date:20081212

Hi,

I'm working on a simple dialer, and I have the following problem: the
audio file starts playing before the user answeres the phone (while it's
ringing). It only works when I introduce a delay, but that doesn't seem
right.

For instance in the asterisk context referred in the call files, I had:

exten = s,4,Answer
exten = s,n,Wait(2)
exten = s,n,Background(${SOUNDFILE})
And indeed it played a soundfile 2 seconds after the called person
picked up the phone

In FS I currently initiate calls like this:

session.waitForAnswer(1);

if (session.ready()) {
session.sleep(2000);
session.streamFile(/*...*/);
}

Is this right?


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] Freeswitch logging

Hi,

I see that mod_cdr is marked as being non-functional on the wiki. I'm working 
on a dialer and I need a way to log information about calls.

What module should I use?

Thanks,


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Freeswitch logging

2008-12-12 Thread Hadley Rich

On Saturday 13 December 2008 01:10:29 Alexandru Nedelcu wrote:
 Hi,

 I see that mod_cdr is marked as being non-functional on the wiki. I'm
 working on a dialer and I need a way to log information about calls.

 What module should I use?

This was answered on IRC and a note added to the mod_cdr wiki page.

hads
-- 
http://nicegear.co.nz
New Zealands Open Source Hardware Supplier

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] fifo.conf.xml usage

2008-12-12 Thread Jon Bruel

I'm happy to see that you can add consumers to queues using the fifo.conf.xml 
configuration file. I have made some tests and I hope it may lead to a more 
universal way of setting up queues for small organisations than the one I have 
described in the wiki, and which includes (too) many javascripts. I have some 
questions to clarify my understanding. Using the fifo.conf.xml, I find:
 - That the consumers continue to ring after the caller has abandoned the 
queue. Is there a way to avoid this?
Further:
 - Is there a way to control the caller_id_name/number presented to the 
consumer?
 - Is there a way to control the ringing tone in the consumers like the one 
which can be used in the dialplan?
 - Can the fifo.conf.xml refer to an ODBC connection in order to get the 
members from a database?
Finally, thanks for all the good work everybody in the FS community has put 
into FS, I truly believe in the possibilities of this product. Checking the 
hits on Google certainly indicates you moving into the right direction. /Jon

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Freeswitch streamFile when the user answers

2008-12-12 Thread Darren Schreiber

How are you originating calls? You probably need to add
{ignore_early_media=true}. This tells FreeSWITCH not to return from
origination when early media (progress/ringing) was received (I think
anyway)...

See http://wiki.freeswitch.org/wiki/Channel_Variables#ignore_early_media

There is a sample of this in use with the originate command here:
http://wiki.freeswitch.org/wiki/Mod_commands#originate (about halfway down)

Setting channel variables before doing the originate 

   originate {ignore_early_media=true}sofia/mydomain.com/18005551...@1.2.3.4
1551212



Since you are making a dialer, you may want to start the originations in the
background and move on to the next call while tweaking the timeout value for
originated calls. From the WIKI again:

You can originate a call in the background (asynchronously) and playback a
message with a 60 second timeout. 

   bgapi originate
{ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number
playback(message)

- Darren

 

-Original Message-
From: Alexandru Nedelcu [mailto:a...@sinapticode.ro] 
Sent: Friday, December 12, 2008 3:39 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch streamFile when the user answers

Hi,

I'm working on a simple dialer, and I have the following problem: the audio
file starts playing before the user answeres the phone (while it's ringing).
It only works when I introduce a delay, but that doesn't seem right.

For instance in the asterisk context referred in the call files, I had:

exten = s,4,Answer
exten = s,n,Wait(2)
exten = s,n,Background(${SOUNDFILE})
And indeed it played a soundfile 2 seconds after the called person picked up
the phone

In FS I currently initiate calls like this:

session.waitForAnswer(1);

if (session.ready()) {
session.sleep(2000);
session.streamFile(/*...*/);
}

Is this right?


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] conference_auto_outcall_announce

2008-12-12 Thread Carole O.


Hello,

First, I would like to apologize for a mistake I have made: by adding the
following line in the profile
 param name=enter-sound value=path/to/file.wav / 
the enter sound is played.
I am sorry for this. I did not hear it because in the case I have been
analyzing the members of the conference the caller automatically invites are
VoIP speakers which beep before playing anything and apparently miss the
enter sound. (both the beep and the enter-sound have about the same length).

I still have the following questions:
1- Is it possible to introduce a delay so that the enter sound is played
only after 2s?

2- I have noticed that if the caller of the conference talks or makes some
noises at the very beginning when he is entering the conference and the
enter sound is played, we can hear it through the VoIP speakers. Is there
any way to prevent from this? I would like to mute the caller during the
enter-sound and I would need this to be done statically, I mean in the xml
files, and not from the shell.

Thanks!!
Carole



Carole O. wrote:
 
 Hello,
 
 Actually, I have already tried it but nothing happens: the file is not
 played and there is no error. 
 There is still a difference: if I configure it as you said, I can not be
 listening anymore, there is simply nothing.
 
 Would you have an idea? I have checked the path and the syntax 1 million
 times so I do not think I make mistake there.
 
 Thanks,
 Carole
 
 
 
 Brian West-3 wrote:
 
 Don't have play: in there and it should be fine.  Also if you want the  
 absolute path you start it with /path/to/file.wav
 
 
 /b
 
 On Dec 11, 2008, at 7:13 AM, Carole O. wrote:
 
 [/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
 [System error : no such file or directory]
 
 
 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
 
 
 
 

-- 
View this message in context: 
http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Sounds for pending 1.0.2/Hardware

FreeSWITCHers,
I would like to thank everyone that donated.  Enough was raised to  
cover the sound order.  ;)

Thanks,
Brian West
FreeSWITCH.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue

2008-12-12 Thread Helmut Kuper

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello,

I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
server in a non-root environment.

We experienced a timer problem which led to this FS console error message:

[ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries


During anylizing this we found that q921 T203 is never reset when link
is in state Multiple Frame Mode Established and SABME frames are
received by FS. So it must timeout regardless if SABME frames are
received or not.
Additionally we found that the default T203 value (10 sec) was too short
for AVAYA (it has to be =19 sec)

To fix the problem we changed two things in q921.c:

Change T203 default value from 10 sec to 2 sec
Line 406: trunk-T203Timeout = 2;

Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
received SABME frame
Line 1996: Q921T203TimerReset(trunk, tei);

After recompiling FS the Error disapeared. Next week we will do some
calls over the link to make sure there are no other side effects.

Is it planned to make the q921 timeouts configurable in openzap.conf or
in openzap.conf.xml?

best regards
Helmut


PS: My openzap configs:

openzap.conf

[span wanpipe PRI_1]
trunk_type = E1
b-channel = 1:1-15
d-channel = 1:16
b-channel = 1:17-31




openzap.conf.xml

configuration name=openzap.conf description=OpenZAP Configuration
  settings
param name=debug value=7/
!--param name=hold-music value=$${moh_uri}/--
!--param name=enable-analog-option value=call-swap/--
!--param name=enable-analog-option value=3-way/--
  /settings
   pri_spans
 span name=PRI_1
   !-- Log Levels: none, alert, crit, err, warning, notice, info,
debug --
   param name=q921loglevel value=debug/
   param name=q931loglevel value=debug/
   param name=mode value=user/
   param name=dialect value=Oh this is not my dialect/
   param name=dialplan value=XML/
   param name=context value=default/
 /span
   /pri_spans
/configuration

Very interesting here is, that the dialect parameter doesn't seem to
have an effect on FS. I use that one above without any errors or warning
and I guess that was not intended.



-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)

iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
=geXp
-END PGP SIGNATURE-

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] LDAP Integration

2008-12-12 Thread Vinicius Kobashi





does anyone have a sample of the
config file for mod_xml_ldap? and know where to put it?

Vinicius Kobashi escreveu:

  
  i did it... i still had some
problems with sasl, but i managed to fix them.
  
now the module is up and running but i still dunno where to put the
mod_xml_ldap configuration file.
  
does anyone have a sample of the config file? and know where to put it?
  
Michael Collins escreveu:
  
Please confirm your svn rev - I believe this was fixed recently. Do
"make current" in your source directory.
-MC

On Thu, Dec 11, 2008 at 1:35 PM, Vinicius Kobashi
vkoba...@ydeasolutions.com.br wrote:
  

  ok ill try that

i found another module thats mod_xml_ldap

but when i try to load it, during compiling i get the 404 error
http://svn.freeswitch.org/downloads/libs/openldap-2.4.11.tgz file not
found
ill try to download it myself and then try to compile freeswitch again and
test

=D thankz for the fast answer

Hadley Rich escreveu:

On Friday 12 December 2008 09:16:56 Vinicius Kobashi wrote:


i found another module called mod_xml_curl and loaded it to freeswitch
too... but still it shows me the following error:

 2008-12-11 17:04:04 [WARNING] sofia_reg.c:1501 sofia_reg_parse_auth()
Can't find user [usern...@freeswitchserver.com] You must define a domain
called 'freeswitchserver.com' in your directory and add a user with the
id="username" attribute and you must configure your device to use the
proper domain in it's authentication credentials.

 does anyone got an idea?


Yes, you need to define a domain called 'freeswitchserver.com' in your
directory and add a user with the id="username" just like the error message
says.

The directory files are in conf/directory/

If you would like to read up on mod_xml_curl there is a detailed page on the
wiki;

http://wiki.freeswitch.org/wiki/Mod_xml_curl

hads


--


Vinicius Kobashi
Infra-Estrutura

Ydea Desenvolvimento de Software LTDA.
Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista
CEP.:04734-004 - So Paulo - SP
Tel.: 55-11-5523-0333
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  
  
  
  -- 
   
  
   Vinicius Kobashi 
Infra-Estrutura 
  
Ydea Desenvolvimento de Software LTDA. 
Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista 
CEP.:04734-004 - So Paulo - SP 
Tel.: 55-11-5523-0333  
  

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
  


-- 
 

 Vinicius Kobashi 
Infra-Estrutura 

Ydea Desenvolvimento de Software LTDA. 
Av. Adolfo Pinheiro, 2338 - Alto da Boa Vista 
CEP.:04734-004 - So Paulo - SP 
Tel.: 55-11-5523-0333  


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Bug in Q921.c? AVAYA-PBX issue

if you open a jira issue on it we can probably add your patch and/or the
config option.
the users-list is a tough place to manage TDM issues.


On Fri, Dec 12, 2008 at 9:01 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Hello,

 I try to establish a ISDN PRI (euroisdn/Q931) link between FS and an
 AVAYA PBX. We use Sangoma A101 and FS revision 10729M on a Ubuntu 8.04
 server in a non-root environment.

 We experienced a timer problem which led to this FS console error message:

 [ERR] Span:0 Q.921() Failed to establish Q.921 link in 3 retries


 During anylizing this we found that q921 T203 is never reset when link
 is in state Multiple Frame Mode Established and SABME frames are
 received by FS. So it must timeout regardless if SABME frames are
 received or not.
 Additionally we found that the default T203 value (10 sec) was too short
 for AVAYA (it has to be =19 sec)

 To fix the problem we changed two things in q921.c:

 Change T203 default value from 10 sec to 2 sec
 Line 406: trunk-T203Timeout = 2;

 Change Q921T203TimerStart to Q921T203TimerReset to reset T203 on each
 received SABME frame
 Line 1996: Q921T203TimerReset(trunk, tei);

 After recompiling FS the Error disapeared. Next week we will do some
 calls over the link to make sure there are no other side effects.

 Is it planned to make the q921 timeouts configurable in openzap.conf or
 in openzap.conf.xml?

 best regards
 Helmut


 PS: My openzap configs:

 openzap.conf

 [span wanpipe PRI_1]
 trunk_type = E1
 b-channel = 1:1-15
 d-channel = 1:16
 b-channel = 1:17-31




 openzap.conf.xml

 configuration name=openzap.conf description=OpenZAP Configuration
  settings
param name=debug value=7/
!--param name=hold-music value=$${moh_uri}/--
!--param name=enable-analog-option value=call-swap/--
!--param name=enable-analog-option value=3-way/--
  /settings
   pri_spans
 span name=PRI_1
   !-- Log Levels: none, alert, crit, err, warning, notice, info,
 debug --
   param name=q921loglevel value=debug/
   param name=q931loglevel value=debug/
   param name=mode value=user/
   param name=dialect value=Oh this is not my dialect/
   param name=dialplan value=XML/
   param name=context value=default/
 /span
   /pri_spans
 /configuration

 Very interesting here is, that the dialect parameter doesn't seem to
 have an effect on FS. I use that one above without any errors or warning
 and I guess that was not intended.



 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.9 (MingW32)

 iEYEARECAAYFAklCfB0ACgkQ4tZeNddg3dwZ2gCgovym/7R+5caEp1+fkupitN4p
 BWsAn3FGWcT1CUsVx4W2cQ7chKM5qixB
 =geXp
 -END PGP SIGNATURE-

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] fifo.conf.xml usage

the entries are standard originate strings so all of the {} variables apply.


On Fri, Dec 12, 2008 at 7:30 AM, Jon Bruel j...@consiglia.dk wrote:

  I'm happy to see that you can add consumers to queues using the
 fifo.conf.xml configuration file. I have made some tests and I hope it may
 lead to a more universal way of setting up queues for small organisations
 than the one I have described in the wiki, and which includes (too) many
 javascripts. I have some questions to clarify my understanding. Using the
 fifo.conf.xml, I find:

  - That the consumers continue to ring after the caller has abandoned the
 queue. Is there a way to avoid this?

 Further:

  - Is there a way to control the caller_id_name/number presented to the
 consumer?

  - Is there a way to control the ringing tone in the consumers like the one
 which can be used in the dialplan?

  - Can the fifo.conf.xml refer to an ODBC connection in order to get the
 members from a database?

 Finally, thanks for all the good work everybody in the FS community has put
 into FS, I truly believe in the possibilities of this product. Checking the
 hits on Google certainly indicates you moving into the right direction. /Jon



 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] conference_auto_outcall_announce

No,
there is currently no way.

On Fri, Dec 12, 2008 at 8:26 AM, Carole O. carole.oliv...@enst.fr wrote:

Hello,

First, I would like to apologize for a mistake I have made: by adding the
following line in the profile
param name=enter-sound value=path/to/file.wav /
the enter sound is played.
I am sorry for this. I did not hear it because in the case I have been
analyzing the members of the conference the caller automatically invites
are
VoIP speakers which beep before playing anything and apparently miss the
enter sound. (both the beep and the enter-sound have about the same
length).

I still have the following questions:
1- Is it possible to introduce a delay so that the enter sound is played
only after 2s?

2- I have noticed that if the caller of the conference talks or makes some
noises at the very beginning when he is entering the conference and the
enter sound is played, we can hear it through the VoIP speakers. Is there
any way to prevent from this? I would like to mute the caller during the
enter-sound and I would need this to be done statically, I mean in the xml
files, and not from the shell.

Thanks!!
Carole

Carole O. wrote:

Hello,

Actually, I have already tried it but nothing happens: the file is not
played and there is no error.
There is still a difference: if I configure it as you said, I can not be
listening anymore, there is simply nothing.

Would you have an idea? I have checked the path and the syntax 1 million
times so I do not think I make mistake there.

Thanks,
Carole

Brian West-3 wrote:

Don't have play: in there and it should be fine. Also if you want the
absolute path you start it with /path/to/file.wav

On Dec 11, 2008, at 7:13 AM, Carole O. wrote:

[/opt/freeswitch/sounds/en/us/callie/play:path_file_to_play/file.wav]
[System error : no such file or directory]

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
View this message in context:
http://www.nabble.com/conference_auto_outcall_announce-tp20955216p20976612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] call transfer question

2008-12-12 Thread jonathan augenstine

I have a call scenario that involves transferring the call and dropping out
of the SIP/RTP stream.  I need to accept the SIP call, play a prompt, and
retrieve a pin code.  After a database lookup, I need to transfer the call
to another FS server and drop out of the SIP path.  I have done this with
the RTP media stream previously.  I am not sure what I need to do to drop
out of the SIP path.  Is this possible on FS?

Jonathan
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] call transfer question

You can use deflect to accomplish this.. it will do a refer to the  
other FS box.

/b

On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:

 I have a call scenario that involves transferring the call and  
 dropping out of the SIP/RTP stream.  I need to accept the SIP call,  
 play a prompt, and retrieve a pin code.  After a database lookup, I  
 need to transfer the call to another FS server and drop out of the  
 SIP path.  I have done this with the RTP media stream previously.  I  
 am not sure what I need to do to drop out of the SIP path.  Is this  
 possible on FS?

 Jonathan


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] call transfer question

2008-12-12 Thread jonathan augenstine

Thank you, that is exactly what I need.

On Fri, Dec 12, 2008 at 9:14 AM, Brian West br...@freeswitch.org wrote:

 You can use deflect to accomplish this.. it will do a refer to the
 other FS box.

 /b

 On Dec 12, 2008, at 11:08 AM, jonathan augenstine wrote:

  I have a call scenario that involves transferring the call and
  dropping out of the SIP/RTP stream.  I need to accept the SIP call,
  play a prompt, and retrieve a pin code.  After a database lookup, I
  need to transfer the call to another FS server and drop out of the
  SIP path.  I have done this with the RTP media stream previously.  I
  am not sure what I need to do to drop out of the SIP path.  Is this
  possible on FS?
 
  Jonathan


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] CDR logs - adding a custom field

In Asterisk I was able to set a custom CDR field by doing something
like: 
   Set(CDR(userfield)=${SOMETHING})

I need to set a custom field in FreeSwitch, and preferably I want to
have control over its value from Javascript.

Can someone tell me how? :)

Thanks,

-- 
Alexandru Nedelcu
Software Developer, Sinapticode


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

Yes, I'm familiar with that since i invented that feature for Asterisk =D


In FreeSWITCH, All variables are already available from the cdr
just set regular channel variables.

for xml cdr they are all there right away
for csv cdr you can reference any channel variable in your template.




On Fri, Dec 12, 2008 at 12:37 PM, Alexandru Nedelcu a...@sinapticode.rowrote:

 In Asterisk I was able to set a custom CDR field by doing something
 like:
   Set(CDR(userfield)=${SOMETHING})

 I need to set a custom field in FreeSwitch, and preferably I want to
 have control over its value from Javascript.

 Can someone tell me how? :)

 Thanks,

 --
 Alexandru Nedelcu
 Software Developer, Sinapticode


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

Are you using CSV or XML? The reason I ask is because I personally use
XML and I find that having lots of information (even too much) is
better than not enough. The only drawback to XML that I find is that
you have to know how to parse it properly. :) The level of detail in
the XML CDRs is unmatched by any telephony system I've ever
encountered. I highly recommend it.

Also, check out this wiki page if you haven't already:
http://wiki.freeswitch.org/wiki/Mod_xml_cdr

-MC

On Fri, Dec 12, 2008 at 10:37 AM, Alexandru Nedelcu a...@sinapticode.ro wrote:
 In Asterisk I was able to set a custom CDR field by doing something
 like:
   Set(CDR(userfield)=${SOMETHING})

 I need to set a custom field in FreeSwitch, and preferably I want to
 have control over its value from Javascript.

 Can someone tell me how? :)

 Thanks,

 --
 Alexandru Nedelcu
 Software Developer, Sinapticode


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

Jason,

If I understand correctly software other than PA can lock up the sound
card so that PA doesn't see it. That might explain why PA reports
number of devices = 0. Could you check to see if possibly something
else has control of your sound card, perhaps ALSA? Turn off anything
that might use the sound card and then restart FS to see if PA can
then detect your device.

-MC

On Fri, Dec 12, 2008 at 1:38 AM, Jason White ja...@jasonjgw.net wrote:
 I am new to FreeSWITCH; hence this is the first of what will probably be a
 number of questions as I learn.

 I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.

 Whenever I try to load the portaudio module I get the following in the logs. I
 haven't changed anything in the default portaudio configuration that comes
 with FreeSWITCH.

 PortAudio version number = 1899
 PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
 Number of devices = 0
 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
 input
  device!
 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
 input
  device!
 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
 switch_loadable_module_l
 oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so

 Other software that uses portaudio is known to work. I would expect FreeSWITCH
 to detect my Alsa sound devices.

 Suggestions welcome.


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

On Fri, 2008-12-12 at 13:18 -0600, Anthony Minessale wrote:
 Yes, I'm familiar with that since i invented that feature for Asterisk
 =D
 
 
 In FreeSWITCH, All variables are already available from the cdr
 just set regular channel variables.
 
 for xml cdr they are all there right away
 for csv cdr you can reference any channel variable in your template.
 

Thank you Anthony,

In case someone wants to know how to set channel variables, there's a
link on the wiki here: http://wiki.freeswitch.org/wiki/Channel_Variables



___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

Thanks Michael,

I'm going to use XML, since I don't really know what variables I want.
Another problem with CSV is that many people parse them with regular
expressions and scripts break when you add a new column.


On Fri, 2008-12-12 at 11:50 -0800, Michael Collins wrote:
 Are you using CSV or XML? The reason I ask is because I personally use
 XML and I find that having lots of information (even too much) is
 better than not enough. The only drawback to XML that I find is that
 you have to know how to parse it properly. :) The level of detail in
 the XML CDRs is unmatched by any telephony system I've ever
 encountered. I highly recommend it.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Freeswitch logging

On Sat, 2008-12-13 at 01:26 +1300, Hadley Rich wrote:
 This was answered on IRC and a note added to the mod_cdr wiki page.

Thanks Hadley,

I'm a total newbie to FreeSwitch and voip in general, sorry for my
persistence :) I'll try writing an article about my setup this weekend.



___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

What I think would be neat is to have a perl script to parse the XML  
cdr and spit out a graphic of the call path... now that would be neat.
/b

On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:

 Thanks Michael,

 I'm going to use XML, since I don't really know what variables I want.
 Another problem with CSV is that many people parse them with regular
 expressions and scripts break when you add a new column.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu a...@sinapticode.ro wrote:
 Thanks Michael,

 I'm going to use XML, since I don't really know what variables I want.
 Another problem with CSV is that many people parse them with regular
 expressions and scripts break when you add a new column.


This is true. If you build a proper parser for your XML it will easily
be able to handle new channel variables.
-MC

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

2008-12-12 Thread Giovanni Maruzzelli

Hi there,

you have to use the default ALSA audio device to share it, and to
have it automatically format and rate converted.

the default ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).

You have to find out what device id it has under portaudio.

But in this specific case, no device at all was found.

So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA


Sincerely,

Giovanni Maruzzelli
=
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins m...@freeswitch.org wrote:
 Jason,

 If I understand correctly software other than PA can lock up the sound
 card so that PA doesn't see it. That might explain why PA reports
 number of devices = 0. Could you check to see if possibly something
 else has control of your sound card, perhaps ALSA? Turn off anything
 that might use the sound card and then restart FS to see if PA can
 then detect your device.

 -MC

 On Fri, Dec 12, 2008 at 1:38 AM, Jason White ja...@jasonjgw.net wrote:
 I am new to FreeSWITCH; hence this is the first of what will probably be a
 number of questions as I learn.

 I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.

 Whenever I try to load the portaudio module I get the following in the logs. 
 I
 haven't changed anything in the default portaudio configuration that comes
 with FreeSWITCH.

 PortAudio version number = 1899
 PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
 Number of devices = 0
 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
 input
  device!
 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
 input
  device!
 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
 switch_loadable_module_l
 oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so

 Other software that uses portaudio is known to work. I would expect 
 FreeSWITCH
 to detect my Alsa sound devices.

 Suggestions welcome.


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

2008-12-12 Thread Shelby Ramsey

Are there any good examples floating around of XML parsers for this to dump
to MySQL?

On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins m...@freeswitch.org wrote:

 On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu a...@sinapticode.ro
 wrote:
  Thanks Michael,
 
  I'm going to use XML, since I don't really know what variables I want.
  Another problem with CSV is that many people parse them with regular
  expressions and scripts break when you add a new column.
 

 This is true. If you build a proper parser for your XML it will easily
 be able to handle new channel variables.
 -MC

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Configuring FreeSwitch

On Thu, 2008-12-11 at 09:55 -0500, Raymond Chandler wrote:
 
 i think i answered all of this for you on irc yesterday
 

Yes you did, thanks for your help.
I'm a total newbie, but the good news is that I'm almost finished with
my setup. FS is great :)

 use the bridge dialplan app to dial by ip similar to the following:
 action application=bridge 
 data=sofia/${use_profile}/num...@ip.address/

I'm using originate initially. And I think I did something stupid. Is
there anything wrong with the following code ...

var new_session = new Session();
new_session.originate(session, URL);
bridge(session, new_session);

 http://wiki.freeswitch.org/wiki/Sofia#Syntax might also help you out
 a 
 little

It worked great. Thanks.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

2008-12-12 Thread Giovanni Maruzzelli

Sorry, the previous one was sent by mistake.

This one is complete:


Hi there,

you have to use the default ALSA audio device to share it, and to
have it automatically format and rate converted.

the default ALSA device is not the default portaudio device (not in
the portaudio version used currently by FS).

You have to find out what device id it has under portaudio.

But in this specific case, no device at all was found.

So, maybe portaudio was not commpiled with ALSA support (do you have
the ALSA development library installed?).

Also, after recompiling portaudio and mod_portaudio, you can launch FS
giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will
use the plughw devices (that are automatically converted to the
desired rate/format) and not the raw devices.

Giovanni Maruzzelli
=
Company : Celliax
Website: www.celliax.org
Address : via Pierlombardo 9, 20135 Milano
Country/Territory : Italy
Business Email: gmaruzz at celliax dot org
Cell : 39-347-2665618
Fax : 39-02-87390039




On Fri, Dec 12, 2008 at 9:25 PM, Giovanni Maruzzelli
gmar...@celliax.org wrote:
 Hi there,

 you have to use the default ALSA audio device to share it, and to
 have it automatically format and rate converted.

 the default ALSA device is not the default portaudio device (not in
 the portaudio version used currently by FS).

 You have to find out what device id it has under portaudio.

 But in this specific case, no device at all was found.

 So, maybe portaudio was not commpiled with ALSA support (do you have the ALSA


 Sincerely,

 Giovanni Maruzzelli
 =
 Company : Celliax
 Website: www.celliax.org
 Address : via Pierlombardo 9, 20135 Milano
 Country/Territory : Italy
 Business Email: gmaruzz at celliax dot org
 Cell : 39-347-2665618
 Fax : 39-02-87390039




 On Fri, Dec 12, 2008 at 8:58 PM, Michael Collins m...@freeswitch.org wrote:
 Jason,

 If I understand correctly software other than PA can lock up the sound
 card so that PA doesn't see it. That might explain why PA reports
 number of devices = 0. Could you check to see if possibly something
 else has control of your sound card, perhaps ALSA? Turn off anything
 that might use the sound card and then restart FS to see if PA can
 then detect your device.

 -MC

 On Fri, Dec 12, 2008 at 1:38 AM, Jason White ja...@jasonjgw.net wrote:
 I am new to FreeSWITCH; hence this is the first of what will probably be a
 number of questions as I learn.

 I've compiled the latest code from svn trunk under Debian Sid (Linux kernel
 2.6.27, x86_64 architecture), with the portaudio19-dev package installed.

 Whenever I try to load the portaudio module I get the following in the 
 logs. I
 haven't changed anything in the default portaudio configuration that comes
 with FreeSWITCH.

 PortAudio version number = 1899
 PortAudio version text = 'PortAudio V19-devel (built Dec 12 2008)'
 Number of devices = 0
 2008-12-12 20:17:51 [ERR] mod_portaudio.c:883 load_config() Cannot find an
 input
  device!
 2008-12-12 20:17:51 [ERR] mod_portaudio.c:893 load_config() Cannot find an
 input
  device!
 2008-12-12 20:17:51 [CRIT] switch_loadable_module.c:839
 switch_loadable_module_l
 oad_file() Error Loading module /opt/freeswitch/mod/mod_portaudio.so

 Other software that uses portaudio is known to work. I would expect 
 FreeSWITCH
 to detect my Alsa sound devices.

 Suggestions welcome.


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org



___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

On Fri, Dec 12, 2008 at 12:21 PM, Brian West br...@freeswitch.org wrote:
 What I think would be neat is to have a perl script to parse the XML
 cdr and spit out a graphic of the call path... now that would be neat.
 /b

I think that is a great idea. I was kicking that around as an add-on
feature to a simple CDR database. For example, when browsing the db
for calls, you could click a link that says view call path and it
would print a nice purty graph/chart of the call flow. I'll put that
on my rainy-day list...

-MC


 On Dec 12, 2008, at 2:14 PM, Alexandru Nedelcu wrote:

 Thanks Michael,

 I'm going to use XML, since I don't really know what variables I want.
 Another problem with CSV is that many people parse them with regular
 expressions and scripts break when you add a new column.


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] CDR logs - adding a custom field

I don't know about good examples. I just hacked together a perl
script to extract the very specific elements for my application. If
anyone out there has a sample XML-to-db parser that would be very
welcomed...
-MC

On Fri, Dec 12, 2008 at 12:28 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
 Are there any good examples floating around of XML parsers for this to dump
 to MySQL?

 On Fri, Dec 12, 2008 at 2:22 PM, Michael Collins m...@freeswitch.org wrote:

 On Fri, Dec 12, 2008 at 12:14 PM, Alexandru Nedelcu a...@sinapticode.ro
 wrote:
  Thanks Michael,
 
  I'm going to use XML, since I don't really know what variables I want.
  Another problem with CSV is that many people parse them with regular
  expressions and scripts break when you add a new column.
 

 This is true. If you build a proper parser for your XML it will easily
 be able to handle new channel variables.
 -MC

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org



___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] Cepstral SDK

2008-12-12 Thread Pedro .

Hi,

I'm trying to integrate Cepstral TTS I read in the wiki that I need
Ceptral's SDK to compile the mod_ceptral, can somebody tell me where can I
get the trial version of this SDK?.

Thanks.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Cepstral SDK

If you're on linux you need to go download and install any voice.  If  
you're on windows I have to forward your request to Cepstral to get  
the SDK for windows.

/b

On Dec 12, 2008, at 3:47 PM, Pedro . wrote:

 Hi,

 I'm trying to integrate Cepstral TTS I read in the wiki that I need  
 Ceptral's SDK to compile the mod_ceptral, can somebody tell me where  
 can I get the trial version of this SDK?.

 Thanks.

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

On Fri, Dec 12, 2008 at 09:30:16PM +0100, Giovanni Maruzzelli wrote:
 
 But in this specific case, no device at all was found.
 
 So, maybe portaudio was not commpiled with ALSA support (do you have
 the ALSA development library installed?).

Yes, and in any case the version of PortAudio which is installed came from the
Debian package.

Does FreeSWITCH support PortAudio 19? If not, maybe there are API differences.
 
 Also, after recompiling portaudio and mod_portaudio, you can launch FS
 giving it the PA_ALSA_PLUGHW=1 environment variable, so portaudio will
 use the plughw devices (that are automatically converted to the
 desired rate/format) and not the raw devices.

I'll try that.

To answer another question that arose in this thread, I have no other software
currently using the audio devices. Alsa is known to work, as is other software
that accesses the Alsa devices with PortAudio.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

[Freeswitch-users] schedule a DTMF tone into bridge

2008-12-12 Thread Frank @ Impact

Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] schedule a DTMF tone into bridge


sched_api  (hint uuid_send_dtmf)

API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [...@]time group_name command_string


/b

On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:

Is there a way to schedule a certain DTMF tone to be played into a  
bridge (both a and b legs) after a scheduled number of seconds into  
the call?

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

This is interesting...

I wrote the following test.


#includestdio.h
#include portaudio.h
int main(int argc, char **argv) 
{
  Pa_Initialize();
  printf(Number of devices: %d\n,Pa_GetDeviceCount());
  Pa_Terminate();
}

then I compiled and executed it:
gcc -o pa_test pa_test.c -lportaudio
./portaudio

Number of devices: 10

From what I can see, the code in mod_portaudio.c is using exactly the same API
call, but it seems to be returning 0 in that case.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

With apologies for the noise on the list, I just realized that FreeSWITCH is
building its own version of PortAudio.

I can confirm that Alsa is being detected and support for it included.

So, there must be some difference between the version of PortAudio that comes
with FreeSWITCH, and the version installed in my /usr/lib, such that the
FreeSWITCH version fails to detect my Alsa devices.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] missing 3 seconds of audio on bridge calls

2008-12-12 Thread Angel Carpintero

Thanks again Anthony ! 

You fixed the issue with DTMF i had reported :

http://jira.freeswitch.org/browse/FSCORE-251



Chris Danielson added to Wiki a nice page collecting these issues with
Sonus :

http://wiki.freeswitch.org/wiki/RTP_Issues


Cheers,

El mié, 10-12-2008 a las 03:10 +0100, Angel Carpintero escribió:
 Thanks Anthony , you did a great work ! this is fixed in svn r10691.
 
 Some notes for people using Sonus and L3 as was my case :
 
 in var.xml in some scenario you may need :
 
 X-PRE-PROCESS cmd=set data=send_silence_when_idle=400/
 
 in sip_profiles/internal.xml :
 
 param name=rtp-rewrite-timestamps value=true/
 
 might help for some people with rtp issues :
 
 param name=rtp-timer-name value=none/
 
 If you have issues with DTMF and timestamps add also :
 
 param name=pass-rfc2833 value=true/
 
 I've a little issues with DTMF from VOIP , i i'll figure out can could
 be the issue , from PSTN all works like a charm :)
 
 Cheers,
 
 El jue, 04-12-2008 a las 09:34 -0600, Anthony Minessale escribió:
  most likely it's because during the time you are dong artificial
  ringback the other side is not doing RTP right.
  
  When the call is answered we flush the rtp buffer and your missing
  audio is probably flushed with it.
  so you can choose to have a 3 second delay or erase the 3 seconds as
  it does now.
  
  This is a known problem with sonus which has been proven to build up
  an audio delay during the time
  you are waiting for the call to answer.  I'm sure you prefer the way
  it is to a large audio delay.
  
  
  
  On Wed, Dec 3, 2008 at 10:46 PM, Angel Carpintero a...@telefonica.net
  wrote:
  No TDM , all is SIP :
  
  
  PSTN --- Sip Proxy_A -- FS ( brigde )
  ringback/transfer_ringback
  - Sip Proxy_B -- PSTN
  
  
  In logfile i think you can get some details about Media
  Gateways
  ( Sonus ) PSTN inbound / outbound is provided by Level3.
  
  I can get a capture of a call if you want, in capture the
  audio is not
  missing, issue with :
  
  - rtp buffer ?
  - Sonus ?
  
  Let me know anything you need so i can provide a log or create
  a new
  scenario.
  
  
  Thanks,
  
  El mié, 03-12-2008 a las 22:12 -0600, Anthony Minessale
  escribió:
  
   what does PSTN represent?
  
   I know what the PSTN is but how are you reaching it?
   is it TDM, SIP etc... what gateway type other details.
  
  
   On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero
  a...@telefonica.net
   wrote:
   Hi guys,
  
I've a strange issue with FS , version svn
  -r10584 ,
   when FS bridges a call first 3 seconds of audio are
  missing ,
   looks that
   only happens on PSTN calls and using ringback or
   transfer_ringback. This
   only happens in calls from PSTN , not from VOIP.
  Some
   scenarios i tried
   to isolate this issue :
  
  
   - Issue
  
   PSTN -- FS ( brigde ) ringback/transfer_ringback -
  PSTN
  
   - Good setting bypass_media before run bridge but i
  need rtp
   in FS path
  
   PSTN -- FS ( brigde ) ringback/transfer_ringback -
  PSTN
  
   - Good
  
   PSTN -- FS ( brigde ) WITHOUT
  ringback/transfer_ringback -
   PSTN
  
   - Always good
  
   VOIP -- FS ( brigde ) - PSTN
  
  
   Dialplan has nothing wrong ( i guess ):
  
   extension name=Transfers
  condition field=destination_number
   expression=^1??XX$
action application=answer/
action application=speak data=cepstral|
  Allison-8kHz|
   blah/
action application=set
   data=hangup_after_bridge=false/
action application=set
  data=playback_terminators=#/
action application=set data=ringback=
  $${us-ring}/
action application=set
  data=transfer_ringback=
   $${hold_music}/
action application=set
  data=effective_caller_id_name=
   ${caller_id_name}/
action application=set
   data=effective_caller_id_number=
   ${caller_id_number}/
action

Re: [Freeswitch-users] Error loading portaudio module

The problem is now solved.

It turned out to be permissions: the freeswitch user wasn't added to the audio
group in /etc/group, hence didn't have permission to interrogate the audio
devices.

Perhaps a future version of the Debian package could address this, or at least
it should be noted somewhere.


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] Error loading portaudio module

Jason,

Thanks for troubleshooting this! At the very least I will add a note
to the PA section of the wiki.
-MC

On Fri, Dec 12, 2008 at 6:01 PM, Jason White ja...@jasonjgw.net wrote:
 The problem is now solved.

 It turned out to be permissions: the freeswitch user wasn't added to the audio
 group in /etc/group, hence didn't have permission to interrogate the audio
 devices.

 Perhaps a future version of the Debian package could address this, or at least
 it should be noted somewhere.


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] schedule a DTMF tone into bridge

2008-12-12 Thread Frank @ Impact

Not much written in the wiki on this.  Also searched the list and not
much on either sched_api or uuid_send_dtmf.
So from an xml dialplan, can sched_api as an application?
Is there any way to have the time offset reference the point at which
the call started ?  ie. When the called party answers?
 
Ultimately, I was trying to insert some xml into my dial plan that would
play a dtmf tone 10 seconds after the called party picked up the phone.
But from the little that has been written so far that I can find, it is
not clear to me how to piece this together.  Am I being dense and
missing anything that has already been written?
 
/f
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West


sched_api  (hint uuid_send_dtmf)
 
API CALL [sched_api()] output:
-ERR Invalid syntax. USAGE: [...@]time group_name command_string
 
 
/b
 
On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:



Is there a way to schedule a certain DTMF tone to be played into a
bridge (both a and b legs) after a scheduled number of seconds into the
call?
___
 
 
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] schedule a DTMF tone into bridge

Frank,

I'm sure this is possible. Please give me a little bit to look into
this. I'm going to see if I can lab it up and give you a sample
dialplan. Also, thanks for the heads up on the wiki not having this
information. I will put that on my not-so-short wiki todo list.

Thanks,
MC

On Fri, Dec 12, 2008 at 7:37 PM, Frank @ Impact fr...@impactfax.com wrote:
 Not much written in the wiki on this.  Also searched the list and not much
 on either sched_api or uuid_send_dtmf.

 So from an xml dialplan, can sched_api as an application?

 Is there any way to have the time offset reference the point at which the
 call started ?  ie. When the called party answers?



 Ultimately, I was trying to insert some xml into my dial plan that would
 play a dtmf tone 10 seconds after the called party picked up the phone.  But
 from the little that has been written so far that I can find, it is not
 clear to me how to piece this together.  Am I being dense and missing
 anything that has already been written?



 /f

 -Original Message-
 From: freeswitch-users-boun...@lists.freeswitch.org
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
 West

 sched_api  (hint uuid_send_dtmf)



 API CALL [sched_api()] output:

 -ERR Invalid syntax. USAGE: [...@]time group_name command_string





 /b



 On Dec 12, 2008, at 5:51 PM, Frank @ Impact wrote:

 Is there a way to schedule a certain DTMF tone to be played into a bridge
 (both a and b legs) after a scheduled number of seconds into the call?

 ___




 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org



___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

Re: [Freeswitch-users] schedule a DTMF tone into bridge