg outbound_uuid hangup to hangup the
 outbound call. but, if the uuid of the outbound call does not exist
 (because there was a problem or something), the inbound will be hung
 up instead.
 the same happens with all sent messages to an uuid, which does not
 exist. if we want to do a playback for the same outbound, the inbound
 will hear it, if the outbound_uuid does not exist.

 perhaps this is a feature, but i think that it would be nicer and more
 reliable, if the sendmsg is only executed on the given uuid. if the
 given uuid does not exist, nothing should happen or even nicer, an
 event with an error should be sent to the socket.


 thanks
 dennis

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-- 
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ClueCon http://www.cluecon.com/

AIM: anthm
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Re: [Freeswitch-users] Proto specific hangup cause issue

2008-12-08 Thread Apostolos Pantsiopoulos

Not necessarily. For instance I got a send_cancel when the
calling party hanged up before the other party could pick up.
Also, shouldn't something like that be protocol/technology
neutral?



Anthony Minessale wrote:
sip_hangup_disposition will be set to recv_bye on the side that was 
hungup.



On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos 
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:


Any updates about the which side hanged up potential variable?

Michael S Collins wrote:
Makes sense. I will look into this. 
-MC



On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:


I am sending this second email to request/suggest/enquire about
something relevant :

Wouldn't it be useful to know which end of a specific call leg
send the protocol
specific hangup cause? Otherwise it would be difficult to
understand what really happened.



Michael S Collins wrote:
I will do some research on this and let you know what I find out.  
Question: are these internal calls or pstn or ?? Just curious about  
your environment.


Thanks,
MC



On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos  mailto:[EMAIL PROTECTED][EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  
wrote:


  

The proto_specific_hangup_cause is missing on one of the two
call legs. When the caller hangs up it is missing from the a-leg CDR.
When the callee hangs up it is missing from the b-leg CDR. Is this  
nornal?


And another question : what piece of info could inform me about who
hanged up?


-- 
---

Apostolos Pantsiopoulos
Kinetix Tele.com R  D
email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
---


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---

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Re: [Freeswitch-users] rootkit?

2008-12-08 Thread Steven Ayre
A few files which you can choose to install to let the Freeswitch
developers access your machine remotely if you ask them to look at a
problem you're having with Freeswitch.

It's not a real rootkit - just the SSH public key for their private
key (so they don't need a password) and configuration files for bash +
emacs (so they're working in the environment they're used to). It
doesn't burrow into your OS, hide itself or any of the other insidious
things a real rootkit does, and once you delete the files they can no
longer access your machine.

It isn't installed unless you choose to do so.

-Steve



2008/12/7 Martin Joseph [EMAIL PROTECTED]:
 What is the rootkit item that appears to be added to the SVN of trunk?

 Thanks,
 Marty


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Re: [Freeswitch-users] Two nice to have features in fs

2008-12-08 Thread Dennis
 Huh?

src/switch_core_session.c vom line 899 to 901:

if (seconds  10) {
seconds = 60;
}

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Re: [Freeswitch-users] Two nice to have features in fs

2008-12-08 Thread Anthony Minessale
done

On Mon, Dec 8, 2008 at 9:18 AM, Dennis [EMAIL PROTECTED] wrote:

  Huh?

 src/switch_core_session.c vom line 899 to 901:

if (seconds  10) {
seconds = 60;
 }

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Re: [Freeswitch-users] Two nice to have features in fs

2008-12-08 Thread Anthony Minessale
both done

On Mon, Dec 8, 2008 at 9:30 AM, Anthony Minessale 
[EMAIL PROTECTED] wrote:

 done


 On Mon, Dec 8, 2008 at 9:18 AM, Dennis [EMAIL PROTECTED] wrote:

  Huh?

 src/switch_core_session.c vom line 899 to 901:

if (seconds  10) {
seconds = 60;
 }

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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Dennis
 #2 was because when you sendmsg with no uuid on an outbound socket it
 defaults to the session who called you.
 I changed to code to make a distinction between not supplying a uuid and
 supplying an invalid uuid.

anthony, thanks for the quick reaction!

we just tested you changes and it works the opposite way it should.

this means: when we do not send an uuid, we get an an error
(Reply-Text = -ERR invalid session id []). if we send a wrong/not
existing uuid, the command will be executed on the inbound uuid.




 #1 seems hard to believe.  Please provide a console trace of the channel
 *ignoring* the hangup command.

i know it is hard to believe, we didn't believe it either ;-)

have a look at http://pastebin.freeswitch.org/6367

what we simply do here: the inbound is coming in, then we do an
originate and hang up the inbound. then we directly send a hangup for
the outbound. the outbound will go on ringing.
then, when the ringing outbound is answered, we directly get the hangup.
fs gets the hangup and remembers it, but seems to wait till the answer
to execute this command.

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Re: [Freeswitch-users] Proto specific hangup cause issue

2008-12-08 Thread Anthony Minessale
it is protocol neutral, that's why it starts with sip_

the variable can be any of:

send_bye
recv_bye
send_cancel
send_refuse


using that value you can determine the information you asked.  I answered
your specific question which was:
determining which side hanged up.  Maybe you should beat around the bush
less with your requirements for your application you are expecting me to
support for you.

I already added 2 patches for you right.  Just be clear about what you want.



On Mon, Dec 8, 2008 at 8:13 AM, Apostolos Pantsiopoulos [EMAIL 
PROTECTED]wrote:

  Not necessarily. For instance I got a send_cancel when the
 calling party hanged up before the other party could pick up.
 Also, shouldn't something like that be protocol/technology
 neutral?



 Anthony Minessale wrote:

 sip_hangup_disposition will be set to recv_bye on the side that was hungup.


 On Mon, Dec 8, 2008 at 4:11 AM, Apostolos Pantsiopoulos [EMAIL 
 PROTECTED]wrote:

  Any updates about the which side hanged up potential variable?

 Michael S Collins wrote:

  Makes sense. I will look into this.
 -MC


 On Dec 5, 2008, at 8:17 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED]
 wrote:

  I am sending this second email to request/suggest/enquire about
 something relevant :

 Wouldn't it be useful to know which end of a specific call leg send the
 protocol
 specific hangup cause? Otherwise it would be difficult to understand what
 really happened.



 Michael S Collins wrote:

 I will do some research on this and let you know what I find out.
 Question: are these internal calls or pstn or ?? Just curious about
 your environment.

 Thanks,
 MC



 On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos  [EMAIL 
 PROTECTED][EMAIL PROTECTED]
 wrote:



  The proto_specific_hangup_cause is missing on one of the two
 call legs. When the caller hangs up it is missing from the a-leg CDR.
 When the callee hangs up it is missing from the b-leg CDR. Is this
 nornal?

 And another question : what piece of info could inform me about who
 hanged up?


 --
 ---
 Apostolos Pantsiopoulos
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Re: [Freeswitch-users] Two nice to have features in fs

2008-12-08 Thread Dennis
great, that works! thanks a lot!

just tested the changes according an error, when a file is missing.
thanks again!


2008/12/8 Anthony Minessale [EMAIL PROTECTED]:
 done

 On Mon, Dec 8, 2008 at 9:18 AM, Dennis [EMAIL PROTECTED] wrote:

  Huh?

 src/switch_core_session.c vom line 899 to 901:

if (seconds  10) {
seconds = 60;
}

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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Anthony Minessale
try the sendmsg issue again

are you doing the hangup with

api uuid_kill uuid



On Mon, Dec 8, 2008 at 9:47 AM, Dennis [EMAIL PROTECTED] wrote:

  #2 was because when you sendmsg with no uuid on an outbound socket it
  defaults to the session who called you.
  I changed to code to make a distinction between not supplying a uuid and
  supplying an invalid uuid.

 anthony, thanks for the quick reaction!

 we just tested you changes and it works the opposite way it should.

 this means: when we do not send an uuid, we get an an error
 (Reply-Text = -ERR invalid session id []). if we send a wrong/not
 existing uuid, the command will be executed on the inbound uuid.




  #1 seems hard to believe.  Please provide a console trace of the channel
  *ignoring* the hangup command.

 i know it is hard to believe, we didn't believe it either ;-)

 have a look at http://pastebin.freeswitch.org/6367

 what we simply do here: the inbound is coming in, then we do an
 originate and hang up the inbound. then we directly send a hangup for
 the outbound. the outbound will go on ringing.
 then, when the ringing outbound is answered, we directly get the hangup.
 fs gets the hangup and remembers it, but seems to wait till the answer
 to execute this command.

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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Dennis
i have to shift places. will be back in a few minutes and test.

no, we are using the simple sendmsg uuid hangup. as far as we
remember, we do not use api uuid_kill, because we do not get a hangup
event with this.


2008/12/8 Anthony Minessale [EMAIL PROTECTED]:
 try the sendmsg issue again

 are you doing the hangup with

 api uuid_kill uuid

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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Anthony Minessale
you would get a hangup event in either case.


On Mon, Dec 8, 2008 at 10:19 AM, Dennis [EMAIL PROTECTED] wrote:

 i have to shift places. will be back in a few minutes and test.

 no, we are using the simple sendmsg uuid hangup. as far as we
 remember, we do not use api uuid_kill, because we do not get a hangup
 event with this.


 2008/12/8 Anthony Minessale [EMAIL PROTECTED]:
  try the sendmsg issue again
 
  are you doing the hangup with
 
  api uuid_kill uuid

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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Dennis
 you would get a hangup event in either case.

yes, you are right. we just tested and saw that. the reason for
sendmsg hangup, was the sometimes useful event-lock.

it works with api uuid_kill as we wanted. but with sendmsg hangup it
still does not work. shouldn't sendmsg hangup work like uuid_kill
here? how useful could it be, to let it ring, when the hangup was
already sent and is immediately executed when the anser is sent?


#2 now works perfectly. thanks for the great support!

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[Freeswitch-users] Catching hangups

2008-12-08 Thread Joe Bain
Hi,

I'm writing an IVR in Lua and am having problems dealing with hangups
cleanly. Very often session:ready() reports true long after I have hung up
and the hangup hook function I have set doesn't get called either. It seems
to report that the session is active indefinitely in some cases where a loop
keeps trying to get some dtmf key presses. Is there any trick to using
session:ready() or the hangup hook that I might have missed?

On a slightly related point I can't seem to access the session properties,
e.g. session.caller_id_num has a value of nil. Any thoughts here?

Thanks in advance,

Joe Bain
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Re: [Freeswitch-users] Two major flaws: Could they be fixed?

2008-12-08 Thread Anthony Minessale
channels in originate were not checking for private events.
now they should but if send them commands to do crazy stuff like play a file
while they are
in the middle of originating there could be ill side effects (e.g. play file
before media was established etc which could cause the call to abort)

btw you can send

call-command: hangup
hangup-cause: normal_clearing

in place of
call-command: execute
execute-app-name: hangup
execute-app-arg: normal_clearing


On Mon, Dec 8, 2008 at 10:56 AM, Dennis [EMAIL PROTECTED] wrote:

  you would get a hangup event in either case.

 yes, you are right. we just tested and saw that. the reason for
 sendmsg hangup, was the sometimes useful event-lock.

 it works with api uuid_kill as we wanted. but with sendmsg hangup it
 still does not work. shouldn't sendmsg hangup work like uuid_kill
 here? how useful could it be, to let it ring, when the hangup was
 already sent and is immediately executed when the anser is sent?


 #2 now works perfectly. thanks for the great support!

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Re: [Freeswitch-users] Catching hangups

2008-12-08 Thread Michael Collins
Joe,

A few questions... what svn rev are you running? Which operating
system? Finally, is it possible for you to put your dialplan and Lua
script up at pastebin.freeswitch.org?

Thanks,
MC

On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain [EMAIL PROTECTED] wrote:
 Hi,

 I'm writing an IVR in Lua and am having problems dealing with hangups
 cleanly. Very often session:ready() reports true long after I have hung up
 and the hangup hook function I have set doesn't get called either. It seems
 to report that the session is active indefinitely in some cases where a loop
 keeps trying to get some dtmf key presses. Is there any trick to using
 session:ready() or the hangup hook that I might have missed?

 On a slightly related point I can't seem to access the session properties,
 e.g. session.caller_id_num has a value of nil. Any thoughts here?

 Thanks in advance,

 Joe Bain
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Re: [Freeswitch-users] Read app ignores custom variable when called via socket interface

2008-12-08 Thread Anthony Minessale
i added a patch to index the variables on the
SWITCH_EVENT_CHANNEL_EXECUTE_COMPLETE
if you want to update

otherwise you can use uuid_getvar to retrieve the variable


On Mon, Dec 8, 2008 at 4:48 AM, Jan Kubr [EMAIL PROTECTED] wrote:

 OK my bad. The variable is set (I can see it in the Freeswitch console
 when I use the info app), but they are only not send to me via the
 socket interface. I get the variable_* variables only in the
 beginning (after calling connect), but not in the events. How do I
 enable this?
 Thanks,
 Jan

 On Sun, Dec 7, 2008 at 8:44 PM, Anthony Minessale
 [EMAIL PROTECTED] wrote:
  That seems unlikely.
  You sure about that?
 
  The var param is in the middle of the data which is passed as 1 giant
 string
  to the same exact app execution code.
  I don't see how it could differentiate
 
  did you try executing the info app right after to see all the vars.
 
  I'm not saying i don't believe you but it seems fishy.
 
 
 
  On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr [EMAIL PROTECTED] wrote:
 
  Hi,
  I checked out the current trunk (rev 10641) and found out that the
  read app ignores the varname parameter, it always puts the result in
  the DTMF-Digit variable. I'm calling it via the socket interface:
 
  sendmsg
  call-command: execute
  execute-app-name: read
  execute-app-arg: 1 1 en/us/callie/conference/8000/conf-pin.wav res 1
 #
  event-lock:true
 
 
  In the XML dialplan it works fine:
 
  action application=read data=1 1
  sounds/en/us/callie/conference/8000/conf-pin.wav res 1 #/
  action application=phrase data=spell,${res}/
 
 
  I have been using the socket call above successfully in the 1.0.1
 release.
 
  Any ideas? Thanks,
 
  Jan Kubr
 
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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]
  pstn:213-799-1400
 
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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]
pstn:213-799-1400
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Re: [Freeswitch-users] key tone trigger event during call

2008-12-08 Thread Dan Le
From my understanding, I didn't think tone_detect detects DTMF since it's
dual frequencies, rather tone_detect detects single frequencies like fax
tones.

I would just run an IVR with a session.read or session.getDigits to collect
DTMF.

Dan

On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact [EMAIL PROTECTED] wrote:

 Same thing with version 10640 build.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Brian West
 Sent: Saturday, December 06, 2008 1:01 AM
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] key tone trigger event during call

 make current or install current svn on a different box.

 /b

 On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:

 
  Ideas?  Am I doing something stupid or is tone_detect not just right
  here?


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Re: [Freeswitch-users] key tone trigger event during call

2008-12-08 Thread Michael Jerris
Are you really trying to detect a tone, or are you trying to detect  
dtmf (could be delivered via rfc2833, info, etc) ?


Mike

On Dec 8, 2008, at 12:51 PM, Dan Le wrote:

From my understanding, I didn't think tone_detect detects DTMF since  
it's dual frequencies, rather tone_detect detects single frequencies  
like fax tones.


I would just run an IVR with a session.read or session.getDigits to  
collect DTMF.

Dan


On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact [EMAIL PROTECTED]  
wrote:

Same thing with version 10640 build.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian West
Sent: Saturday, December 06, 2008 1:01 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] key tone trigger event during call

make current or install current svn on a different box.

/b

On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:


 Ideas?  Am I doing something stupid or is tone_detect not just right
 here?



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[Freeswitch-users] No audio after transfer

2008-12-08 Thread John Rutherford
I'm trying to get an attended transfer work with freeSWITCH, but it's
not quite working.  I have Microsoft Speech Server on one side and
Televantage on the other.  

 

MSS is originating a call, which freeSWITCH is bridging to Televantage.
That calls connects just fine.  Then, MSS sends a re-INVITE to
Televantage to put the call on hold.  This works too.  Then, MSS
originates another call to freeSWITCH, which is again bridged to
Televantage.  This works fine too.  

 

Then, MSS sends a REFER to freeSWITCH to do the transfer.  The transfer
should be complete, but there is no audio between the two calls-just
silence.  I have looked at pcaps and the freeSWITCH logs, but I'm not
seeing anything obviously wrong.  

 

After the REFER, I can see audio for both calls going between freeSWITCH
and Televantage, so it seems that the only thing missing is freeSWITCH
routing the audio from one call to the other call and vice-versa.

 

 

Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this.  

 

Thanks!

John

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Re: [Freeswitch-users] No audio after transfer

2008-12-08 Thread Brian West

Are you on SVN trunk?  If not what rev?

/b

On Dec 8, 2008, at 3:36 PM, John Rutherford wrote:

Any help would be greatly appreciated.  I have a pcap and the  
freeSWITCH logs, and I can easily reproduce this.


Thanks!
John


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Re: [Freeswitch-users] No audio after transfer

2008-12-08 Thread John Rutherford
Sorry.  I forgot to mention that.  

 

I checked out the trunk last week.  I have revision 10597.  

 

John

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brian West
Sent: Monday, December 08, 2008 4:48 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] No audio after transfer

 

Are you on SVN trunk?  If not what rev?

 

/b

 

On Dec 8, 2008, at 3:36 PM, John Rutherford wrote:





Any help would be greatly appreciated.  I have a pcap and the freeSWITCH
logs, and I can easily reproduce this. 

 

Thanks!

John

 

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[Freeswitch-users] Jitter + Packet Loss

2008-12-08 Thread Jonathan Palley
I'm curious to start a discussion on being able to query a channel and get
statistics on the incoming jitter and packet loss (calculated from the RTP,
not RTCP).

Is this on the roadmap?  Is it hard to do?

Would be very useful for us indeed!

Thanks -
JP
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