[Freeswitch-users] Fail over

2009-10-05 Thread srinivasula reddy
Hi,

can any know how to implement fail over with freeswitch, please help me

Regards
-- 
Srinivasula Reddy K
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Re: [Freeswitch-users] Fail over

2009-10-05 Thread Giovanni Maruzzelli
On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:

 can any know how to implement fail over with freeswitch, please help me


This issue has been debated many many times in the mailing lists.
(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Giovanni Maruzzelli
On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
 Very nice :)

 On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com wrote:

 Folks;
 Here's something that I did playing around w/ learning Apple Motion.

Me too: very nice!

-gmaruzz

-- 
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Cell : +39-347-2665618

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[Freeswitch-users] oz dump Saying error

2009-10-05 Thread lakshmanan ganapathy
HI all,
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Re: [Freeswitch-users] oz dump Saying error

2009-10-05 Thread Diego Viola
Hello?

On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:

 HI all,


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[Freeswitch-users] oz debug says error

2009-10-05 Thread lakshmanan ganapathy
Hi all,
I've compiled the freeswitch with libpri support. But when I execute

oz libpri debug 1 all, I got the following error.

API CALL [oz(libpri debug 1 all )] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.


Here is my openzap configurations.

openzap.conf

[span zt 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=9/
/settings
pri_spans
span id=1 name=PRI_1
param name=q921loglevel value=alert/
param name=q931loglevel value=alert/
param name=mode value=user/
param name=dialect value=q931/
param name=dialplan value=XML/
param name=context value=default/
/span
/pri_spans
/configuration

I feel something I've missed in configurations. Please tell me how to get
rid of that error.
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Re: [Freeswitch-users] oz dump Saying error

2009-10-05 Thread lakshmanan ganapathy
Sorry my mail client has some problem. I've send another mail with my
question. Kindly ignore this one.

On Mon, Oct 5, 2009 at 12:43 PM, Diego Viola diego.vi...@gmail.com wrote:

 Hello?

 On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy lakindi...@gmail.com
  wrote:

 HI all,


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Re: [Freeswitch-users] Fail over

2009-10-05 Thread Giovanni Maruzzelli
On Mon, Oct 5, 2009 at 8:59 AM, Giovanni Maruzzelli gmar...@celliax.org wrote:
 On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
 srinivas.ksvre...@gmail.com wrote:

 can any know how to implement fail over with freeswitch, please help me


 This issue has been debated many many times in the mailing lists.
 (hint: no live call failover, HA with OpenSERet similia as load-balancers).

 Please have a look at the archives:

 http://lists.freeswitch.org/pipermail/freeswitch-dev/

 http://lists.freeswitch.org/pipermail/freeswitch-users/


On Mon, Oct 5, 2009 at 9:15 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
 Hi Giovanni Maruzzelli

 Thanks for your reply,
 i am new to is there any way to do live call failover.

Srinivas, are you joking ? Please take the time to read the answer,
when you ask a question.

In my previous mail, I have replied to you:

This issue has been debated many many times in the mailing lists.

(hint: no live call failover, HA with OpenSERet similia as load-balancers).

Please have a look at the archives:

http://lists.freeswitch.org/pipermail/freeswitch-dev/

http://lists.freeswitch.org/pipermail/freeswitch-users/

-gm
-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Fail over

2009-10-05 Thread Jan Berger

hi,

 

FreeSWITCH as is have no live fail-over support, but this will change.

 

A live fail-over and redundancy mechanism is part of what SIGTRAN provides of 
added values. I am working on this, but it will take time before this is on a 
functional stage and available.

 

Also - SIGTRAN only provide failover and redundancy on L2/L3 signalling or 
higher.

 

This will however not provide failover on E1/T1 hardware level. The later can 
be achieved by different techniques, but the easiest is if the xternal switch 
is configured to re-connect a lost call on a different E1/T1. Most proper 
switches can provide this service. 

 

Jan


 
 From: gmar...@celliax.org
 Date: Mon, 5 Oct 2009 08:59:34 +0200
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] Fail over
 
 On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
 srinivas.ksvre...@gmail.com wrote:
 
  can any know how to implement fail over with freeswitch, please help me
 
 
 This issue has been debated many many times in the mailing lists.
 (hint: no live call failover, HA with OpenSERet similia as load-balancers).
 
 Please have a look at the archives:
 
 http://lists.freeswitch.org/pipermail/freeswitch-dev/
 
 http://lists.freeswitch.org/pipermail/freeswitch-users/
 
 -gm
 
 
 -- 
 Sincerely,
 
 Giovanni Maruzzelli
 Cell : +39-347-2665618
 
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Re: [Freeswitch-users] oz debug says error

2009-10-05 Thread Russell Mosemann
 On Behalf Of lakshmanan ganapathy
...
 I've compiled the freeswitch with libpri support. But when I execute
 oz libpri debug 1 all, I got the following error.
 
 API CALL [oz(libpri debug 1 all )] output:
 src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.

If you would start freeswitch from the command line or look at 
freeswitch/log/freeswitch.log, you will see during startup that libpri does not 
find a span (note the ozmod lines). That's because the configuration below is 
not for libpri.

 openzap.conf.xml
...
 pri_spans

You need to use a libpri span configuration.

http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29

--
Russell Moseman


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Re: [Freeswitch-users] Detecting a fax

2009-10-05 Thread Mark Campbell-Smith
Thanks for the response Mike,

I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost.  This is an extract of my dialplan

extension name=Local
  condition field=destination_number expression=^(10[01][0-9])$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@${domain}/

I would assume that on detecting a fax, the dialplan 'fax' is called
in context features.  This never happens.

When is the fax tone detected?   Is it while the call is ringing or
can it be detected after the call is answered?  My goal is to be able
to have the same extension for a voice and fax call.  i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
 Also, I guess this doesn't work with media bypass (which I don't
use).

Thanks!


On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com wrote:
 check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

 Note, you can't just have tone_detect as your last iten in the
 dialplan as the call will just get hung up.

 Mike

 On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:

 Hi

 I was hoping someone could help me to setup the fax detection / tone
 detection application.

 I want to be able to transfer an incoming fax to a specific extension.
 In my default.xml file, I have the following (extracted):

    extension name=1000
      condition field=destination_number expression=^(10[01][0-9])
 $
        action application=fax_detect/
        action application=tone_detect data=fax 1100 r +5000
 transfer fax XML features /

 I can't get the fax to be detected and transferred.  Is there any way
 this can be done?

 Thanks!

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Re: [Freeswitch-users] configure FS: config.status: error: cannot find input file: Makefile.in

2009-10-05 Thread Tihomir Culjaga
it works,

thx!


T.

On Mon, Oct 5, 2009 at 12:31 AM, Michael Jerris m...@jerris.com wrote:

 I updated the tiff lib to build better inline, try make  tiff-reconf
 Mike

 On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:

 hello,
 i just got the last trunk and tried to compile it on one of my development
 machines... Well configure fails on tiff-3.8.2 where it is unable to find
 Makefile.in ... Can someone advice?



 checking if g++ static flag -static works... yes
 checking if g++ supports -c -o file.o... yes
 checking if g++ supports -c -o file.o... (cached) yes
 checking whether the g++ linker (/usr/bin/ld) supports shared libraries...
 yes
 checking dynamic linker characteristics... GNU/Linux ld.so
 checking how to hardcode library paths into programs... immediate
 checking for OpenGL Utility library... no
 checking for GLUT library... no
 configure: creating ./config.status
 config.status: error: cannot find input file: Makefile.in



 tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$ ls -l
 total 2224
 -rw-r--r--  1 tculjaga tculjaga  23741 2009-10-02 13:19 acinclude.m4
 -rw-r--r--  1 tculjaga tculjaga 316978 2009-10-02 13:28 aclocal.m4
 -rwxr-xr-x  1 tculjaga tculjaga121 2009-10-02 13:19 autogen.sh
 -rw-r--r--  1 tculjaga tculjaga 124047 2009-10-02 13:19 ChangeLog
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 config
 -rw-r--r--  1 tculjaga tculjaga 103926 2009-10-02 14:00 config.log
 -rwxr-xr-x  1 tculjaga tculjaga  73065 2009-10-02 14:00 config.status
 -rwxr-xr-x  1 tculjaga tculjaga 740145 2009-10-02 13:28 configure
 -rw-r--r--  1 tculjaga tculjaga  20492 2009-10-02 13:19 configure.ac
 -rwxr-xr-x  1 tculjaga tculjaga 56 2009-10-02 13:19 configure.gnu
 -rwxr-xr-x  1 tculjaga tculjaga 737794 2009-10-02 13:57 configure.lineno
 drwxr-xr-x 16 tculjaga tculjaga   4096 2009-10-02 13:19 contrib
 -rw-r--r--  1 tculjaga tculjaga   1146 2009-10-02 13:19 COPYRIGHT
 -rw-r--r--  1 tculjaga tculjaga   1570 2009-10-02 13:19 HOWTO-RELEASE
 drwxr-xr-x  5 tculjaga tculjaga   4096 2009-10-02 13:19 html
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:28 libtiff
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 m4
 -rw-r--r--  1 tculjaga tculjaga   1908 2009-10-02 13:19 Makefile.am
 -rw-r--r--  1 tculjaga tculjaga   1724 2009-10-02 13:19 Makefile.vc
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 man
 -rw-r--r--  1 tculjaga tculjaga   6270 2009-10-02 13:19 nmake.opt
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 port
 -rw-r--r--  1 tculjaga tculjaga   2363 2009-10-02 13:19 README
 -rw-r--r--  1 tculjaga tculjaga  9 2009-10-02 13:19 RELEASE-DATE
 -rw-r--r--  1 tculjaga tculjaga   5893 2009-10-02 13:19 SConstruct
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 test
 -rw-r--r--  1 tculjaga tculjaga433 2009-10-02 13:19 TODO
 drwxr-xr-x  3 tculjaga tculjaga   4096 2009-10-02 13:19 tools
 -rw-r--r--  1 tculjaga tculjaga  6 2009-10-02 13:19 VERSION
 tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$
 tculj...@subzero:~/freeswitch-trunk/libs/tiff-3.8.2$


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Re: [Freeswitch-users] oz debug says error

2009-10-05 Thread lakshmanan

Thanks for pointing that.
I also tried that.
But in that case, I'm not able to make a call through openzap.

When I say 
  originate openzap/1/A/number number

It reported the following error
2009-10-05 17:45:47.733495 [ERR] ozmod_libpri.c:88 Can't destroy call 0!
API CALL [originate(openzap/1/1/9952248266 9952248266)] output:
-ERR INVALID_IE_CONTENTS

I also gone and looked up the hangup_cause page for the reason. But I was
unable to understand that.
Can u please tell why it is reporting this error?



Russell.Mosemann wrote:
 
 On Behalf Of lakshmanan ganapathy
 ...
 I've compiled the freeswitch with libpri support. But when I execute
 oz libpri debug 1 all, I got the following error.
 
 API CALL [oz(libpri debug 1 all )] output:
 src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.
 
 If you would start freeswitch from the command line or look at
 freeswitch/log/freeswitch.log, you will see during startup that libpri
 does not find a span (note the ozmod lines). That's because the
 configuration below is not for libpri.
 
 openzap.conf.xml
 ...
 pri_spans
 
 You need to use a libpri span configuration.
 
 http://wiki.freeswitch.org/wiki/Openzap.conf.xml_Examples#Using_with_PRI_.28libpri_compatibility_stack.29
 
 --
 Russell Moseman
 
 
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[Freeswitch-users] Recording creates a 388-byte long file and deletes it

2009-10-05 Thread Maciej Aniserowicz
Hi,
When I record a call in FS, it only creates a 388-byte-long wav file. The 
conversation is no written there, and FS deletes the file when the session 
finishes.
What can cause this strange behavior?

Br/
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[Freeswitch-users] gateway FS informs it's client FS about users hanged up with a long delay

2009-10-05 Thread Maciej Aniserowicz
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all users 
register to the 'external' instance which acts as a gateway by 'internal' 
instance (which in turn is controlled by my applicaiton with commands sent by 
socket).
When user hangs up, the 'hanged up' event is propagated to the 'internal' 
instance after a long time (~3 minutes) instead of being propagated immediately.
What can cause this issue?

Br/
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[Freeswitch-users] Bad sound quality while eavesdropping

2009-10-05 Thread Maciej Aniserowicz
Hello,
When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is 
there any way to improve it? Is this a known problem?
Br/
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Re: [Freeswitch-users] Detecting a fax

2009-10-05 Thread Michael Jerris
Fax tones are not played by the remote machine until after answer, the  
tone_detect application starts a media bug that listens for the tone,  
can you confirm the tone is happening at all.  Maybe the issue here is  
the timeout, try making that longer, or doing the tone_detect in  
execute_on_answer

Mike

On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:

 Thanks for the response Mike,

 I read that page and this one (among others)
 http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
 I'm still lost.  This is an extract of my dialplan

extension name=Local
  condition field=destination_number expression=^(10[01][0-9]) 
 $
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
 transfer fax XML features /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@$ 
 {domain}/

 I would assume that on detecting a fax, the dialplan 'fax' is called
 in context features.  This never happens.

 When is the fax tone detected?   Is it while the call is ringing or
 can it be detected after the call is answered?  My goal is to be able
 to have the same extension for a voice and fax call.  i assume that
 the fax 'tones' are standardised and the ones on the wiki are correct?
 Also, I guess this doesn't work with media bypass (which I don't
 use).

 Thanks!


 On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com  
 wrote:
 check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

 Note, you can't just have tone_detect as your last iten in the
 dialplan as the call will just get hung up.

 Mike

 On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:

 Hi

 I was hoping someone could help me to setup the fax detection / tone
 detection application.

 I want to be able to transfer an incoming fax to a specific  
 extension.
 In my default.xml file, I have the following (extracted):

extension name=1000
  condition field=destination_number expression=^(10[01] 
 [0-9])
 $
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
 transfer fax XML features /

 I can't get the fax to be detected and transferred.  Is there any  
 way
 this can be done?

 Thanks!


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[Freeswitch-users] stun not working in fs 1.0.4?

2009-10-05 Thread Vladimir Elizarov
Hello.

I'm trying to configure stun in fs 1.0.4

vars.xml
X-PRE-PROCESS cmd=set data=bind_server_ip=auto/
X-PRE-PROCESS cmd=set data=external_sip_ip=stun:example.com/
X-PRE-PROCESS cmd=set data=external_rtp_ip=stun:example.com/

external.xml
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=ext-rtp-ip value=$${external_rtp_ip}/

In this configuration, the address in ext_rtp_ip transmitted literally
(it is: stun:stun.exmaple.com)

If you do not specify ext_rtp_ip stun then allegedly began to work. But
as in bug SFSIP-163
(http://jira.freeswitch.org/browse/SFSIP-163?page=com.atlassian.jira.plugin.system.issuetabpanels%3Acomment-tabpanel
) Via: IP is replaced, and SDP - no.

How do I fix this? Upgrade fs to trunk?



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Re: [Freeswitch-users] oz debug says error

2009-10-05 Thread Russell.Mosemann
lakshmanan lakindi...@gmail.com said:
 Thanks for pointing that.
 I also tried that.
 But in that case, I'm not able to make a call through openzap.

What is in openzap.conf.xml? If you start fs_cli and enter oz list,
what does it show? Copy the ozmod lines from freeswitch.log to
pastebin.freeswitch.org and post the link here so that we can see what
openzap does when freeswitch starts.

-- 
Russell Mosemann




Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


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[Freeswitch-users] Re corded file as voicemail.

2009-10-05 Thread Nagalenoj

Is it possible to treat a recorded voice as voice mail?

Assume that, I've recorded a conversation and I want this recorded file to
be treated like voicemail. So, I could check it like voicemail!!

-- 
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[Freeswitch-users] UUID of the newly originated call?

2009-10-05 Thread Nagalenoj

Dear friends,
 I am trying with ESL outbound socket. I'm trying to make a call when I
receive ANSWER event. Now, I would want to do something like,
 * Receive the events only for this uuid - I have done by registering
all events and filtering only for this uuid($uuid).
 * If it is CHANNEL_ANSWER, originate a new call.

Now, How can I get the uuid of the new call and receive events for this new
call.? I want to receive the events for both uuids.



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Re: [Freeswitch-users] Bad sound quality while eavesdropping

2009-10-05 Thread Anthony Minessale
That's is a somewhat vague position.

You did not mention which version of FreeSWITCH you are running, the phones
being used in your example, your configuration, the codecs in use etc.

BTW,
I think you should only ask one question at a time on this list.  The list
is run by volunteers and it's sort of rude to expect 3 or 4 threads to be
tended to concerning the same one individual.


2009/10/5 Maciej Aniserowicz maciej.aniserow...@gmail.com

  Hello,
 When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is
 there any way to improve it? Is this a known problem?
 Br/
 Maciej Aniserowicz

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Re: [Freeswitch-users] Detecting a fax

2009-10-05 Thread Tihomir Culjaga
hi Mark,

This is an inbound call leg and media channel (so far)  is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to hear the fax tone.

T.



On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:

 Fax tones are not played by the remote machine until after answer, the
 tone_detect application starts a media bug that listens for the tone,
 can you confirm the tone is happening at all.  Maybe the issue here is
 the timeout, try making that longer, or doing the tone_detect in
 execute_on_answer

 Mike

 On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:

  Thanks for the response Mike,
 
  I read that page and this one (among others)
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
  I'm still lost.  This is an extract of my dialplan
 
 extension name=Local
   condition field=destination_number expression=^(10[01][0-9])
  $
 action application=set data=dialed_extension=$1/
 action application=export data=dialed_extension=$1/
 action application=set data=ringback=${au-ring}/
 action application=fax_detect/
 action application=tone_detect data=fax 1100 r +5000
  transfer fax XML features /
 action application=set data=hangup_after_bridge=true/
 action application=set data=continue_on_fail=true/
 action application=bridge data=user/${dialed_extensi...@$
  {domain}/
 
  I would assume that on detecting a fax, the dialplan 'fax' is called
  in context features.  This never happens.
 
  When is the fax tone detected?   Is it while the call is ringing or
  can it be detected after the call is answered?  My goal is to be able
  to have the same extension for a voice and fax call.  i assume that
  the fax 'tones' are standardised and the ones on the wiki are correct?
  Also, I guess this doesn't work with media bypass (which I don't
  use).
 
  Thanks!
 
 
  On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
  wrote:
  check out
 http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
 
  Note, you can't just have tone_detect as your last iten in the
  dialplan as the call will just get hung up.
 
  Mike
 
  On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
 
  Hi
 
  I was hoping someone could help me to setup the fax detection / tone
  detection application.
 
  I want to be able to transfer an incoming fax to a specific
  extension.
  In my default.xml file, I have the following (extracted):
 
 extension name=1000
   condition field=destination_number expression=^(10[01]
  [0-9])
  $
 action application=fax_detect/
 action application=tone_detect data=fax 1100 r +5000
  transfer fax XML features /
 
  I can't get the fax to be detected and transferred.  Is there any
  way
  this can be done?
 
  Thanks!


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Re: [Freeswitch-users] stun not working in fs 1.0.4?

2009-10-05 Thread Rupa Schomaker
Yes, the stun thing was fixed after 1.4 I believe.

On Mon, Oct 5, 2009 at 6:43 AM, Vladimir Elizarov
xengelpubl...@gmail.com wrote:

 How do I fix this? Upgrade fs to trunk?



-- 
-Rupa

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[Freeswitch-users] overriding conference preference

2009-10-05 Thread Woody Dickson
Hi,

Is is possible to override any of the setting specified in the conference
profile?

What I want to do is to have a default profile, and be able to modify
certain fields if necessary in the dialplan.


Alternatively, I would prefer to have a dynamic profile setting for the
conference to obtain those parameters from odbc.

Is it possible?

woody
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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Anthony Minessale
neat,

Here's some suggestions for your next ones. =p

Have them standing around the hologram trying to destroy the Death
Star(tm) that happens to look a lot like a giant 3d unix '*' character.
Then have one rebel say, wait!, why are we wasting our time... watch
this... and dial a number on his cellphone as the whole thing explodes in
the background.

Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you have
fashioned your own TDM card vroom..
Join me and together we can make linked lists and monolithic processes,
NEVER!... vroom vroom Master Coda has taught you well.You are no
match for me...JOIN THE ORANGE SIDE OF THE FORCE





On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli gmar...@celliax.orgwrote:

 On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
  Very nice :)
 
  On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com
 wrote:
 
  Folks;
  Here's something that I did playing around w/ learning Apple Motion.

 Me too: very nice!

 -gmaruzz

 --
 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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Re: [Freeswitch-users] Re corded file as voicemail.

2009-10-05 Thread Michael Jerris
http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject

On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:


 Is it possible to treat a recorded voice as voice mail?

 Assume that, I've recorded a conversation and I want this recorded  
 file to
 be treated like voicemail. So, I could check it like voicemail!!


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Re: [Freeswitch-users] stun not working in fs 1.0.4?

2009-10-05 Thread Brian West
Yes!

/b

On Oct 5, 2009, at 7:43 AM, Vladimir Elizarov wrote:

 How do I fix this? Upgrade fs to trunk?


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Re: [Freeswitch-users] overriding conference preference

2009-10-05 Thread Michael Jerris

On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:

 Is is possible to override any of the setting specified in the  
 conference profile?

Just the flags you can pass per user such as pin and mute


 What I want to do is to have a default profile, and be able to  
 modify certain fields if necessary in the dialplan.


 Alternatively, I would prefer to have a dynamic profile setting for  
 the conference to obtain those parameters from odbc.

you can do this with mod_xml_curl

Mike


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[Freeswitch-users] Gateways, Limits, Routes

2009-10-05 Thread Kevin Green
It seems many people are looking for ways to control gateways, resiliency of
termination, and limit on connections easily in FS. Here are some of the
thoughts I had, and I would like to hear what others think of this.

In tradition phone hardware you would define lines, put them into a group,
and then assign a route to go through that group. For resiliency you could
group multiple routes together into a route list, if the first route failed,
was all busy, or unavailable it would start to try the second route. If you
go out a secondary route you can also play a warning tone to indicate this
might be going out a more expensive connection. For example you may have an
IP link between two boxes, then fail back to TDM if the IP links go down,
the TDM would be more expensive so you would want to warn business users so
they don't spend hours on the phone.

Inbound and outbound calls could both go on the lines so when a call comes
in or goes out a route it takes up one slot.

My thought is, why don't we create something similar to this that will allow
us to handle a lot of these cases without complex dialplans. We could create
routes that are assigned to gateways and limit the number of incoming,
outgoing, and total connections that can be on that route. We could also
specify what cases we would consider as failure to move onto the next route
if we are in a route list. Route lists would then be similar to using
multiple limits, and failover, but in this case it would simply be a list.

The following is an example of how I think this could be prgrammed.

routes
   route id=route1 gateway=provider1 max-inbound=5 max-outbound=5
max-total=5
countinue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION/
   route id=route2 gateway=provider2 max-inbound=0 max-outbound=5
max-total=5 countinue_on_fail=NORMAL_TEMPORARY_FAILURE/
   route id=route3 gateway=provider3 max-inbound=5 max-outbound=5
max-total=10 countinue_on_fail=NO_ROUTE_DESTINATION/
/routes
route_lists
   route_list id=list1
  route id=route1/
  route id=route2 warning_tone=expensive.wav/
  route id=route3 warning_tone=really_expensive.wav/
   /route_list
/route_litsts

Then in your dialplan you would simply put:

extension name=outbound_rout
   condition field=destination_number expression=^(\d{10})$
  action application=bridge data=route/route2/$1/
   /condition
/extension

OR

extension name=outbound_route
   condition field=destination_number expression=^(\d{10})$
  action application=bridge data=route_list/list1/$1/
   /condition
/extension

   The module would need to track not only outbound calls, but also inbound
calls that come in through the specified gateways. This would help track
cases where you have say only two channels that can be used for both
incoming and outgoing calls.

   I'm not 100% sure if this is all feasable or if it would be that much of
an improvement compaired to what is already there so I put it out to all of
you for feedback.

Regards,
   Kevin Green

JohnnyVoIP
350 Legget Drive
Kanata, ON, Canada
K2K 2W7

Phone:   613 271 5993 Ext 1203
Fax:   613 271 9810
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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Fredy Gonzales
Espectacular!!.

FG
- Original Message - 
From: Giovanni Maruzzelli gmar...@celliax.org
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, October 05, 2009 2:04 AM
Subject: Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video


 On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
 Very nice :)

 On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com wrote:

 Folks;
 Here's something that I did playing around w/ learning Apple Motion.

 Me too: very nice!

 -gmaruzz

 -- 
 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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Re: [Freeswitch-users] New to freeswitch and have a few questions

2009-10-05 Thread Michael Collins
I've got a copy of Mike G's document that I'm reviewing. Like he said, it's
not 100% complete, however at first glance it looks like it would be a
perfect fit for a config example page. I'll get with Mike G shortly and
we'll have it up on the wiki in the next few days for everyone to review.

-MC

On Sun, Oct 4, 2009 at 3:35 PM, Michael Jerris m...@jerris.com wrote:

 Getting documentation on like this on the wiki would be awesome.
 Mike

 On Oct 2, 2009, at 12:10 PM, Michael Gende wrote:

 Hey Orien,

 I'm not using exactly your set up, but am using pfsense/FreeBSD. Since
 you're using that, I assume you're going dual homed. I've got a starter
 guide that might help you out. If nothing else, I'd be interested in a
 candid assessment of its usefulness or lack thereof, especially to a guy
 like you.

 I've included it here. Its all just text at the moment so be advised. Also
 be advised that there's a lot of great information on the freeswitch site
 and in this group. The goal of my document was so that someone just starting
 would have to hunt a little less.

 Hope its good for something, let me know either way, especially if you find
 errors.

 Regards,

 Mike G.

 On Thu, Oct 1, 2009 at 8:11 PM, Orien Love or...@tx.rr.com wrote:

 Hello Everybody,
I am new to freeswitch, so forgive me if I ask stupid questions. I
 am planning a test setup consisting of:
1 - Pfsense router with the freeswitch package installed.
1 - Cisco WS-C3524-PWR-XL-EN 24 FE Switch, POE for the phones.
1 - LINKSYS SPA3000 to connect to my existing land line and phones.
2 - POLYCOM SOUNDPOINT IP601 SIP IP PHONEs

 The first question I have, Are the IP601 phones supported? The wiki
 lists 320, 431, 501, 550, 650 but not the 601.

 Second, is there a place that helps a person new to the IP phone world
 learn what is needed to set up a PBX using freeswitch at a small office?

 Finally is my test setup a good one? is there something I am missing or
 that I need to get the learning process started, I have found in the
 past, with a little information and a test system, I can learn what I am
 doing by breaking and fixing the test bed.

 Thanks for your time
 Orien

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Re: [Freeswitch-users] Problem with gateway registration

2009-10-05 Thread Michael Collins
On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote:

 Mike, how exactly should I format the file? I got the pcap file, how do I
 convert it to text so that you can easily read it?


you can open it with wireshark, follow the TCP or UDP stream, then just copy
 paste the text as needed...

-MC
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Re: [Freeswitch-users] FS Does Not RelayPresencePUBLISHToSubscribing Phones

2009-10-05 Thread Jerry Richards
Okay, I added the ; at the end of the sqlite3 select command and it just
returned to the sqlite prompt.  No error was returned.  Do you see
anything in my database (in the pastebin) that is incorrect?  By the way,
the select command I put in the pastebin refers to the external config,
but the internal config does the  same thing.

Best Regards,
Jerry


-Original Message-
From: Rupa Schomaker [mailto:r...@rupa.com] 
Sent: Friday, October 02, 2009 11:42 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not
RelayPresencePUBLISHToSubscribing Phones

You are missing the trailing ;

On Fri, Oct 2, 2009 at 1:28 PM, Jerry Richards jerry.richa...@teotech.com
wrote:
 I put the sqlite3 select query in the paste bin, and prior to that, I 
 entered the .dump command.  The select command came back with a ...
 prompt which I don't understand.  I don't know enough about sqlite3 to 
 know what that means?

 Best Regards,
 Jerry


--
-Rupa




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Re: [Freeswitch-users] UUID of the newly originated call?

2009-10-05 Thread Benedikt Fraunhofer
Hi,

2009/10/5 Nagalenoj nagale...@gmail.com:
     * Receive the events only for this uuid - I have done by registering
 all events and filtering only for this uuid($uuid).
     * If it is CHANNEL_ANSWER, originate a new call.

it's a filter in, not filter out :)

 Now, How can I get the uuid of the new call and receive events for this new
 call.? I want to receive the events for both uuids.

You can specify the UUID of an originated call by doing the following:

* Use create_uuid to generate a UUID to use.
* This will allow you to kill an originated call before it is
answered by using uuid_kill.
* The UUID of the answered call leg will not be the same UUID as
the origination_uuid specified (Each call leg always gets its own
UUID)

originate [origination_uuid=...]user/1...@domain.name.com

shamelessly ripped from
http://wiki.freeswitch.org/wiki/Mod_commands#originate

at least it worked for me.

  Beni.

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Re: [Freeswitch-users] Connecting FS to Hicom 300

2009-10-05 Thread Russell.Mosemann
Michael Collins m...@freeswitch.org said:

 Confirmed: the Hicomm isn't sending anything at all in the SETUP message
 except the usual stuff: dialed number, channel number, etc. Does the Hicomm
 have any config parameters, like Caller ID presentation?

I believe it does, but I don't have access to the Hicom. I have to go
through the phone guy. It's kind of a delicate situation. At least I have
something to suggest for him to investigate. I appreciate the confirmation.

-- 
Russell Mosemann




Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


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[Freeswitch-users] SLAs and BLAs

2009-10-05 Thread Jerry Richards

I can see how BLFs and Presence are managed, however I haven't found much
documentation on SLAs and BLAs.  What is the RFC(s) that Freeswitch used to
implement SLAs and BLAs?  Do they differ from BLFs?

Best Regards,
Jerry


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Re: [Freeswitch-users] stun not working in fs 1.0.4?

2009-10-05 Thread Brian West
Because TRUNK is stable... its only fixes going in usually and if  
things do break they don't stay broken for long.

Ask anyone our trunk is more table then most commercial products.

/b

On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:

 Ok. Brian, why fs no two branches of the stable and trunk?


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Re: [Freeswitch-users] Connecting FS to Hicom 300

2009-10-05 Thread Michael Collins
On Mon, Oct 5, 2009 at 10:50 AM, russell.mosem...@cune.org wrote:

 Michael Collins m...@freeswitch.org said:

  Confirmed: the Hicomm isn't sending anything at all in the SETUP message
  except the usual stuff: dialed number, channel number, etc. Does the
 Hicomm
  have any config parameters, like Caller ID presentation?

 I believe it does, but I don't have access to the Hicom. I have to go
 through the phone guy. It's kind of a delicate situation. At least I have
 something to suggest for him to investigate. I appreciate the confirmation.


If you need proverbial ammo let me know. If he speaks Q931 then your
pastebin is the ultimate proof for him that the hicom is not sending any
caller ID info. In any case, I'm here if you need assistance.
-MC
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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Diego Viola
Nice script Anthony, that would be amazing to have on video ;)

On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale 
anthony.miness...@gmail.com wrote:

 neat,

 Here's some suggestions for your next ones. =p

 Have them standing around the hologram trying to destroy the Death
 Star(tm) that happens to look a lot like a giant 3d unix '*' character.
 Then have one rebel say, wait!, why are we wasting our time... watch
 this... and dial a number on his cellphone as the whole thing explodes in
 the background.

 Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you have
 fashioned your own TDM card vroom..
 Join me and together we can make linked lists and monolithic processes,
 NEVER!... vroom vroom Master Coda has taught you well.You are no
 match for me...JOIN THE ORANGE SIDE OF THE FORCE






 On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli 
 gmar...@celliax.orgwrote:

 On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com
 wrote:
  Very nice :)
 
  On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com
 wrote:
 
  Folks;
  Here's something that I did playing around w/ learning Apple Motion.

 Me too: very nice!

 -gmaruzz

 --
 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

 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

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[Freeswitch-users] FreeSWITCH Weekly Conference Call Followup, Agenda For This Week

2009-10-05 Thread Michael Collins
Ladies and Gentlemen,

Thank you for calling in to the weekly FreeSWITCH conference call. Last
week's agenda was rather light, so if you have things that you would like to
have discussed please be sure to add them here:

http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09

Here are a few updates for everyone to keep in mind:
* Starting with the upcoming meeting (Oct 9) the conference will support
48kHz CELT codec.
* When you call in, please mute your phone if you are just listening or if
you will not be speaking a lot. Background noise from one user isn't usually
too bad, but when 10 or 15 people are not muted it can get a little
distracting. :)
* We are always looking for people to help out. If you are looking for ways
to help out then please by all means call the conference. If you cannot call
the conference for some reason but still want to help out, please email me
off list. We have numerous documentation, janitorial, code review, etc.
sub-projects that FreeSWITCH users can help with.
* Be sure to update to latest SVN trunk and test test test! We are prepping
for 1.0.5 and more people testing means a more stable release delivered more
quickly.

One topic that came up was the testing of Mike van Lammerman's Ultramonkey
setup. (http://bit.ly/8zZUS) Several people agreed to try it out. We would
love to see others try it out and report back their experiences.

Lastly, we'd like to thank everyone who has been helping out with testing,
documentation, and answering questions on the email list and IRC channels. A
special thanks to Diego Viola for doing lots of wiki cleanup lately.
Everyone's efforts are appreciated.

-Michael
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Re: [Freeswitch-users] Connecting FS to Hicom 300

2009-10-05 Thread Russell.Mosemann
Michael Collins m...@freeswitch.org said:

 If you need proverbial ammo let me know. If he speaks Q931 then your
 pastebin is the ultimate proof for him that the hicom is not sending any
 caller ID info. In any case, I'm here if you need assistance.

Heh, I have the exact opposite problem. I don't think he configured a PRI
T1 before, and the debug output would be meaningless to him. He usually
handles everyday my phone doesn't work kind of issues. I try to nudge
in the right direction without being pushy. :-)

-- 
Russell Mosemann




Concordia University, Nebraska
See http://www.cune.edu/ for the latest news and events!


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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread Dmitry Bely
On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
 http://wiki.freeswitch.org/wiki/Mod_limit

Of course I've read that. But it only affects the number of outgoing
calls (at least for gateways - chapter Using mod_limit with an
outbound gateway). But I would like to limit the number of all calls
(incoming+outgoing) via specific gateway. Any idea?

- Dmitry Bely

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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread SP
Direction doesn't matter, it uses realm's and a few other vars.  Use
the same vars for both directions.

On Mon, Oct 5, 2009 at 14:57, Dmitry Bely dmitry.b...@gmail.com wrote:
 On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
 http://wiki.freeswitch.org/wiki/Mod_limit

 Of course I've read that. But it only affects the number of outgoing
 calls (at least for gateways - chapter Using mod_limit with an
 outbound gateway). But I would like to limit the number of all calls
 (incoming+outgoing) via specific gateway. Any idea?

 - Dmitry Bely

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-- 
Shannon

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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread M.Emran
Incoming Call-Limit http://www.howtonix.com/?p=89

Outbound Call-Limit http://www.howtonix.com/?p=86



On Tue, Oct 6, 2009 at 2:19 AM, SP spr...@gmail.com wrote:

 Direction doesn't matter, it uses realm's and a few other vars.  Use
 the same vars for both directions.

 On Mon, Oct 5, 2009 at 14:57, Dmitry Bely dmitry.b...@gmail.com wrote:
  On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
  http://wiki.freeswitch.org/wiki/Mod_limit
 
  Of course I've read that. But it only affects the number of outgoing
  calls (at least for gateways - chapter Using mod_limit with an
  outbound gateway). But I would like to limit the number of all calls
  (incoming+outgoing) via specific gateway. Any idea?
 
  - Dmitry Bely
 
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-- 
Regards
--
M Emran
Managing Director
E-SOFT BILLING PTE. LTD.
Web: www.e-softbilling.com
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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread Dmitry Bely
On Tue, Oct 6, 2009 at 12:19 AM, SP spr...@gmail.com wrote:
 Direction doesn't matter, it uses realm's and a few other vars.  Use
 the same vars for both directions.

Unfortunately it does.

  action application=limit data=$${domain} my_provider 1/

generates limit_exceeded for the second outbound call, but if an
incoming call is active FreeSWITCH still tries to use this gateway.

 On Mon, Oct 5, 2009 at 14:57, Dmitry Bely dmitry.b...@gmail.com wrote:
 On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
 http://wiki.freeswitch.org/wiki/Mod_limit

 Of course I've read that. But it only affects the number of outgoing
 calls (at least for gateways - chapter Using mod_limit with an
 outbound gateway). But I would like to limit the number of all calls
 (incoming+outgoing) via specific gateway. Any idea?

 - Dmitry Bely

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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread Dmitry Bely
On Tue, Oct 6, 2009 at 12:32 AM, M.Emran monem...@gmail.com wrote:
 Incoming Call-Limit http://www.howtonix.com/?p=89

 Outbound Call-Limit http://www.howtonix.com/?p=86

But what if I need to limit the total number of calls (in my case == 1)?

 On Tue, Oct 6, 2009 at 2:19 AM, SP spr...@gmail.com wrote:

 Direction doesn't matter, it uses realm's and a few other vars.  Use
 the same vars for both directions.

 On Mon, Oct 5, 2009 at 14:57, Dmitry Bely dmitry.b...@gmail.com wrote:
  On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
  http://wiki.freeswitch.org/wiki/Mod_limit
 
  Of course I've read that. But it only affects the number of outgoing
  calls (at least for gateways - chapter Using mod_limit with an
  outbound gateway). But I would like to limit the number of all calls
  (incoming+outgoing) via specific gateway. Any idea?

- Dmitry Bely

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Re: [Freeswitch-users] How to limit the number of incoming+outgoing calls via specific gateway?

2009-10-05 Thread SP
did you use the application limit on the inbound call?  You'll need to
in order to account for it.

On Mon, Oct 5, 2009 at 15:39, Dmitry Bely dmitry.b...@gmail.com wrote:
 On Tue, Oct 6, 2009 at 12:19 AM, SP spr...@gmail.com wrote:
 Direction doesn't matter, it uses realm's and a few other vars.  Use
 the same vars for both directions.

 Unfortunately it does.

      action application=limit data=$${domain} my_provider 1/

 generates limit_exceeded for the second outbound call, but if an
 incoming call is active FreeSWITCH still tries to use this gateway.

 On Mon, Oct 5, 2009 at 14:57, Dmitry Bely dmitry.b...@gmail.com wrote:
 On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
 http://wiki.freeswitch.org/wiki/Mod_limit

 Of course I've read that. But it only affects the number of outgoing
 calls (at least for gateways - chapter Using mod_limit with an
 outbound gateway). But I would like to limit the number of all calls
 (incoming+outgoing) via specific gateway. Any idea?

 - Dmitry Bely

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Re: [Freeswitch-users] Basic compile question.

2009-10-05 Thread David Clark
Ok using windows xp x64 here.  I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say 
unsupported.

But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error 
C1083: Cannot open include file: 'ptlib.h': No such file or directory
.\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 
'spandsp.h': No such file or directory

Even if the project file is wrong or out of date I should be able to 
find the include files some place in the fileset.
I can't find either file in the freeswitch directory or below it.

Any idea what is up?

Thanks,
David Clark


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[Freeswitch-users] mod_xml_curl http POST is inconsistent/bug

2009-10-05 Thread Christian Damianidis
Trying to achieve dynamic binding of user directory. In short: It's not
doing the authorization properly. I can use curl in the command line and
it works perfectly, specifying BASIC auth.. however with the freeswitch
module it returns HTTP 401. 

 

So I've taken a close look at the network packets being sent and there
are some issues.

 

This is between the bindings/bindings tags in my xml_curl.conf.xml
(1.2.3.4 represents my webserver's IP)

 

binding name=users

param name=gateway-url
value=http://1.2.3.4:2000/users.aspx; bindings=directory/

param name=gateway-credentials
value=username:password/

/binding

 

When I run curl -basic -u username http://1.2.3.4:2000/users.aspx; it
asks me for a password and returns the correct thing. I use tshark to
monitor, and it sends a GET request, with the correct authorization
credentials in the header. I receive an HTTP 200 OK packet and the xml
follows. 

 

When I startup freeswitch, I guess the xml curl module gets to run, and
it makes the request. However this time it's a POST, and oddly DOES NOT
include the Authorization: Basic encodedstring line in the packet. I
get back two HTTP 401 Unauthorized responses, and then freeswitch sends
out another POST, this time includes the authorization line, and I get
back an OK with the xml.

My user directory is updated and we're all good.

 

The inconsistent POST request sent by the module causes freeswitch to
hang for 1-2 minutes during start-up.

 

Has anyone else had this issue? Is this a bug or intended functionality
(ping the server before making a real request?). I'd love to sort this
out, otherwise getting an updated directory isn't real-time, thus
defeating the purpose.

Thanks,

 

Christian

 

 

 

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Re: [Freeswitch-users] UPDATED: Basic compile question.

2009-10-05 Thread David Clark
Ok I found spandsp.h.  It is a case of the project file being out of 
date.  No surprise.  ptlib.h is still not found.


Ok using windows xp x64 here.  I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say 
unsupported.

But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error 
C1083: Cannot open include file: 'ptlib.h': No such file or directory
.\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 
'spandsp.h': No such file or directory


Even if the project file is wrong or out of date I should be able to 
find the include files some place in the fileset.
I can't find either file in the freeswitch directory or below it.

Any idea what is up?

Thanks,
David Clark


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Re: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug

2009-10-05 Thread Brian West

Are you using something other than apache?

/b

On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:



The inconsistent POST request sent by the module causes freeswitch  
to hang for 1-2 minutes during start-up.


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Re: [Freeswitch-users] Basic compile question.

2009-10-05 Thread Brian West
Have you updated today?

/b

On Oct 5, 2009, at 3:49 PM, David Clark wrote:

 Any idea what is up?


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Re: [Freeswitch-users] UPDATED: Basic compile question.

2009-10-05 Thread Michael Jerris
voip codecs is fixed, ptlib I can't recall if we ever did full build  
integration or if you needed to manually download the libraries, can  
someone who has done mod_opal build on windows comment?

Mike


On Oct 5, 2009, at 5:14 PM, David Clark wrote:

 Ok I found spandsp.h.  It is a case of the project file being out of
 date.  No surprise.  ptlib.h is still not found.

 
 Ok using windows xp x64 here.  I download from the trunk as expected.
 I fire up VS 2005 and I open the VS 2005 solution.  Yes it does say
 unsupported.

 But I get two missing header files:
 freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error
 C1083: Cannot open include file: 'ptlib.h': No such file or directory
 .\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file:
 'spandsp.h': No such file or directory


 Even if the project file is wrong or out of date I should be able to
 find the include files some place in the fileset.
 I can't find either file in the freeswitch directory or below it.


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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Jay Binks

Haha classic !!!

Can't wait for the next installment in the series !!

J


On 06/10/2009, at 1:02, Anthony Minessale  
anthony.miness...@gmail.com wrote:



neat,

Here's some suggestions for your next ones. =p

Have them standing around the hologram trying to destroy the Death  
Star(tm) that happens to look a lot like a giant 3d unix '*'  
character.  Then have one rebel say, wait!, why are we wasting our  
time... watch this... and dial a number on his cellphone as the  
whole thing explodes in the background.


Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you  
have fashioned your own TDM card vroom..
Join me and together we can make linked lists and monolithic  
processes, NEVER!... vroom vroom Master Coda has taught you  
well.You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE






On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli gmar...@celliax.org 
 wrote:
On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com  
wrote:

 Very nice :)

 On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com  
wrote:


 Folks;
 Here's something that I did playing around w/ learning Apple  
Motion.


Me too: very nice!

-gmaruzz

--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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MSN:anthony_miness...@hotmail.com
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sip:8...@conference.freeswitch.org
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Re: [Freeswitch-users] SLAs and BLAs

2009-10-05 Thread Jerry Richards
We are building our own in-house developed Teo phones.  I also have
CounterPath's Bria Professional phone.  For test purposes, I have one snom
phone and a couple Polycomm phones.

Jerry
 

-Original Message-
From: Brian West [mailto:br...@freeswitch.org] 
Sent: Monday, October 05, 2009 11:02 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] SLAs and BLAs

First off what phones are you going to be using?

/b

On Oct 5, 2009, at 12:58 PM, Jerry Richards wrote:


 I can see how BLFs and Presence are managed, however I haven't found 
 much documentation on SLAs and BLAs.  What is the RFC(s) that 
 Freeswitch used to implement SLAs and BLAs?  Do they differ from BLFs?

 Best Regards,
 Jerry





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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Giovanni Maruzzelli
The Revenge of the Sip

On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks jaybi...@gmail.com wrote:
 Haha classic !!!
 Can't wait for the next installment in the series !!

 J


 On 06/10/2009, at 1:02, Anthony Minessale anthony.miness...@gmail.com
 wrote:

 neat,

 Here's some suggestions for your next ones. =p

 Have them standing around the hologram trying to destroy the Death
 Star(tm) that happens to look a lot like a giant 3d unix '*' character.
 Then have one rebel say, wait!, why are we wasting our time... watch
 this... and dial a number on his cellphone as the whole thing explodes in
 the background.

 Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you have
 fashioned your own TDM card vroom..
 Join me and together we can make linked lists and monolithic processes,
 NEVER!... vroom vroom Master Coda has taught you well.You are no
 match for me...JOIN THE ORANGE SIDE OF THE FORCE





 On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli gmar...@celliax.org
 wrote:

 On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
  Very nice :)
 
  On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com
  wrote:
 
  Folks;
  Here's something that I did playing around w/ learning Apple Motion.

 Me too: very nice!

 -gmaruzz

 --
 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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 --
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 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

 AIM: anthm
 MSN:anthony_miness...@hotmail.com
 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
 IRC: irc.freenode.net #freeswitch

 FreeSWITCH Developer Conference
 sip:8...@conference.freeswitch.org
 iax:gu...@conference.freeswitch.org/888
 googletalk:conf+...@conference.freeswitch.org
 pstn:213-799-1400

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-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Karl Vesterling


Sadly the budget of time and props can't afford such extravagance...
But OMG I'm still laughing...

I was e-mailing earlier off the list, and came up with some nice  
names that could be put in the credits...


Like:

Anthony Minnessale -as- Obi-Code-Kenobi  (But I do like Master Coda  
from below.)


Mike Jerris -as- Luke Skypewalker

Richard Stallman -as- cpp30

Stuff like that...

I corrected to speech vs speach (my bad, sorry...)

Now, as far as the below, I imagine we could pull that off using the  
REAL footage, and dubbing in the audio if someone can do a decent  
Darth Spencer, errr Vader voice...  But something tells me that Mr.  
Lucas might get a bit peeved at such a thing...


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

On Oct 5, 2009, at 2:34 PM, Diego Viola wrote:


Nice script Anthony, that would be amazing to have on video ;)

On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale anthony.miness...@gmail.com 
 wrote:

neat,

Here's some suggestions for your next ones. =p

Have them standing around the hologram trying to destroy the Death  
Star(tm) that happens to look a lot like a giant 3d unix '*'  
character.  Then have one rebel say, wait!, why are we wasting our  
time... watch this... and dial a number on his cellphone as the  
whole thing explodes in the background.


Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you  
have fashioned your own TDM card vroom..
Join me and together we can make linked lists and monolithic  
processes, NEVER!... vroom vroom Master Coda has taught you  
well.You are no match for me...JOIN THE ORANGE SIDE OF THE FORCE







On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli gmar...@celliax.org 
 wrote:
On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com  
wrote:

 Very nice :)

 On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com  
wrote:


 Folks;
 Here's something that I did playing around w/ learning Apple  
Motion.


Me too: very nice!

-gmaruzz

--
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618

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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_miness...@hotmail.com
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sip:8...@conference.freeswitch.org
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Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video

2009-10-05 Thread Karl Vesterling
NOW THAT might be worth doing!


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

On Oct 5, 2009, at 6:35 PM, Giovanni Maruzzelli wrote:

 The Revenge of the Sip

 On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks jaybi...@gmail.com wrote:
 Haha classic !!!
 Can't wait for the next installment in the series !!

 J


 On 06/10/2009, at 1:02, Anthony Minessale anthony.miness...@gmail.com 
 
 wrote:

 neat,

 Here's some suggestions for your next ones. =p

 Have them standing around the hologram trying to destroy the Death
 Star(tm) that happens to look a lot like a giant 3d unix '*'  
 character.
 Then have one rebel say, wait!, why are we wasting our time... watch
 this... and dial a number on his cellphone as the whole thing  
 explodes in
 the background.

 Have Darth Forkium face Luke ThreadSpawner in a dual.  I see you  
 have
 fashioned your own TDM card vroom..
 Join me and together we can make linked lists and monolithic  
 processes,
 NEVER!... vroom vroom Master Coda has taught you well.You  
 are no
 match for me...JOIN THE ORANGE SIDE OF THE FORCE





 On Mon, Oct 5, 2009 at 2:04 AM, Giovanni Maruzzelli gmar...@celliax.org 
 
 wrote:

 On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola  
 diego.vi...@gmail.com wrote:
 Very nice :)

 On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com
 wrote:

 Folks;
 Here's something that I did playing around w/ learning Apple  
 Motion.

 Me too: very nice!

 -gmaruzz

 --
 Sincerely,

 Giovanni Maruzzelli
 Cell : +39-347-2665618

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 FreeSWITCH http://www.freeswitch.org/
 ClueCon http://www.cluecon.com/
 Twitter: http://twitter.com/FreeSWITCH_wire

 AIM: anthm
 MSN:anthony_miness...@hotmail.com
 GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
 IRC: irc.freenode.net #freeswitch

 FreeSWITCH Developer Conference
 sip:8...@conference.freeswitch.org
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Re: [Freeswitch-users] oz debug says error

2009-10-05 Thread lakshmanan

Openzap.conf.xml

configuration name=openzap.conf description=OpenZAP Configuration
settings

/settings
libpri_spans






/libpri_spans
/configuration

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none

freeswitch.log

http://pastebin.freeswitch.org/10604


Russell.Mosemann wrote:
 
 lakshmanan lakindi...@gmail.com said:
 Thanks for pointing that.
 I also tried that.
 But in that case, I'm not able to make a call through openzap.
 
 What is in openzap.conf.xml? If you start fs_cli and enter oz list,
 what does it show? Copy the ozmod lines from freeswitch.log to
 pastebin.freeswitch.org and post the link here so that we can see what
 openzap does when freeswitch starts.
 
 -- 
 Russell Mosemann
 
 
 
 
 Concordia University, Nebraska
 See http://www.cune.edu/ for the latest news and events!
 
 
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View this message in context: 
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Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] oz debug says error

2009-10-05 Thread lakshmanan ganapathy
Openzap.conf.xml
configuration name=openzap.conf description=OpenZAP Configuration
settings
param name=debug value=7/
/settings
libpri_spans
span name=PRI_1
param name=node value=cpe/
param name=switch value=euroisdn/
param name=dialplan value=XML/
param name=context value=default/
/span
/libpri_spans
/configuration

Output of oz list in fs_cli

span: 1 (PRI_1)
type: isdn
chan_count: 47
dialplan: XML
context: default
dial_regex:
fail_dial_regex:
hold_music:
analog_options none


freeswitch.log

http://pastebin.freeswitch.org/10604





On Mon, Oct 5, 2009 at 6:18 PM, russell.mosem...@cune.org wrote:

 lakshmanan lakindi...@gmail.com said:
  Thanks for pointing that.
  I also tried that.
  But in that case, I'm not able to make a call through openzap.

 What is in openzap.conf.xml? If you start fs_cli and enter oz list,
 what does it show? Copy the ozmod lines from freeswitch.log to
 pastebin.freeswitch.org and post the link here so that we can see what
 openzap does when freeswitch starts.

 --
 Russell Mosemann



 
 Concordia University, Nebraska
 See http://www.cune.edu/ for the latest news and events!


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Re: [Freeswitch-users] Detecting a fax

2009-10-05 Thread Mark Campbell-Smith
Thanks for your help Mike and Tihomir.

A little more playing around and I found that having action
application=fax_detect/ as well as action
application=tone_detect data=fax 1100 r +5000 transfer fax XML
features / do not work together.

Simply by removing fax_detect, the fax is detected beautifully.

My problem now is trying to email the fax.  I followed the
instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
but the dialplan action application=system .../ is not executed
after the rxfax command.  I know the script works because if I put the
system command in another part of the dialplan and hard code the
filename to attach, then the email is sent.

extension name=receivefax
  condition field=destination_number expression=^receivefax$
action application=answer /
action application=playback data=silence_stream://2000/
action application=rxfax
data=//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff/
action application=system
data=/usr/local/freeswitch/scripts/emailfax.sh
/usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff/
action application=hangup/
  /condition
/extension

ideas?
Thanks!

On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga tculj...@gmail.com wrote:
 hi Mark,

 This is an inbound call leg and media channel (so far)  is open in reverse
 direction only (application ringback). I'm afraid you have to answer the
 call to be able to hear the fax tone.

 T.



 On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:

 Fax tones are not played by the remote machine until after answer, the
 tone_detect application starts a media bug that listens for the tone,
 can you confirm the tone is happening at all.  Maybe the issue here is
 the timeout, try making that longer, or doing the tone_detect in
 execute_on_answer

 Mike

 On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:

  Thanks for the response Mike,
 
  I read that page and this one (among others)
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
  I'm still lost.  This is an extract of my dialplan
 
     extension name=Local
       condition field=destination_number expression=^(10[01][0-9])
  $
         action application=set data=dialed_extension=$1/
         action application=export data=dialed_extension=$1/
         action application=set data=ringback=${au-ring}/
         action application=fax_detect/
         action application=tone_detect data=fax 1100 r +5000
  transfer fax XML features /
         action application=set data=hangup_after_bridge=true/
         action application=set data=continue_on_fail=true/
         action application=bridge data=user/${dialed_extensi...@$
  {domain}/
 
  I would assume that on detecting a fax, the dialplan 'fax' is called
  in context features.  This never happens.
 
  When is the fax tone detected?   Is it while the call is ringing or
  can it be detected after the call is answered?  My goal is to be able
  to have the same extension for a voice and fax call.  i assume that
  the fax 'tones' are standardised and the ones on the wiki are correct?
  Also, I guess this doesn't work with media bypass (which I don't
  use).
 
  Thanks!
 
 
  On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
  wrote:
  check out
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
 
  Note, you can't just have tone_detect as your last iten in the
  dialplan as the call will just get hung up.
 
  Mike
 
  On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
 
  Hi
 
  I was hoping someone could help me to setup the fax detection / tone
  detection application.
 
  I want to be able to transfer an incoming fax to a specific
  extension.
  In my default.xml file, I have the following (extracted):
 
     extension name=1000
       condition field=destination_number expression=^(10[01]
  [0-9])
  $
         action application=fax_detect/
         action application=tone_detect data=fax 1100 r +5000
  transfer fax XML features /
 
  I can't get the fax to be detected and transferred.  Is there any
  way
  this can be done?
 
  Thanks!


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Re: [Freeswitch-users] Problem with gateway registration

2009-10-05 Thread Nicolas Brenner
There was no sane way of doing that, so I ended up logging the trace from
the cli.

Here's the bad registration:

- http://pastebin.freeswitch.org/10605

Here's the good one:

- http://pastebin.freeswitch.org/10606

I am not sure if the second one is complete because for some reason the
first few packages don't appear on the console when doing 'sofia profile
external restart reloadxml' and 'sofia profile external siptrace on' or
viceversa.

Anyway, thanks for your time, and I hope those traces help in figuring out
what's going on.


Nicolas


PS: Is there anyway to get the same format from a pcap dump as with the
siptrace feature on the cli?


On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins m...@freeswitch.org wrote:



 On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote:

 Mike, how exactly should I format the file? I got the pcap file, how do I
 convert it to text so that you can easily read it?


 you can open it with wireshark, follow the TCP or UDP stream, then just
 copy  paste the text as needed...

 -MC


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Re: [Freeswitch-users] Wiki Info: Multi Homed Tutorial

2009-10-05 Thread Michael Gende
Michael,

Thanks for wiki-fying my text-only attempt at some user doc. I should have
done that for you. I actually have an updated version with many corrections
and the end tabs filled in. Can you point me to info on how I can amend and
append what you have kindly put up?

Mike G.

On Mon, Oct 5, 2009 at 5:27 PM, Michael Collins m...@freeswitch.org wrote:

 FYI,

 For those who've been following the thread about Michael Gende's tutorial I
 just wanted to let you know that I his document on the wiki. It can be found
 here:

 http://wiki.freeswitch.org/wiki/Multi_home_tutorial

 Please feel free to get in there and try it out, make editorial changes,
 etc. I'm sure you won't hurt Mike's feelings by adding your thoughts. :)

 If you have any questions please reply to this thread and we'll take it
 from there.

 Thanks,
 MC


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Re: [Freeswitch-users] Problem with gateway registration

2009-10-05 Thread Nicolas Brenner
Ignore my previous email, the traces were incomplete, got much better (and
complete) traces with ngrep (found a suggestion from Brian in the list
archive, thanks!)

The gateway that registers:

- http://pastebin.freeswitch.org/10607

The one that doesn't:

- http://pastebin.freeswitch.org/10608


Thanks again for your time and help!


Nicolas


On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner nico...@medularis.comwrote:

 There was no sane way of doing that, so I ended up logging the trace from
 the cli.

 Here's the bad registration:

 - http://pastebin.freeswitch.org/10605

 Here's the good one:

 - http://pastebin.freeswitch.org/10606

 I am not sure if the second one is complete because for some reason the
 first few packages don't appear on the console when doing 'sofia profile
 external restart reloadxml' and 'sofia profile external siptrace on' or
 viceversa.

 Anyway, thanks for your time, and I hope those traces help in figuring out
 what's going on.


 Nicolas


 PS: Is there anyway to get the same format from a pcap dump as with the
 siptrace feature on the cli?


 On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins m...@freeswitch.orgwrote:



 On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote:

 Mike, how exactly should I format the file? I got the pcap file, how do I
 convert it to text so that you can easily read it?


 you can open it with wireshark, follow the TCP or UDP stream, then just
 copy  paste the text as needed...

 -MC


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Re: [Freeswitch-users] Detecting a fax

2009-10-05 Thread Mark Campbell-Smith
Further playing around and everything is working fine (even the
emailing).  I'm not sure what I changed though to document it.

cheers
/M



On Mon, Oct 5, 2009 at 12:03 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
 Hi

 I was hoping someone could help me to setup the fax detection / tone
 detection application.

 I want to be able to transfer an incoming fax to a specific extension.
  In my default.xml file, I have the following (extracted):

    extension name=1000
      condition field=destination_number expression=^(10[01][0-9])$
        action application=fax_detect/
        action application=tone_detect data=fax 1100 r +5000
 transfer fax XML features /

 I can't get the fax to be detected and transferred.  Is there any way
 this can be done?

 Thanks!


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Re: [Freeswitch-users] Detecting a fax

2009-10-05 Thread SP
try using the hanup hook

action application=set data=api_hangup_hook=system ${syscmd}/

On Mon, Oct 5, 2009 at 23:15, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
 Thanks for your help Mike and Tihomir.

 A little more playing around and I found that having action
 application=fax_detect/ as well as action
 application=tone_detect data=fax 1100 r +5000 transfer fax XML
 features / do not work together.

 Simply by removing fax_detect, the fax is detected beautifully.

 My problem now is trying to email the fax.  I followed the
 instructions on the wiki at http://wiki.freeswitch.org/wiki/Mod_fax,
 but the dialplan action application=system .../ is not executed
 after the rxfax command.  I know the script works because if I put the
 system command in another part of the dialplan and hard code the
 filename to attach, then the email is sent.

    extension name=receivefax
      condition field=destination_number expression=^receivefax$
        action application=answer /
        action application=playback data=silence_stream://2000/
        action application=rxfax
 data=//usr//local//freeswitch//storage//${caller_id_number}-${uuid}.rxfax.tiff/
        action application=system
 data=/usr/local/freeswitch/scripts/emailfax.sh
 /usr/local/freeswitch/storage/${caller_id_number}-${uuid}.rxfax.tiff/
        action application=hangup/
      /condition
    /extension

 ideas?
 Thanks!

 On Tue, Oct 6, 2009 at 1:32 AM, Tihomir Culjaga tculj...@gmail.com wrote:
 hi Mark,

 This is an inbound call leg and media channel (so far)  is open in reverse
 direction only (application ringback). I'm afraid you have to answer the
 call to be able to hear the fax tone.

 T.



 On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:

 Fax tones are not played by the remote machine until after answer, the
 tone_detect application starts a media bug that listens for the tone,
 can you confirm the tone is happening at all.  Maybe the issue here is
 the timeout, try making that longer, or doing the tone_detect in
 execute_on_answer

 Mike

 On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:

  Thanks for the response Mike,
 
  I read that page and this one (among others)
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
  I'm still lost.  This is an extract of my dialplan
 
     extension name=Local
       condition field=destination_number expression=^(10[01][0-9])
  $
         action application=set data=dialed_extension=$1/
         action application=export data=dialed_extension=$1/
         action application=set data=ringback=${au-ring}/
         action application=fax_detect/
         action application=tone_detect data=fax 1100 r +5000
  transfer fax XML features /
         action application=set data=hangup_after_bridge=true/
         action application=set data=continue_on_fail=true/
         action application=bridge data=user/${dialed_extensi...@$
  {domain}/
 
  I would assume that on detecting a fax, the dialplan 'fax' is called
  in context features.  This never happens.
 
  When is the fax tone detected?   Is it while the call is ringing or
  can it be detected after the call is answered?  My goal is to be able
  to have the same extension for a voice and fax call.  i assume that
  the fax 'tones' are standardised and the ones on the wiki are correct?
  Also, I guess this doesn't work with media bypass (which I don't
  use).
 
  Thanks!
 
 
  On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
  wrote:
  check out
  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
 
  Note, you can't just have tone_detect as your last iten in the
  dialplan as the call will just get hung up.
 
  Mike
 
  On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
 
  Hi
 
  I was hoping someone could help me to setup the fax detection / tone
  detection application.
 
  I want to be able to transfer an incoming fax to a specific
  extension.
  In my default.xml file, I have the following (extracted):
 
     extension name=1000
       condition field=destination_number expression=^(10[01]
  [0-9])
  $
         action application=fax_detect/
         action application=tone_detect data=fax 1100 r +5000
  transfer fax XML features /
 
  I can't get the fax to be detected and transferred.  Is there any
  way
  this can be done?
 
  Thanks!


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