[Freeswitch-users] How to modify the Subject and Body when sending voicemail??

2009-07-07 Thread Brad Tuan
As title, How to custom the Subject and Body and ... of the mail ??
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Re: [Freeswitch-users] How to modify the Subject and Body when sending voicemail??

2009-07-07 Thread Jason White
Brad Tuan brad.t...@gmail.com wrote:
 As title, How to custom the Subject and Body and ... of the mail ??

Have a look at the notify-voicemail.tpl  and voicemail.tpl files, and the
template parameters in voicemail.conf.xml in the default FreeSWITCH
configuration to see how it all works and to decide what to edit.


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[Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback

2009-07-07 Thread seven

# play test.wav

Input File : 'test.wav'
Sample Size: 16-bit (2 bytes)
Sample Encoding: signed (2's complement)
Channels   : 2
Sample Rate: 16000


1) is it the default behavior that uuid_record record with 2 channels
2) is it reasonable that FS can record to 2 channels but cannot  
playback?

3) do I need to set RECORD_STEREO=false before uuid_record?


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Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback

2009-07-07 Thread Jason White
seven dujinf...@gmail.com wrote:

 1) is it the default behavior that uuid_record record with 2 channels

Yes.
 2) is it reasonable that FS can record to 2 channels but cannot  
 playback?

Could you explain what happens when you play back the files?
 3) do I need to set RECORD_STEREO=false before uuid_record?

That depends on whether you want two-channel output files or not.


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Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback

2009-07-07 Thread seven

 2) is it reasonable that FS can record to 2 channels but cannot
 playback?

 Could you explain what happens when you play back the files?

yes, it's here: http://pastebin.freeswitch.org/9641


 3) do I need to set RECORD_STEREO=false before uuid_record?

 That depends on whether you want two-channel output files or not.


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Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback

2009-07-07 Thread Jason White
seven dujinf...@gmail.com wrote:
 
 yes, it's here: http://pastebin.freeswitch.org/9641

Judging by the error message, it's a known limitation. You are welcome to work
on a fix, or pay the develoeprs to fix it, or offer a bounty that might
encourage someone to work on it, or wait until it gets  fixed.

Meanwhile, convert the file to mono and try again.

Sox should be able to do this, for example.


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Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback

2009-07-07 Thread seven

On Jul 7, 2009, at 3:56 PM, Jason White wrote:

 seven dujinf...@gmail.com wrote:

 yes, it's here: http://pastebin.freeswitch.org/9641

 Judging by the error message, it's a known limitation. You are  
 welcome to work
 on a fix, or pay the develoeprs to fix it, or offer a bounty that  
 might
 encourage someone to work on it, or wait until it gets  fixed.

 Meanwhile, convert the file to mono and try again.

 Sox should be able to do this, for example.



Thanks, I'm using sox.



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[Freeswitch-users] Record_session cutting wav files

2009-07-07 Thread Mariusz Kołodziejczyk
Hello

I saw a strange behavior when i'm using record_session for outbound 
call. Recorded file is 24:20 time length, but in logs should have about 
25:18.

Here ma log:

start recording about 2009-07-07 10:56:15
2009-07-07 10:56:15.108447 [DEBUG] mod_dptools.c:748 sofia/internal/1062 
SET [czas]=[2009-07-07-10-56-15] - variable $czas = 
2009-07-07-10-56-15 i use it in filename below
EXECUTE sofia/internal/1062 
record_session(/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135X.2009-07-07-10-56-15.out.ISDN.wav)
.
stop recording
2009-07-07 11:21:33.291177 [DEBUG] switch_ivr_async.c:444 Stop recording 
file 
/records/2009-07-07/069630d2-6ad4-11de-aa6e-e31e1f9431c0.10XX.07135X.2009-07-07-10-56-15.out.ISDN.wav

file should have about: 25:18 time length
when i listen a file it's really cut

My piece of dialplan:

condition field=destination_number expression=^(\d{4,10})$

  action application=set data=RECORD_ANSWER_REQ=false/

!-- dir for record files --
  action application=set data=dir=${strftime(%Y-%m-%d)}/
   
!-- create dir if doesnt exists--
  action application=system data=mkdir -p /records/${dir}/

!-- polish ringback --
  action application=set data=ringback=${pl-ring}/

!-- setting var $czas --
  action application=set data=czas=${strftime(%Y-%m-%d-%H-%M-%S)}/

!-- start record session --
  action application=record_session 
data=/records/${dir}/${uuid}.${caller_id_number}.$1.${czas}.out.ISDN.wav/

 !-- bridge channels --
  action application=bridge 
data={origination_caller_id_number=${cti_gateway_number},effective_caller_id_number=${cti_gateway_number}}openzap/1/A/$1/
/condition

freeswitch version: FreeSWITCH Version 1.0.trunk (14013)
linux: Linux pbx2 2.6.26-1-686 #1 SMP Fri Mar 13 18:08:45 UTC 2009 i686 
GNU/Linux

Thanks

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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-07 Thread Jason Garland
I have FreeSwitch running on a $50 Linksys NSLU2

Sent from my iPhone

On Jul 6, 2009, at 5:23 AM, David Goodenough david.goodeno...@linkchoose.co.uk 
  wrote:

 On Monday 06 July 2009, Fred-145 wrote:
 EdPimentl wrote:
 Back in April I posted these links on the list, in regards to a  
 similar
 question

 Thanks Ed, but the problem with all those, is:
 - they typically have so little RAM/Flash RAM that they can't run a  
 regular
 Linux distro, which means that we're stuck with whatever software is
 available with the customized distro for the appliance
 - they don't have room for a PCI card, which means that we have to  
 have an
 external VoIP gateway to connect the appliance to the POTS
 - they're as expensive or more expensive than a regular PC

 At this point, there doesn't seem to be any appliance that can run
 FreeSwitch and handle a POTS line  in a compact, sub-$200 price- 
 range.

 I though that I had read somewhere about someone using the Marvel
 ShevaPlug and a 2 line USB POTS adapter.  That should be under $200.

 David

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Re: [Freeswitch-users] How to modify my INVITE msg??

2009-07-07 Thread Brian West
Well put the right profile name in there... instead of just profile

/b

On Jul 6, 2009, at 9:48 PM, Brad Tuan wrote:

 2009-07-07 10:38:46 [ERR] mod_sofia.c:2681 sofia_outgoing_channel()  
 Invalid Profile

 FS return it is a Invalid ProfileWhy??
 __


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Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback

2009-07-07 Thread Brian West
OK let me comment ion this.  Your voip connection is a single channel  
mono.  The recording is two channel stereo.. the caller is in the left  
and the callee is in the right. This is a very helpful tool for call  
centers when your agent gets into a fight with the caller.  FreeSWITCH  
can not shove a stereo signal down a mono line.  Its rather obvious  
that it has to mux it into one channel... which it does... the other  
alternative is to just hang up the call and say WOOPS can't do it.


/b

On Jul 7, 2009, at 2:56 AM, Jason White wrote:


seven dujinf...@gmail.com wrote:


yes, it's here: http://pastebin.freeswitch.org/9641


Judging by the error message, it's a known limitation. You are  
welcome to work
on a fix, or pay the develoeprs to fix it, or offer a bounty that  
might

encourage someone to work on it, or wait until it gets  fixed.

Meanwhile, convert the file to mono and try again.

Sox should be able to do this, for example.


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[Freeswitch-users] freeswitch sipnet.ru caller id

2009-07-07 Thread Alexey Lubimov
Good day.

I have external sip gateway on sipnet.ru

Logs from sipnet.ru contain my ip address  (193.112.5.111) instead 
actual number (1...@193.112.5.111) in field caller id.

Is it possible to set caller id in actual number instead ip address?
 


07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14 
0.01750 0.00408

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Re: [Freeswitch-users] freeswitch sipnet.ru caller id

2009-07-07 Thread Brian West

I would need to see the sip packets to know why its doing that.

/b

On Jul 7, 2009, at 8:53 AM, Alexey Lubimov wrote:


Good day.

I have external sip gateway on sipnet.ru

Logs from sipnet.ru contain my ip address  (193.112.5.111) instead
actual number (1...@193.112.5.111) in field caller id.

Is it possible to set caller id in actual number instead ip address?



07/07/09 14:08 74959345610 193.112.5.111 Russia Moscow 00381982 0:14
0.01750 0.00408


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[Freeswitch-users] 2 voicemail questions

2009-07-07 Thread Mark Campbell-Smith
Hi!

I have 2 questions regarding voicemail ...

1. Can I email the voicemail message to multiple email addresses?  If
so, what format is this in?
  param name=vm-mailto value=m...@myemail.com/

2. How can I make Freeswitch dial a number AFTER a voicemail is left?

Thanks!

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[Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Max Bridgewater
Hi Guys,

I keep getting this message printed in red on my consolde; and whenevr i get
it, DTMF will stop being trasmitting in that session.

[ERR] switch_rtp.c:2013 Failed DTMF sanity check.

What does that mean and how can i prevent this from occurring?  I'm using
the socket API to send DTMF signals to freeswitch.

Thanks,
Max.
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[Freeswitch-users] Follow FreeSWITCH_wire on Twitter!

2009-07-07 Thread Michael Collins
Okay everyone, spread the word: we have a Twitter channel for everyone to
follow. It's called FreeSWITCH_wire and it's were you'll see up-to-the
minute updates on FreeSWITCH development, new releases, important news and
the like.

Everyone go follow FreeSWITCH_wire right now, and don't forget to tell
everyone in the VoIP world that they need to follow it as well!

-Michael
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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Max Bridgewater
Actually I'm using Voxeo to generate DTMFs. They have the following
construct that allows me to play DTMF:

audio src=dtmf:4367#/audio

I think it's not standard VXML. How can i track this easily or at least
capture the RTP stream so i can send it to them?

Max.



On Tue, Jul 7, 2009 at 1:55 PM, Brian West br...@freeswitch.org wrote:

 That usually means your device you're using is broken for sending
 rfc2833 can you tell me what device are you using?

 /b

 On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote:

  Hi Guys,
 
  I keep getting this message printed in red on my consolde; and
  whenevr i get it, DTMF will stop being trasmitting in that session.
 
  [ERR] switch_rtp.c:2013 Failed DTMF sanity check.
 
  What does that mean and how can i prevent this from occurring?  I'm
  using the socket API to send DTMF signals to freeswitch.
 
  Thanks,
  Max.


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Michael Collins
There are some troubleshooting tips here:
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies

There are several ways of capturing packets on your system and the above
link explains how to set them up.
-MC

On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater
max.bridgewa...@gmail.comwrote:

 Actually I'm using Voxeo to generate DTMFs. They have the following
 construct that allows me to play DTMF:

 audio src=dtmf:4367#/audio

 I think it's not standard VXML. How can i track this easily or at least
 capture the RTP stream so i can send it to them?

 Max.



 On Tue, Jul 7, 2009 at 1:55 PM, Brian West br...@freeswitch.org wrote:

 That usually means your device you're using is broken for sending
 rfc2833 can you tell me what device are you using?

 /b

 On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote:

  Hi Guys,
 
  I keep getting this message printed in red on my consolde; and
  whenevr i get it, DTMF will stop being trasmitting in that session.
 
  [ERR] switch_rtp.c:2013 Failed DTMF sanity check.
 
  What does that mean and how can i prevent this from occurring?  I'm
  using the socket API to send DTMF signals to freeswitch.
 
  Thanks,
  Max.


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Brian West
Can you get me an RTP trace bet they might be doing it wrong... seems  
to be common.

/b

On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote:

 Actually I'm using Voxeo to generate DTMFs. They have the following  
 construct that allows me to play DTMF:

 audio src=dtmf:4367#/audio

 I think it's not standard VXML. How can i track this easily or at  
 least capture the RTP stream so i can send it to them?

 Max.


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Max Bridgewater
 Can you get me an RTP trace bet they might be doing it wrong... seems
 to be common.



Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to
run Something like Wireshark?

Pascal.


 /b

 On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote:

  Actually I'm using Voxeo to generate DTMFs. They have the following
  construct that allows me to play DTMF:
 
  audio src=dtmf:4367#/audio
 
  I think it's not standard VXML. How can i track this easily or at
  least capture the RTP stream so i can send it to them?
 
  Max.


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Michael Collins
You'll need Wireshark or similar. Some tips can be found here:
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies

-MC

On Tue, Jul 7, 2009 at 11:19 AM, Max Bridgewater
max.bridgewa...@gmail.comwrote:


 Can you get me an RTP trace bet they might be doing it wrong... seems
 to be common.



 Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to
 run Something like Wireshark?

 Pascal.


 /b

 On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote:

  Actually I'm using Voxeo to generate DTMFs. They have the following
  construct that allows me to play DTMF:
 
  audio src=dtmf:4367#/audio
 
  I think it's not standard VXML. How can i track this easily or at
  least capture the RTP stream so i can send it to them?
 
  Max.


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Brian West
email it directly to me off list please.

/b

On Jul 7, 2009, at 1:48 PM, Max Bridgewater wrote:

 Thanks! What is the best way to send you the 4M pcap file?


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Michael Collins
Put it out on a webserver where one of the devs can grab it with a browser.
-MC

On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater
max.bridgewa...@gmail.comwrote:

 Thanks! What is the best way to send you the 4M pcap file?


 On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins m...@freeswitch.orgwrote:

 There are some troubleshooting tips here:

 http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies

 There are several ways of capturing packets on your system and the above
 link explains how to set them up.
 -MC


 On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater 
 max.bridgewa...@gmail.com wrote:

 Actually I'm using Voxeo to generate DTMFs. They have the following
 construct that allows me to play DTMF:

 audio src=dtmf:4367#/audio

 I think it's not standard VXML. How can i track this easily or at least
 capture the RTP stream so i can send it to them?

 Max.



 On Tue, Jul 7, 2009 at 1:55 PM, Brian West br...@freeswitch.org wrote:

 That usually means your device you're using is broken for sending
 rfc2833 can you tell me what device are you using?

 /b

 On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote:

  Hi Guys,
 
  I keep getting this message printed in red on my consolde; and
  whenevr i get it, DTMF will stop being trasmitting in that session.
 
  [ERR] switch_rtp.c:2013 Failed DTMF sanity check.
 
  What does that mean and how can i prevent this from occurring?  I'm
  using the socket API to send DTMF signals to freeswitch.
 
  Thanks,
  Max.


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Re: [Freeswitch-users] Failed DTMF Sanity check

2009-07-07 Thread Tamas
http://filebin.ca/
(up to 50MB)

Tamas

Michael Collins írta:
 Put it out on a webserver where one of the devs can grab it with a
 browser.
 -MC

 On Tue, Jul 7, 2009 at 11:48 AM, Max Bridgewater
 max.bridgewa...@gmail.com mailto:max.bridgewa...@gmail.com wrote:

 Thanks! What is the best way to send you the 4M pcap file?


 On Tue, Jul 7, 2009 at 2:15 PM, Michael Collins
 m...@freeswitch.org mailto:m...@freeswitch.org wrote:

 There are some troubleshooting tips here:

 
 http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Specific_Technologies

 There are several ways of capturing packets on your system and
 the above link explains how to set them up.
 -MC


 On Tue, Jul 7, 2009 at 11:07 AM, Max Bridgewater
 max.bridgewa...@gmail.com mailto:max.bridgewa...@gmail.com
 wrote:

 Actually I'm using Voxeo to generate DTMFs. They have the
 following construct that allows me to play DTMF:

 audio src=dtmf:4367#/audio

 I think it's not standard VXML. How can i track this
 easily or at least capture the RTP stream so i can send it
 to them?

 Max.



 On Tue, Jul 7, 2009 at 1:55 PM, Brian West
 br...@freeswitch.org mailto:br...@freeswitch.org wrote:

 That usually means your device you're using is broken
 for sending
 rfc2833 can you tell me what device are you using?

 /b

 On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote:

  Hi Guys,
 
  I keep getting this message printed in red on my
 consolde; and
  whenevr i get it, DTMF will stop being trasmitting
 in that session.
 
  [ERR] switch_rtp.c:2013 Failed DTMF sanity check.
 
  What does that mean and how can i prevent this from
 occurring?  I'm
  using the socket API to send DTMF signals to freeswitch.
 
  Thanks,
  Max.


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[Freeswitch-users] ClueCon 2009 Updates - July 7, 2009

2009-07-07 Thread Michael Collins
Hello folks!

We have a few more updates. First of all, if you haven't already heard,
we've extended the early bird sign up to go through July 21. That's only two
weeks away, so if you haven't already registered then please call us at
877.742.CLUE and we'll get you set up. Secondly, there are some updates on
the ClueCon blog:
http://cluecon.com/blog/1

The breakfast and lunch menus have been posted. (Subject to change, of
course. :) Also, we have a synopsis for Irv Shapiro's talk, entitled Cloud
Telephony on the latest blog post.

Can't wait to see you all in Chicago!
-Michael Collins
http://www.cluecon.com
877.742.CLUE
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Re: [Freeswitch-users] Controlling MOH from a java application

2009-07-07 Thread Anthony Minessale
FSAPI commands are accessed via the API obj

api = new API();

api.execute(uuid_hold, session.get_uuid());
...
api.execute(uuid_hold, off  + session.get_uuid());



On Mon, Jul 6, 2009 at 10:21 PM, Jean-Marc Hyppolite
hyppolit...@yahoo.comwrote:

 Hello Brian,

 Thank you for your quick answer. I tried the two API functions but with no
 result. The caller is not able to hear any music. But, when I use two
 extensions (one calling the other), MOH does work.

 My code on the JAVA side (for test purposes)

 session.answer();
 session.sleep(500);

 session.execute(eval, uuid_hold  + session.get_uuid());

 java_function(); // lasts 30 to 40 seconds

 session.execute(eval, uuid_hold off  + session.get_uuid());

 session.sleep(500);
 session.hangup();

 Thanks for the help.

 Jean-Marc.




 --- On *Mon, 7/6/09, Brian West br...@freeswitch.org* wrote:


 From: Brian West br...@freeswitch.org
 Subject: Re: [Freeswitch-users] Controlling MOH from a java application
 To: freeswitch-users@lists.freeswitch.org
 Received: Monday, July 6, 2009, 7:27 PM


 uuid_hold uuid
 uuid_hold off uuid

 These two api's will do it.

 /b



  On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote:

Hello,

 First of all, I would like to thank Anthony, Brian and all the developers
 for this wonderful piece of software. Very good job.

 I would like to know how I can start and stop Music On Hold from a JAVA
 script (using mod_java) similar to the StartMusicOnHold and StopMusicHold
 functions found in AGI (Asterisk-Java).

 I am using FreeSWITCH as an IVR server. I would like to be able to put the
 caller on hold while doing some other stuff.

 Thanks in advance.

 Jean-Marc.


  --

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[Freeswitch-users] Leaking stream handle

2009-07-07 Thread Kees Varekamp
I am testing Freeswitch as an alternative to Asterisk. So far, so good,
except for the following:


   - I have a lua channel listening to:
  - session:streamFile('local_stream://moh')
   - I have a socket bridging this channel to a sip gateway:
  - SendMsg 1e0cc726-6b33-11de-bae1-5fd843059ad5
  - call-command: execute
  - execute-app-name: bridge
  - execute-app-arg: sofia/gateway/somegateway/somenumber
   - This all works well, but the console says:
  - [CRIT] mod_local_stream.c:234 Leaking stream handle!
  [switch_ivr_play_file() src/switch_ivr_play_say.c:1026]


Is this something I should be worried about? Thanks
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[Freeswitch-users] Clustering Freeswitch

2009-07-07 Thread Andres Gomez
Hello to all

Exist any solution to clustering. Any load balancing appliance o heartbeat
test?.

Regards

Andres Gomez.
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Re: [Freeswitch-users] Clustering Freeswitch

2009-07-07 Thread SP
openser/kamailio/ser/opensips you pick a name

DNS SRV

On Tue, Jul 7, 2009 at 16:29, Andres Gomez vladisl...@gmail.com wrote:

 Hello to all

 Exist any solution to clustering. Any load balancing appliance o heartbeat
 test?.

 Regards

 Andres Gomez.

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Re: [Freeswitch-users] Clustering Freeswitch

2009-07-07 Thread Shelby Ramsey
Andres,

OpenSIP's works very well as a load balancer.  You could also use DNS SRV
(if the clients support it), round robin DNS 

Really just depends on what you are trying to accomplish.

SDR
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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-07 Thread Raul Fragoso
I'm about to order one of these eBox systems:
http://www.wdlsystems.com/modperl/view_services.cgi?r=detailprod_num=1EBX33Jaisle_id=1073

They sell for under $200, have decent specs and some models come with a
mini-PCI slot, which can be used to attach a POTS card. 

Regards,

Raul

On Sun, 2009-07-05 at 18:22 -0700, Fred-145 wrote:
 
 EdPimentl wrote:
  Back in April I posted these links on the list, in regards to a similar
  question
 
 Thanks Ed, but the problem with all those, is:
 - they typically have so little RAM/Flash RAM that they can't run a regular
 Linux distro, which means that we're stuck with whatever software is
 available with the customized distro for the appliance
 - they don't have room for a PCI card, which means that we have to have an
 external VoIP gateway to connect the appliance to the POTS
 - they're as expensive or more expensive than a regular PC
 
 At this point, there doesn't seem to be any appliance that can run
 FreeSwitch and handle a POTS line  in a compact, sub-$200 price-range.


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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-07 Thread Brian West
I'll take two please!  ;)

/b

On Jul 7, 2009, at 8:14 PM, Raul Fragoso wrote:

 I'm about to order one of these eBox systems:
 http://www.wdlsystems.com/modperl/view_services.cgi?r=detailprod_num=1EBX33Jaisle_id=1073

 They sell for under $200, have decent specs and some models come  
 with a
 mini-PCI slot, which can be used to attach a POTS card.

 Regards,

 Raul

 On Sun, 2009-07-05 at 18:22 -0700, Fred-145 wrote:

 EdPimentl wrote:
 Back in April I posted these links on the list, in regards to a  
 similar
 question

 Thanks Ed, but the problem with all those, is:
 - they typically have so little RAM/Flash RAM that they can't run a  
 regular
 Linux distro, which means that we're stuck with whatever software is
 available with the customized distro for the appliance
 - they don't have room for a PCI card, which means that we have to  
 have an
 external VoIP gateway to connect the appliance to the POTS
 - they're as expensive or more expensive than a regular PC

 At this point, there doesn't seem to be any appliance that can run
 FreeSwitch and handle a POTS line  in a compact, sub-$200 price- 
 range.


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Re: [Freeswitch-users] Baby Update!

2009-07-07 Thread Raymond Chandler

Brian West wrote:

Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz

YAY... Congrats mr Lanman!

/b

THANKS BRIAN

And the few others that sent money.  The hospital is a little over an 
hour drive, so the money definitely helped out with gas, etc... You 
really are a great community...


-Ray
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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-07 Thread Hadley Rich
On Sat, 2009-07-04 at 15:18 -0500, Brian West wrote:
 I use one of the intel atom boxes at home.

These guys have a case/PCI-riser for an Intel Atom board which would
make a nice little appliance.

http://www.mini-box.com/I-O-shield-and-riser-card-for-D945GSEJT

hads
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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-07 Thread Brian West

I'll take two of those too!  :)

/b

On Jul 7, 2009, at 8:52 PM, Hadley Rich wrote:


On Sat, 2009-07-04 at 15:18 -0500, Brian West wrote:

I use one of the intel atom boxes at home.


These guys have a case/PCI-riser for an Intel Atom board which would
make a nice little appliance.

http://www.mini-box.com/I-O-shield-and-riser-card-for-D945GSEJT

hads
--
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New Zealand's Open Source Hardware Supplier



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Re: [Freeswitch-users] Controlling MOH from a java application

2009-07-07 Thread Jean-Marc Hyppolite
Hello,
 
Thank you Anthony. 
 
My problem now is when the call is put on hold, no music is heard from the 
caller.
 
Thank you again.
 
Jean-Marc.

--- On Tue, 7/7/09, Anthony Minessale anthony.miness...@gmail.com wrote:


From: Anthony Minessale anthony.miness...@gmail.com
Subject: Re: [Freeswitch-users] Controlling MOH from a java application
To: freeswitch-users@lists.freeswitch.org
Received: Tuesday, July 7, 2009, 4:37 PM


FSAPI commands are accessed via the API obj

api = new API();

api.execute(uuid_hold, session.get_uuid());
...
api.execute(uuid_hold, off  + session.get_uuid());




On Mon, Jul 6, 2009 at 10:21 PM, Jean-Marc Hyppolite hyppolit...@yahoo.com 
wrote:






Hello Brian,
 
Thank you for your quick answer. I tried the two API functions but with no 
result. The caller is not able to hear any music. But, when I use two 
extensions (one calling the other), MOH does work.

 
My code on the JAVA side (for test purposes)
 
session.answer();
session.sleep(500);
 
session.execute(eval, uuid_hold  + session.get_uuid());
 
java_function(); // lasts 30 to 40 seconds
 
session.execute(eval, uuid_hold off  + session.get_uuid());
 
session.sleep(500);
session.hangup();
 
Thanks for the help.
 
Jean-Marc.
 
 
 

--- On Mon, 7/6/09, Brian West br...@freeswitch.org wrote:


From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] Controlling MOH from a java application
To: freeswitch-users@lists.freeswitch.org
Received: Monday, July 6, 2009, 7:27 PM





uuid_hold uuid 


uuid_hold off uuid


These two api's will do it.


/b







On Jul 6, 2009, at 6:20 PM, Jean-Marc Hyppolite wrote:






Hello,
 
First of all, I would like to thank Anthony, Brian and all the developers for 
this wonderful piece of software. Very good job.
 
I would like to know how I can start and stop Music On Hold from a JAVA script 
(using mod_java) similar to the StartMusicOnHold and StopMusicHold functions 
found in AGI (Asterisk-Java).
 
I am using FreeSWITCH as an IVR server. I would like to be able to put the 
caller on hold while doing some other stuff. 
 
Thanks in advance.
 
Jean-Marc.






Yahoo! Canada Toolbar : Search from anywhere on the web and bookmark your 
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ClueCon http://www.cluecon.com/

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Re: [Freeswitch-users] Controlling MOH from a java application

2009-07-07 Thread Brian West

Can you open a jira please.

/b

On Jul 7, 2009, at 9:22 PM, Jean-Marc Hyppolite wrote:


Hello,

Thank you Anthony.

My problem now is when the call is put on hold, no music is heard  
from the caller.


Thank you again.

Jean-Marc.


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Re: [Freeswitch-users] Cheapest, most compact FreeSwitch appliance?

2009-07-07 Thread Craig Askings
Does the mini-pci slot have external access to route the POTS cable?

2009/7/8 Raul Fragoso r...@etellicom.com:
 I'm about to order one of these eBox systems:
 http://www.wdlsystems.com/modperl/view_services.cgi?r=detailprod_num=1EBX33Jaisle_id=1073

 They sell for under $200, have decent specs and some models come with a
 mini-PCI slot, which can be used to attach a POTS card.

 Regards,

 Raul
-- 
Craig Askings

Network Engineer | Over the Wire Pty Ltd
cr...@overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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[Freeswitch-users] Force SUBSCRIBE or sendevent NOTIFY without subscription

2009-07-07 Thread Nick Lemberger
Is it possible to force a sofia profile to subscribe to an event or use 
sendevent to force send a NOTIFY to a SIP endpoint?

I'm trying to use FreeSwitch as a voicemail server but the sending switch 
doesn't send SIP SUBSCRIBE messages.  I'd like to send unsolicited SIP notifies 
to turn on MWI indicators as that's what the switch expects.

Any ideas, or is this even possible with FreeSwitch?

Thanks,
Nick

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Re: [Freeswitch-users] Controlling MOH from a java application

2009-07-07 Thread Jean-Marc Hyppolite
Thank you Brian.

--- On Tue, 7/7/09, Brian West br...@freeswitch.org wrote:


From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] Controlling MOH from a java application
To: freeswitch-users@lists.freeswitch.org
Received: Tuesday, July 7, 2009, 10:28 PM


Can you open a jira please.


/b



On Jul 7, 2009, at 9:22 PM, Jean-Marc Hyppolite wrote:






Hello,
 
Thank you Anthony.
 
My problem now is when the call is put on hold, no music is heard from the 
caller.
 
Thank you again.
 
Jean-Marc.

-Inline Attachment Follows-


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Re: [Freeswitch-users] Force SUBSCRIBE or sendevent NOTIFY without subscription

2009-07-07 Thread Brian West
Yes you can

sendevent NOTIFY

Here is the headers you'll need.. some are optional


const char *profile_name = switch_event_get_header(event, profile);
const char *ct = switch_event_get_header(event, content-type);
const char *es = switch_event_get_header(event, event-string);
const char *user = switch_event_get_header(event, user);
const char *host = switch_event_get_header(event, host);
const char *call_id = switch_event_get_header(event, call-id);
const char *uuid = switch_event_get_header(event, uuid);
const char *body = switch_event_get_body(event);
const char *to_uri = switch_event_get_header(event, to-uri);
const char *from_uri = switch_event_get_header(event, from-uri);


See mod_sofia.c line 2887

/b
PS: you also have SEND_MESSAGE as an event you can send below that in  
mod_sofia.c



On Jul 7, 2009, at 9:46 PM, Nick Lemberger wrote:

 Is it possible to force a sofia profile to subscribe to an event or  
 use sendevent to force send a NOTIFY to a SIP endpoint?

 I'm trying to use FreeSwitch as a voicemail server but the sending  
 switch doesn't send SIP SUBSCRIBE messages.  I'd like to send  
 unsolicited SIP notifies to turn on MWI indicators as that's what  
 the switch expects.

 Any ideas, or is this even possible with FreeSwitch?

 Thanks,
 Nick


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Re: [Freeswitch-users] How to get the hook state?

2009-07-07 Thread Yehavi Bourvine
Hello,

  The problem we are trying to solve here is handling a busy state according
to the user's prefference (some want a busy to be heard, some want the call
to go to voicemail, and some want to get the second call).

  The first step is finding that an extension is busy. It would be nice in
the future to know also other states of an extension (like - not registered,
etc.).

  Thanks, __Yehavi:

2009/7/7 Brian West br...@freeswitch.org

 What are you trying to accomplish?

 /b

 On Jul 6, 2009, at 11:53 PM, Eli Hayun wrote:

  Hi
  I am a newbie in FreeSwitch and my question is:
  When I am calling to an extension, how should I know in advance what
  is
  the hook status. I tried to find out a variable that can get me this
  information but with no success.
  any help?


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