[Freeswitch-users] Unexpected hangup of FIFO consumer

2008-09-26 Thread Juan Backson
Hi,

I found out that if fifo_music is set before fifo out wait, the consumer
would get hung up after 30 s.

action application=set
data=fifo_music=/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav/

action application=fifo data=myfifo out wait /


If I remove the line that sets fifo_music, consumer can stay in the channel
forever.

Does anyone know what may be causing this problem?

Thanks,
JB
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[Freeswitch-users] Bridge call directly to extension

2008-09-26 Thread Gopal krishnan
Hi,

  Is there a possible way as like in Asterisk where the agents will login in
queue, so that the established call will be directly transferred to the
extension. Is there any module for that?

   Any help would be appreciated. Thanks

-- 
Thank you  with regards,
Gopal,
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Re: [Freeswitch-users] Re gistering remote extensions.

2008-09-26 Thread henkoegema

I still haven't been able to solve my problem.

Still getting:
=
2008-09-26 11:02:59 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't
find user [EMAIL PROTECTED]
You must define a domain called '.ftpaccess.cc' in your directory and
add a user with the id=2010 attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2008-09-26 11:03:37 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't
find user [EMAIL PROTECTED]
You must define a domain called '.ftpaccess.cc' in your directory and
add a user with the id=2010 attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
...etc.etcetc 

Have no idea how to solve this. :rules:

Henk




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Re: [Freeswitch-users] Error loading ODBC

2008-09-26 Thread preetha Ayyappan
yes, I did ./configure --enable-core-odbc-support  make  make
install.When i gave make install during installation,i found the following
error:

2008-09-26 19:37:19 [CRIT] switch_loadable_module.c:767
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
object file: No such file or directory**

Thanks

On 9/25/08, Raymond Chandler [EMAIL PROTECTED] wrote:

 if you built from source, make sure you use the --enable-core-odbc-support
 option on your ./configure

 like so
 ./configure --enable-core-odbc-support  make  make install

 -Ray



 preetha Ayyappan wrote:

  You mean i have to uncomment the line  load
 module=mod_spidermonkey_odbc/.?
 yes.I have uncommented that line.eventhough the same error occurs

 On Thu, Sep 25, 2008 at 2:33 PM, Michael Jerris [EMAIL PROTECTED] wrote:


 On Sep 25, 2008, at 4:47 AM, preetha wrote:

 
  Hi,
  when i try to run a sample odbc code from freeswitch console like
  [EMAIL PROTECTED] jsrun odbc.js
 
  I found the following error:
  API CALL [jsrun(odbc.js)] output:
  OK
 
  [EMAIL PROTECTED] 2008-09-25 19:44:35 [ERR] mod_spidermonkey.c:
  3303
  js_api_use() Error loading ODBC
  2008-09-25 19:44:35 [ERR] odbc.js:11 mod_spidermonkey()
  ReferenceError:
  ODBC is not defined
 


 Make sure you have mod_spidermonkey_odbc loaded in your
 spidermonkey.conf.xml file

 Mike


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Re: [Freeswitch-users] Error loading ODBC

2008-09-26 Thread David Knell

Hi Preetha,

You've probably not compiled the module.  Have a look in 
/usr/src/freeswitch (or wherever your source is); you'll
find a modules.conf there.  Uncomment the line 
'langages/mod_spidermonkey_odbc', rebuild, reinstall and things

might get better.

--Dave
yes, I did ./configure --enable-core-odbc-support  make  make 
install.When i gave make install during installation,i found the 
following error:
 
2008-09-26 19:37:19 [CRIT] switch_loadable_module.c:767 
switch_loadable_module_load_file() Error Loading module 
/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open 
shared object file: No such file or directory**


Thanks
 
On 9/25/08, *Raymond Chandler* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


if you built from source, make sure you use the
--enable-core-odbc-support option on your ./configure

like so
./configure --enable-core-odbc-support  make  make install

-Ray



preetha Ayyappan wrote:

You mean i have to uncomment the line  load
module=mod_spidermonkey_odbc/.?
yes.I have uncommented that line.eventhough the same error occurs

On Thu, Sep 25, 2008 at 2:33 PM, Michael Jerris [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:


On Sep 25, 2008, at 4:47 AM, preetha wrote:


 Hi,
 when i try to run a sample odbc code from freeswitch console
like
 [EMAIL PROTECTED] jsrun odbc.js

 I found the following error:
 API CALL [jsrun(odbc.js)] output:
 OK

 [EMAIL PROTECTED] 2008-09-25 19:44:35 [ERR]
mod_spidermonkey.c:
 3303
 js_api_use() Error loading ODBC
 2008-09-25 19:44:35 [ERR] odbc.js:11 mod_spidermonkey()
 ReferenceError:
 ODBC is not defined


 
Make sure you have mod_spidermonkey_odbc loaded in your

spidermonkey.conf.xml file

Mike


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T: 020 8114 8901  F: 020 3002 7257  M: 001 415 630 3031
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Re: [Freeswitch-users] Cause: MANDATORY_IE_MISSING - No dial-string available

2008-09-26 Thread Peter P GMX
Thanks for the hint. With the following dial-string in the directory it 
works:

 param name=dial-string value=[EMAIL 
PROTECTED],transfer_fallback_extension=${dialed_user}}${sofia_contact([EMAIL 
PROTECTED])}/

Best regards
Peter





Michael Jerris schrieb:
 On Sep 25, 2008, at 4:04 PM, Peter P GMX wrote:

   
 Hello Michael,

 thanks for the hint, but how shall a dial-string param look like? I
 looked up the internet but could not find an example.
 Can you provide an example?

 Best regards
 Peter

 

 its just an originate string like you use with bridge app or the  
 originate FSAPI command.  In the default configs we have it just as  
 the default in the domain, but the same setting is good in the user:

 http://svn.freeswitch.org/svn/freeswitch/trunk/conf/directory/default.xml

 MIke

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[Freeswitch-users] Sendevent syntax

2008-09-26 Thread Jon Bruel
I have tried to get an understanding of the possibilities of sendevent,
and with the help from Brian West I had successes with a few, as shown
in the examples below: 
Sendevent  NOTIFY
Event-string: talk
content-length: 2
content-type: application/simple-message-summary
Profile: internal
User: 11000
Host: 10.3.1.11

Sendevent  SEND_MESSAGE
Content-length: 2
content-type: application/simple-message-summary
Profile: internal
User: 11000
Host: 10.3.1.11

Sendevent  MESSAGE_WAITING
MWI-Message-Account: [EMAIL PROTECTED]
MWI-Messages-Waiting: 3
MWI-Voice-Message: 2/1 (1/1)

I lack a general understanding of the syntax. I have tried to hangup a
session using CHANNEL_HANGUP where I made a qualified guess about the
headers, but it did not work (instead I used api uuid_kill uuid).
Could I have an overview of the syntax or a hint about which code I
could look at in order to get a more full understanding? /Jon




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[Freeswitch-users] storing csv file in mysql database

2008-09-26 Thread preetha Ayyappan
hi,
I am getting the call records saved in the folder cdr-csv.I need to save
this cdr details in mysql database.Is it possible to do?If yes please assist
me to do this.

Thanks
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Re: [Freeswitch-users] Bridge call directly to extension

2008-09-26 Thread Anthony Minessale
There is a call group demo in the default config.

XXX represents an imaginary number of your choice so you can have a large
number of group combos
(it's not limited to 3 digits)

80XXX remove yourself from group XXX
81XXX add yourself to group XXX
82XXX call everyone in group XXX at the same time
83XXX call everyone in group XXX one at a time until someone answers


On Fri, Sep 26, 2008 at 3:37 AM, Gopal krishnan [EMAIL PROTECTED] wrote:

 Hi,

   Is there a possible way as like in Asterisk where the agents will login
 in queue, so that the established call will be directly transferred to the
 extension. Is there any module for that?

Any help would be appreciated. Thanks

 --
 Thank you  with regards,
 Gopal,


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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] Sendevent syntax

2008-09-26 Thread Brian West
Jon,
Exact examples are listed on the jira you opened at 
http://jira.freeswitch.org/browse/SFSIP-92

/b

On Sep 26, 2008, at 6:59 AM, Jon Bruel wrote:

 I have tried to get an understanding of the possibilities of  
 sendevent,
 and with the help from Brian West I had successes with a few, as shown
 in the examples below:
 Sendevent  NOTIFY
 Event-string: talk
 content-length: 2
 content-type: application/simple-message-summary
 Profile: internal
 User: 11000
 Host: 10.3.1.11

 Sendevent  SEND_MESSAGE
 Content-length: 2
 content-type: application/simple-message-summary
 Profile: internal
 User: 11000
 Host: 10.3.1.11

 Sendevent  MESSAGE_WAITING
 MWI-Message-Account: [EMAIL PROTECTED]
 MWI-Messages-Waiting: 3
 MWI-Voice-Message: 2/1 (1/1)

 I lack a general understanding of the syntax. I have tried to hangup a
 session using CHANNEL_HANGUP where I made a qualified guess about the
 headers, but it did not work (instead I used api uuid_kill uuid).
 Could I have an overview of the syntax or a hint about which code I
 could look at in order to get a more full understanding? /Jon




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Re: [Freeswitch-users] Unexpected hangup of FIFO consumer

2008-09-26 Thread Brian West
What was the hangup cause?

/b

On Sep 26, 2008, at 2:19 AM, Juan Backson wrote:

 Hi,

 I found out that if fifo_music is set before fifo out wait, the  
 consumer would get hung up after 30 s.

 action application=set data=fifo_music=/usr/local/ 
 freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav/
 action application=fifo data=myfifo out wait /


 If I remove the line that sets fifo_music, consumer can stay in the  
 channel forever.

 Does anyone know what may be causing this problem?

 Thanks,
 JB


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Re: [Freeswitch-users] Sendevent syntax

2008-09-26 Thread Brian West
On this bit of info you don't send an event to hang a channel up...  
when you api uuid_kill the event is fired for all to see.

/b

On Sep 26, 2008, at 6:59 AM, Jon Bruel wrote:

 I lack a general understanding of the syntax. I have tried to hangup a
 session using CHANNEL_HANGUP where I made a qualified guess about the
 headers, but it did not work (instead I used api uuid_kill uuid).
 Could I have an overview of the syntax or a hint about which code I
 could look at in order to get a more full understanding? /Jon


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[Freeswitch-users] IAX outbound trunk and distortion

2008-09-26 Thread Alex Kinch
Hi,

Just setup an IAX trunk from FS to a SIP provider who runs Asterisk,  
but getting a truckload of distortion when I make calls. The  
iax.conf.xml settings are as per the default.

I've turned debugging on but just wondered where would be a good place  
to start trying to track down the problem? The codec appears to be  
PCMU/8000. Any suggestions welcome - this is the first time I've used  
IAX on FS, and have never had audio distortion issues with SIP trunks  
etc.

Thanks,
Alex

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Re: [Freeswitch-users] IAX outbound trunk and distortion

2008-09-26 Thread Brian West
What kind of load and how many channels?

/b

On Sep 26, 2008, at 1:13 PM, Alex Kinch wrote:

 Hi,

 Just setup an IAX trunk from FS to a SIP provider who runs Asterisk,
 but getting a truckload of distortion when I make calls. The
 iax.conf.xml settings are as per the default.

 I've turned debugging on but just wondered where would be a good place
 to start trying to track down the problem? The codec appears to be
 PCMU/8000. Any suggestions welcome - this is the first time I've used
 IAX on FS, and have never had audio distortion issues with SIP trunks
 etc.

 Thanks,
 Alex

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Re: [Freeswitch-users] IAX outbound trunk and distortion

2008-09-26 Thread [EMAIL PROTECTED]
No load as no other calls. Just one channel  active. Sip to the same provider 
has worked previously.

I recorded a test call on xlite if you want a copy of the wav?

Cheers,
Alex
-- I sent this from my 3 mobile --

-original message-
Subject: Re: [Freeswitch-users] IAX outbound trunk and distortion
From: Brian West [EMAIL PROTECTED]
Date: 26/09/2008 9:30 pm

What kind of load and how many channels?

/b

On Sep 26, 2008, at 1:13 PM, Alex Kinch wrote:

 Hi,

 Just setup an IAX trunk from FS to a SIP provider who runs Asterisk,
 but getting a truckload of distortion when I make calls. The
 iax.conf.xml settings are as per the default.

 I've turned debugging on but just wondered where would be a good place
 to start trying to track down the problem? The codec appears to be
 PCMU/8000. Any suggestions welcome - this is the first time I've used
 IAX on FS, and have never had audio distortion issues with SIP trunks
 etc.

 Thanks,
 Alex

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Re: [Freeswitch-users] IAX outbound trunk and distortion

2008-09-26 Thread Brian West
if the recorded wav exhibits the issue sure.

/b

On Sep 26, 2008, at 3:39 PM, [EMAIL PROTECTED] wrote:

 No load as no other calls. Just one channel  active. Sip to the same  
 provider has worked previously.

 I recorded a test call on xlite if you want a copy of the wav?

 Cheers,
 Alex
 -- I sent this from my 3 mobile --


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Re: [Freeswitch-users] IAX outbound trunk and distortion

2008-09-26 Thread [EMAIL PROTECTED]
Sure, will send it over when I get back. Also appear to have a dialplan issue 
where a call on sip matches but an iax trunk doesn't, but I'll put that on a 
separate email.

Alex
-- I sent this from my 3 mobile --

-original message-
Subject: Re: [Freeswitch-users] IAX outbound trunk and distortion
From: Brian West [EMAIL PROTECTED]
Date: 26/09/2008 10:06 pm

if the recorded wav exhibits the issue sure.

/b

On Sep 26, 2008, at 3:39 PM, [EMAIL PROTECTED] wrote:

 No load as no other calls. Just one channel  active. Sip to the same  
 provider has worked previously.

 I recorded a test call on xlite if you want a copy of the wav?

 Cheers,
 Alex
 -- I sent this from my 3 mobile --




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[Freeswitch-users] IVR References

2008-09-26 Thread jflowers

I'm having difficulty in trying to configure my own IVR (not modifying the
demo_ivr files).  I think I have most of it figured out but I don't
understand the structure and my friends grep and find aren't helping much. 
I can't seem to figure out why the macros in
/usr/local/freeswitch/conf/lang/en/demo/demo-ivr.xml are parsed but those in
/usr/local/freeswitch/conf/lang/en/mydir/my-ivr.xml are not.  I must be
missing something.

Any direction as to where this is specified will be greatly appreciated. 
Thanks.
-- 
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Re: [Freeswitch-users] Unexpected hangup of FIFO consumer

2008-09-26 Thread Brian West
This should give you a little bit of clue.  You have a network issue.   
How is your network topology setup?

/b

On Sep 26, 2008, at 9:30 PM, Juan Backson wrote:

 ACK Timeout


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Re: [Freeswitch-users] IVR References

2008-09-26 Thread Andrew Gilbert
No expert on this, but have you tried to update /usr/local/freeswitch/ 
conf/lang/en/en.xml to insure you include my-ivr.xml?

So in addition to this:
 X-PRE-PROCESS cmd=include data=demo/*.xml/

have this as well:

 X-PRE-PROCESS cmd=include data=my-ivr/*.xml/



On Sep 26, 2008, at 9:03 PM, jflowers wrote:


 I'm having difficulty in trying to configure my own IVR (not  
 modifying the
 demo_ivr files).  I think I have most of it figured out but I don't
 understand the structure and my friends grep and find aren't helping  
 much.
 I can't seem to figure out why the macros in
 /usr/local/freeswitch/conf/lang/en/demo/demo-ivr.xml are parsed but  
 those in
 /usr/local/freeswitch/conf/lang/en/mydir/my-ivr.xml are not.  I must  
 be
 missing something.

 Any direction as to where this is specified will be greatly  
 appreciated.
 Thanks.
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Re: [Freeswitch-users] storing csv file in mysql database

2008-09-26 Thread preetha Ayyappan
Thanks wasim.I have changed the line
my $dbh =
DBI-connect(DBI:mysql:database=freeswitchdb;host=localhost,freeswitch,1234)
in  /usr/src/freeswitch/scripts/contrib/wasim/cdrload.pl .then i called an
extension.Now the call details are recorded only in master.csv.I am not
getting the details updated in database.Anything i have to do further after
changing the above line in cdrload.pl ?
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