Re: [Freeswitch-users] FS Slide deck?

2009-10-09 Thread Michael Collins
On Thu, Oct 8, 2009 at 10:38 PM, Gabriel Gunderson g...@gundy.org wrote:

 On Thu, Oct 8, 2009 at 4:38 PM, Michael Collins m...@freeswitch.org
 wrote:
  On Thu, Oct 8, 2009 at 12:01 PM, Gabriel Gunderson g...@gundy.org
 wrote:
  Naturally the wireless is jacked at the conf.  It's no match for the
  100's of geeks sporting laptops.  Thanks for the link. Looks like
  we'll be shooting from the hip. :)
 
  We experienced something similar at ClueCon, but it was that the hotel's
  DHCP server was a total wuss. :)

 So, I just got back from the first day of the conference and there was
 plenty of interest in FreeSWITCH from all the Asterisk users.  We had
 a good time and answered lots of questions.  It was interesting to
 hear what their perceptions about FS were.


Thanks for reporting back. Please let all the Asterisk users know that they
are welcome to join us in #freeswitch on irc.freenode.net and that they will
not be abused like people do in other less friendly IRC channels.

-MC



 Gabe

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Re: [Freeswitch-users] mod_fax compile fails

2009-10-09 Thread Jason White
Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
 Thanks Rob,
 
 Is this a fault in the svn update process?  

Usually, no, it's rather that if you build in a directory that you've checked
out and then run svn update, not everything gets cleaned up properly the next
time you build.

I find it's faster to build in a tmpfs file system anyway, e.g.,

mkdir /tmp/fs  sudo mount -t tmpfs tmpfs /tmpfs  svn export . /tmp/fs
then go into /tmp/fs and run the compilation process, or the package building
command, or whatever you need.

This way, my original svn tree is never altered and it will always update
cleanly.


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Re: [Freeswitch-users] FS Slide deck?

2009-10-09 Thread Gabriel Gunderson
On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins m...@freeswitch.org wrote:
 Thanks for reporting back. Please let all the Asterisk users know that they
 are welcome to join us in #freeswitch on irc.freenode.net and that they will
 not be abused like people do in other less friendly IRC channels.

Funny you mention this.  Many people report that the way the FS
community refers to Asterisk in docs/wikis/irc/whatever makes the FS
camp seem *less* welcoming to them.  After all, they identify as
Asterisk Users and take the criticism as being kinda harsh.  Most of
them acknowledge the shortcomings of Asterisk but are put off when
someone else points them out.  It's crazy, I know.  The thing is, I
remember thinking that too.  After getting to know FS better, I didn't
notice it as much.  Nobody likes to hear their baby is ugly --even if
they know it is.

At our session, and in general, I've noticed people are more
interested in hearing about FS when you don't make direct comparisons
to Asterisk.  Besides, FS stands on it's own merit.

Just what I've observed *and* my 2 additional cents.

Gabe

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Re: [Freeswitch-users] mod_opal - call charged before H.225 connect

2009-10-09 Thread Gabriel Gunderson
Doesn't anyone trim on this list?

Gabe

P.S.  Sorry about the top post, but I was worried that if I bottom
posted, nobody would find it.


2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru:
 On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote 
 freeswitch-us...@lists.fre...:

 TCHi Yuriy,
 TC
 TCcan you share what you have so far, I'm sure we can help with RTP part...

 I think there is a few days and i make it work, after this i start to test 
 and share it.

 TC
 TCT.
 TC
 TC2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru
 TC
 TC On 2009-10-08 13:25 -0400, Tuyan ?zipek wrote
 TC freeswitch-us...@lists.freesw...:
 TC
 TC TzHi,
 TC Tz
 TC Tz2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru:
 TC Tz On 2009-10-08 10:43 -0500, Anthony Minessale wrote
 TC freeswitch-us...@lists.f...:
 TC Tz
 TC Tz AMIf you are going to make that alternate module are you going to
 TC host it in
 TC Tz AMthe FS tree along side mod_opal?
 TC Tz
 TC Tz Yes, but then it be useful, now i have working only signaling part 
 and
 TC some
 TC Tz kinde of not working rtp part :)
 TC Tz
 TC TzIf you dont use fs rtp stack, its unlikely that it will be accepted
 TC Tzinto the tree.
 TC Tz
 TC Tz
 TC Tz AMalso if were working on mod_opal why did you not try to involve 
 us
 TC and the
 TC Tz AMopal team?
 TC Tz
 TC Tz Because i made patches for libopal, one is a bugfix in rtp part, 
 there
 TC is a race condition
 TC Tz in inicialisation in jitter buffer, another patch implements method 
 to
 TC send progress indicator,
 TC Tz and i don't wont spent my time to incorporate this changes into
 TC libopal.
 TC Tz
 TC TzThats bad.
 TC TzAny bugfixes from fs, goes to upstream on any of the used libraries.
 TC TzYou should be doing the same.
 TC TzAnd Opal developers, will either include or refuse your patches. If
 TC Tzthey refuse it, they will give you the reason.
 TC
 TC i make this fix only to freeze my current mod_opal work on working state,
 TC while it now work for me i work on
 TC my new implementation of h323 proto for fs, i think opal developers will
 TC fix this rtp bug himself becouse
 TC it crashes and make library unuseful.
 TC
 TC Tz
 TC Tz without this changes
 TC Tz my work on mod_opal in freeswitch don't useful at all, i provide 
 link
 TC to my work with all
 TC Tz patches, if somebody wont incorporate it in libopal tree and fs - go
 TC on, but i think
 TC Tz better and more elegant make new module based on h323plus.
 TC Tz
 TC TzIf you dont publish your changes, all those you are trying to achieve,
 TC Tzwont happen.
 TC Tz
 TC Tz
 TC Tz AMHow far away from what is in tree are these patches you have?
 TC Tz AM
 TC Tz AM2009/10/8 Georgiewskiy Yuriy bottle...@icf.org.ru
 TC Tz AM
 TC Tz AM On 2009-10-07 15:09 -0500, Brian West wrote
 TC Tz AM freeswitch-us...@lists.freeswit...:
 TC Tz AM
 TC Tz AM opal have addition abstraction layer called opalmgr, and it
 TC implementation
 TC Tz AM is not so good in
 TC Tz AM this case, for example to implemet pre_answer in mod_opal i 
 need
 TC patch
 TC Tz AM libopal, because
 TC Tz
 TC TzThe patch you have in there, adds a method to the OpalCall,  it does
 TC Tznot touch any parts of OpalManager
 TC Tzso, i dont understand why opalmanager would be the cause of your pain?
 TC Tz
 TC Tz AM there is no way to send progress inicator throuch opalmgr. and
 TC there is
 TC Tz AM many another issues like
 TC Tz AM this in that layer.
 TC Tz
 TC TzPlease point me to the issues you have in opal, their bug reports ,
 TC traces etc.
 TC TzI dont think any of the opal people has psychic abilities to detect
 TC Tz-your- problems
 TC Tzand solve them.
 TC Tz
 TC Tzftp://srv.icf.org.ru/pub/soft/f/freeswitch/ - there is
 TC Tz AM my work on mod_opal before
 TC Tz AM i start moving to h323plus, may be this help somebody there.
 TC Tz AM
 TC Tz AM BW From what I have been told h323plus is a based/fork of
 TC OpenH323 which
 TC Tz AM BWOPAL is just a continuation of OpenH323.  So why not support
 TC the
 TC Tz AM BWdevelopers of OPAL/OpenH323 ?
 TC Tz AM BW
 TC Tz AM BW/b
 TC Tz AM BW
 TC Tz AM BWOn Oct 7, 2009, at 7:50 AM, Georgiewskiy Yuriy wrote:
 TC Tz AM BW
 TC Tz AM BW We are developing module to handle h323 proto now, we try 
 to
 TC use
 TC Tz AM BW mod_opal and try improve it, but no luck,
 TC Tz AM BW there is many issues in libopal, and finaly we now move to
 TC h323plus
 TC Tz AM BW library.
 TC Tz
 TC TzDid any of you try to report those issues?
 TC Tz
 TC TzRegards
 TC Tz
 TC Tz/tyn
 TC Tz
 TC Tz AM BW
 TC Tz AM BW
 TC Tz AM BW___
 TC Tz AM BWFreeSWITCH-users mailing list
 TC Tz AM BWFreeSWITCH-users@lists.freeswitch.org
 TC Tz AM 
 BWhttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 TC Tz AM BWUNSUBSCRIBE:
 TC Tz AM http://lists.freeswitch.org/mailman/options/freeswitch-users
 TC Tz AM BWhttp://www.freeswitch.org
 TC Tz AM BW
 TC Tz AM
 TC Tz AM C уважением                       With Best Regards
 TC Tz AM Георгиевский Юрий.                

Re: [Freeswitch-users] FS Slide deck?

2009-10-09 Thread Michael Collins
On Thu, Oct 8, 2009 at 11:41 PM, Gabriel Gunderson g...@gundy.org wrote:

 On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins m...@freeswitch.org
 wrote:
  Thanks for reporting back. Please let all the Asterisk users know that
 they
  are welcome to join us in #freeswitch on irc.freenode.net and that they
 will
  not be abused like people do in other less friendly IRC channels.

 Funny you mention this.  Many people report that the way the FS
 community refers to Asterisk in docs/wikis/irc/whatever makes the FS
 camp seem *less* welcoming to them.  After all, they identify as
 Asterisk Users and take the criticism as being kinda harsh.  Most of
 them acknowledge the shortcomings of Asterisk but are put off when
 someone else points them out.  It's crazy, I know.  The thing is, I
 remember thinking that too.  After getting to know FS better, I didn't
 notice it as much.  Nobody likes to hear their baby is ugly --even if
 they know it is.


Interesting. I'd like to know what specifically turned them off. It could be
node 117, but that was written by Anthony in response to dozens of questions
about the subject. In any case, if the Asterisk guys look at the first
edition of the Starfish/TFOT book they'll see glowing praise of Anthony
Minessale and Brian West, so maybe they'll be willing to listen to those
guys since they've got so much Asterisk knowledge. The other thing is that
many people in the FreeSWITCH community are Asterisk refugees. They left
Asterisk for FreeSWITCH because they had some sort of trouble with Asterisk.
If that comes across in the docs then we'll have to see about cleaning that
up.


 At our session, and in general, I've noticed people are more
 interested in hearing about FS when you don't make direct comparisons
 to Asterisk.  Besides, FS stands on it's own merit.


Agreed. Direct comparisons can always be slanted in any direction. Besides,
FreeSWITCH stacks up very well against a lot of modern (i.e. expensive)
equipment put out by the big boys (Cisco, Avaya, NEC, ShoreTel, etc.).


 Just what I've observed *and* my 2 additional cents.

 Gabe

Thanks,
MC
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Re: [Freeswitch-users] [ERR] mod_dingaling.c:980 Stun Failed!

2009-10-09 Thread Mark Campbell-Smith
It was the svn revision I was using... I have updated now and it is
working again.

On Fri, Oct 2, 2009 at 9:00 PM, Muhammad Shahzad
shaherya...@googlemail.com wrote:
 Yes, i had same problem, then i changed stun server to one of our own
 servers. You may try some of public stun servers listed on below link,

 http://www.voip-info.org/wiki/view/STUN

 Thank you.


 On Fri, Oct 2, 2009 at 2:58 PM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:

 Anyone have this issue?

 On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
 mcampbellsm...@gmail.com wrote:
  Hi!
 
  I have just started to use dingaling again, and noticed I constantly
  get a stun error.
 
  2009-09-29 21:11:03.175002 [ERR] mod_dingaling.c:980 Stun Failed!
  stun.fwdnet.net:3478 [Remote Address Error!]
 
  I have tried with stun.freeswitch.org and stun.fwdnet.net stun servers
  and keep getting this error with dingaling.  I have no problems with
  inbound sip calls, so I don't think  its the actual stun server.
 
  Has anyone else seen this?  I am using: FreeSWITCH Version 1.0.trunk
  (14952)
 
  Thanks!
 

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 --
 
 |
                      |
 | FATAL ERROR                                                           ---
 O X |
 |___|
 |                        You have moved the mouse.
  |
 | Windows must be restarted for the changes to take effect.   |
 |                                        OK
                |
 /


 Muhammad Shahzad
 ---
 CISCO Rich Media Communication Specialist (CRMCS)
 CISCO Certified Network Associate (CCNA)
 Cell: +92 334 422 40 88
 MSN: shari_78...@hotmail.com
 Email: shaherya...@googlemail.com

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Re: [Freeswitch-users] openzap Failure opening channel error

2009-10-09 Thread lakshmanan ganapathy
Thanks all u guys. It got worked when I replaced with the following
configuration and the incoming has no problem with that.
[span zt PRI_1]
trunk_type = e1
b-channel = 1-15
d-channel = 16
b-channel = 17-31

But still I'm facing problem with the outgoing call. It says
INVALID_IE_CONTENTS.
What might be the issue? Even I tried the following dialplan to call by
using bridge.
extension name=outgoing-fxo
  condition field=destination_number
expression=^(\d{10})$
  action application=answer/
  action application=bridge
data=openzap/1/1/${dialed_ext}/
  /condition
  /extension
But it prints the same error.
What might be the issue?


On Thu, Oct 8, 2009 at 10:24 PM, Michael Collins m...@freeswitch.org wrote:



 On Thu, Oct 8, 2009 at 8:02 AM, Moises Silva moises.si...@gmail.comwrote:

 On Thu, Oct 8, 2009 at 5:41 AM, lakshmanan ganapathy 
 lakindi...@gmail.com wrote:

 Hi I tried with the following openzap.conf
 [span zt PRI_1]
 trunk_type = e1
 b-channel = 1:1-15,17-31
 d-channel = 1:16


 This does not look like a healthy config to me. You are using
 span:channels notation in zaptel spans, zaptel channels increment across
 spans, read point number 5 of this web page
 http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , this is for another type
 of signaling, but the span/channel numbering concepts are the same.


 Moy,

 As usual you are right on the money. I'm so used to doing Sangoma configs
 that I forget about the zaptel syntax. I recommend this config:

 [span zt PRI_1]
 trunk_type = e1
 b-channel = 1-15
 d-channel = 16
 b-channel = 17-31


 Give that a try and report back if you have issues.
 -MC

  --
 Moises Silva
 Software Developer
 Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
 9T3 Canada
 t. 1 905 474 1990 x 128 | e. m...@sangoma.com

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[Freeswitch-users] apr_queue

2009-10-09 Thread srinivasula reddy
Hi all,

does any know about How apr_queue is maintaing and retriving all registered
and all stuff


-- 
Srinivasula Reddy K
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Re: [Freeswitch-users] mod_fax compile fails

2009-10-09 Thread Rob Forman
I was in a production window so when the latest trunk worked I moved  
on.  I went back to troubleshoot later when I saw your email but I  
couldn't reproduce it.

I like the tmpfs build- thanks for the tip Jason.


On Oct 9, 2009, at 1:11 AM, Jason White wrote:

 Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
 Thanks Rob,

 Is this a fault in the svn update process?

 Usually, no, it's rather that if you build in a directory that  
 you've checked
 out and then run svn update, not everything gets cleaned up properly  
 the next
 time you build.

 I find it's faster to build in a tmpfs file system anyway, e.g.,

 mkdir /tmp/fs  sudo mount -t tmpfs tmpfs /tmpfs  svn export . / 
 tmp/fs
 then go into /tmp/fs and run the compilation process, or the package  
 building
 command, or whatever you need.

 This way, my original svn tree is never altered and it will always  
 update
 cleanly.


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Re: [Freeswitch-users] openzap Failure opening channel error

2009-10-09 Thread Russell.Mosemann
lakshmanan ganapathy lakindi...@gmail.com said:

 But still I'm facing problem with the outgoing call. It says
 INVALID_IE_CONTENTS.
 What might be the issue? Even I tried the following dialplan to call by
 using bridge.
 extension name=outgoing-fxo
   condition field=destination_number
 expression=^(\d{10})$
   action application=answer/
   action application=bridge
 data=openzap/1/1/${dialed_ext}/

Does answer need to be called here? I haven't used an fxo. So, I don't
know. What value does $dialed_ext have? If you want to use the number
matched in the condition, then it should be

openzap/1/1/$1

-- 
Russell Mosemann




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


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Re: [Freeswitch-users] libpri_span vs prispans

2009-10-09 Thread Russell.Mosemann
lakshmanan ganapathy lakindi...@gmail.com said:
 What difference it will make, if I use pri_span configuration and lib_pri
 span configuration in openzap.conf.xml.
 I'm preety much confused on the difference between this.

Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
libpri is more mature than openzap. In my case openzap was not able to
handle everything invovling a T1 here, and using libpri fixed it.

A libpri_span configuration is handled by libpri. A pri_span
configuration is handled natively by openzap. Use the one that works for you.

-- 
Russell Mosemann




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


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Re: [Freeswitch-users] NAT problems - sorry

2009-10-09 Thread Andy
Hi folks,
 
Just thought I'd post the solution to this problem. In my case the problem
was that my config was based on an older version and so was missing a
crucial parameter that exists in the default config wirth a new install.
 
The external.xml profile needed to contain the parameter:
 
param name=local-network-acl value=localnet.auto/
 
With this in place, DTMF and hangup messages traverse the nat firewall
correctly. Without it they don't. I searched the Wiki and couldn't find any
info on this parameter. Can anyone provide a description of what it does and
why it's significant that can be added to the WIKI?
 
Cheers
Andy

  _  

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: 07 October 2009 19:07
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] NAT problems - sorry


s/auto-nat/$realip/ then forward the rtp ports and sip ports. 

/b
PS chances are you have to ENABLE upnp.

On Oct 7, 2009, at 12:58 PM, Andy wrote:


Many thanks Brian, the firewall docs assure me it is uPnp but is probably
lying or a poor implementation. Could you point me to the right section of
the Wiki to tell me how to do this manually as I've been scouting for some
time and can;t seem to find the right thing. sorry if I'm being blind.



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Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

2009-10-09 Thread Anthony Minessale
or twist that into a feature... =D

alice is bob's secretary so she calls him on his cell when he has an appt
and parks him.
Then she calls the other guy and says I have bob on line for you and xfers
him to the same parking space =p


On Fri, Oct 9, 2009 at 7:43 AM, Seven Du dujinf...@gmail.com wrote:

 It's very cool. But think about the following scenario:

 Alice answers a call from Bob.
 Alice transfers Bob to 6001 and hangs up.
 Bob is now in parking stall 6001, hearing MOH.
 Alice calls Charlie and tells him that he has a call parked in 6001.

   Dee call in and answered by Alice  # at that time, Alice don't know the
 lot is empty or not
   If Alice transfer Dee to 6001 # then Dee will talk to Bob, that's not
 expected

 Charlie calls 6000, waits for prompt, then dials 6001#.
 Charlie Listening to music.

 Event it's rarely happen, I do think Alice would like to now the if lot is
 empty or not before transfer. Make sense?

 2009/10/9 Michael Collins m...@freeswitch.org

 FYI,

 The FreeSWITCH devs have added valet parking! Check it out:
 http://www.freeswitch.org/node/207

 Let us know what you think.

 Thanks,
 Michael

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-- 
Anthony Minessale II

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

2009-10-09 Thread Maciej Aniserowicz

Hello,
The issue is resolved. I feel stupid, because Michael Jerris was right the 
first time. Setting external_rtp_ip and external_sip_ip to $${local_ip_v4} made 
it work.
But the strange thing is: it SOMETIMES worked before without any delay, which 
'should not be possible', because the original IP was my external ip and the 
BYE message was sent straight to it. And there is no way it could reach the 
target 'internal' FS, because it runs on virtual machine, and no ports are 
forwarded on my router.
Any thoughts? Why this could (rarely) work even with the previous config?

Thanks to both of you for your answers.

MA


  - Original Message - 
  From: mercutioviz (via Nabble) 
  To: Maciej Aniserowicz 
  Sent: Thursday, October 08, 2009 7:06 PM
  Subject: Re: [Freeswitch-users] gateway FS informs it's client FS about users 
hanged up with a long delay





  On Thu, Oct 8, 2009 at 6:18 AM, Maciej Aniserowicz [hidden email] wrote:


Both of the instances are run on the same machine, i just changed the 
default
ports they use. Can anything else cause this strange behavior?
MA

  Did a packet capture yield any clues? That is, were you able to confirm that 
each instance sent and received all the packets that you believe they should 
have sent and received? The reason I ask is so that you don't end up chasing a 
ghost because you made an assumption somewhere in your troubleshooting.

  -MC
  Â 




Michael Jerris wrote:

 Incorrect NAT configuration so one of the boxes is not actually
 getting a BYE.


 On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:

 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?


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Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

2009-10-09 Thread Rupa Schomaker
Oh, definitely.  I think both use cases (existing, and park in next
avail slot and read back slot #) are useful.

On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
 or twist that into a feature... =D

 alice is bob's secretary so she calls him on his cell when he has an appt
 and parks him.
 Then she calls the other guy and says I have bob on line for you and xfers
 him to the same parking space =p



-- 
-Rupa

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Re: [Freeswitch-users] mod_fax compile fails

2009-10-09 Thread leo.z...@gmail.com
Hi,

./bootstrap.sh
./configure
make
make install

--
Regards,
Leo



Mark Campbell-Smith wrote:
 HI all,

 I just tried to update to the latest svn and I get these errors right
 at the end after issuing a 'make current'.  I am using Debian Lenny.

 making all mod_fax
 make[5]: Entering directory 
 `/home/mark/freeswitch/src/mod/applications/mod_fax'
 make[6]: Entering directory 
 `/home/mark/freeswitch/src/mod/applications/mod_fax'
 make[7]: Entering directory `/home/mark/freeswitch/libs/tiff-3.8.2'
 running /bin/sh ./configure  --prefix=/usr/local/freeswitch
 --cache-file=/dev/null --srcdir=. --disable-shared --with-pic
 --no-create --no-recursion
 configure: error: cannot run /bin/sh config/config.sub
 make[7]: *** [config.status] Error 1
 make[7]: Leaving directory `/home/mark/freeswitch/libs/tiff-3.8.2'
 make[6]: *** [../../../../libs/tiff-3.8.2/libtiff/libtiff.la] Error 2
 make[6]: Leaving directory 
 `/home/mark/freeswitch/src/mod/applications/mod_fax'
 make[5]: *** [all] Error 1
 make[5]: Leaving directory 
 `/home/mark/freeswitch/src/mod/applications/mod_fax'
 make[4]: *** [mod_fax-all] Error 1
 make[4]: Leaving directory `/home/mark/freeswitch/src/mod'
 make[3]: *** [all-recursive] Error 1
 make[3]: Leaving directory `/home/mark/freeswitch/src'
 Making all in build
 make[3]: Entering directory `/home/mark/freeswitch/build'
  + FreeSWITCH Build Complete ---+
  + FreeSWITCH has been successfully built.  +
  + Install by running:  +
  +  +
  +   make install   +
  +--+
 make[3]: Leaving directory `/home/mark/freeswitch/build'
 make[2]: *** [all-recursive] Error 1
 make[2]: Leaving directory `/home/mark/freeswitch'
 make[1]: *** [all] Error 2
 make[1]: Leaving directory `/home/mark/freeswitch'
 make: *** [current] Error 2

 Also, are the 'Leaving directory / all-recursive' errors going to
 cause a problem?

 Thanks!
 Any ideas what the cause is?

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Re: [Freeswitch-users] SLAs and BLAs

2009-10-09 Thread Jerry Richards

I gather from the mailing archive that BLAs are implemented using the
draft-anil-sipping-bla-04.txt document.  According to the draft, the
Appearance Agent is supposed to initiate a SUBSCRIBE request, but I don't
see FS doing this.

What phone types/models are known to work with the FS BLA implementation?

Best Regards,
Jerry
 

-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com] 
Sent: Monday, October 05, 2009 3:24 PM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] SLAs and BLAs

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] Heartbeat question

2009-10-09 Thread Anthony Minessale
Update to trunk and try it with fs_cli it for sure will let you do every 1
second

in fs_cli type

/events plain all

if you make that call you will see one every 1 second


On Fri, Oct 9, 2009 at 12:45 AM, Diego Viola diego.vi...@gmail.com wrote:

 Nope, I was just wondering why it didn't work at 1 second exactly...


 On Fri, Oct 9, 2009 at 3:36 AM, William Suffill william.suff...@gmail.com
  wrote:

 Why do you need it every second? If you want real time channel counts
 you would  be able to track each create/destroy even instead of
 relying on the heartbeat summary.

 -- W

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-- 
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FreeSWITCH http://www.freeswitch.org/
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sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
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[Freeswitch-users] FreeSWITCH Weekly Conference Call Starting, Please Join Us!

2009-10-09 Thread Michael Collins
Agenda and call-in info: http://bit.ly/lzEYy
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Re: [Freeswitch-users] apr_queue

2009-10-09 Thread Michael Jerris
On Oct 9, 2009, at 7:58 AM, srinivasula reddy wrote:

 Hi all,

 does any know about How apr_queue is maintaing and retriving all  
 registered and all stuff


parse error


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Re: [Freeswitch-users] libpri_span vs prispans

2009-10-09 Thread Michael Collins
On Fri, Oct 9, 2009 at 5:57 AM, russell.mosem...@cune.org wrote:

 lakshmanan ganapathy lakindi...@gmail.com said:
  What difference it will make, if I use pri_span configuration and lib_pri
  span configuration in openzap.conf.xml.
  I'm preety much confused on the difference between this.

 Openzap can handle a T1/E1 itself, or openzap can use libpri to do that.
 libpri is more mature than openzap. In my case openzap was not able to
 handle everything invovling a T1 here, and using libpri fixed it.

 A libpri_span configuration is handled by libpri. A pri_span
 configuration is handled natively by openzap. Use the one that works for
 you.


Right now the libpri stack is more mature than the ozmod_isdn stuff. See the
OpenZAP page on the wiki for instructions on how to set up ozmod_libpri.
-MC
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Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

2009-10-09 Thread Michael Collins
On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker r...@rupa.com wrote:

 Oh, definitely.  I think both use cases (existing, and park in next
 avail slot and read back slot #) are useful.


They do have some usefulness. The most important reason for this feature,
though, is for the person who handles calls all day and moves them around.
That person will most likely have a pretty good idea of which parking stalls
are available. However, I do like the idea of a webby interface showing what
stalls are in use...
-MC


 On Fri, Oct 9, 2009 at 8:25 AM, Anthony Minessale
 anthony.miness...@gmail.com wrote:
  or twist that into a feature... =D
 
  alice is bob's secretary so she calls him on his cell when he has an appt
  and parks him.
  Then she calls the other guy and says I have bob on line for you and
 xfers
  him to the same parking space =p



 --
 -Rupa

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[Freeswitch-users] On the handling of SIP headers

2009-10-09 Thread Kristian Kielhofner
Hello everyone,

  In using FS for various scenarios I've noticed some behavior that
I'm not sure is completely proper.  Given that this probably lives
in mod_sofia who knows what's really proper.  It is SIP after all...

  So the issue comes up when using FreeSWITCH as a B2BUA and bridging
between endpoints (very common).  Should FreeSWITCH copy the X-
headers (possibly others) as it does now?  I'd like to think it
shouldn't by default and the behavior should be one of:

1)  Don't pass X-* (or anything else, really) from one leg to another.
 If you want to pass specific X- headers (or anything else), set them
explicitly on the outbound leg.
2)  Make the behavior configurable with a channel variable and/or
sofia config option:

{sip_pass_headers=all|none|X-MyCustomHeaderByName}

  Thoughts?

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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[Freeswitch-users] Sending an Event to a Session for onInput

2009-10-09 Thread Matthew Fong
I'm used to using the onInput callbacks inside lua and javascript to listen
for dtmf and other events and perform a task accordingly. I'm wondering if
there is a way to send an event to a session or channel that can be caught
using the setInputCallback inside lua from outside the session program.
Maybe an API command that can generate an event for a specific UUID. Does a
mechanism exist to do this that I'm over looking? Thanks.
--matt
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[Freeswitch-users] FW: FS Does Not Relay PresencePUBLISHToSubscribing Phones

2009-10-09 Thread Jerry Richards
I put the sqlite3 select query in the paste bin again, and prior to that, I
entered the .dump command.  The select command came back with the sqlite3
prompt, which I guess means it didn't find an entry.  How do I go about
isolating this problem?  I'm using CounterPath's Bria Professional
softphone.  They are the same company that make the Eyebeam.
 
Any ideas?
 
Best Regards,
Jerry


  _  

From: Jerry Richards [mailto:jerry.richa...@teotech.com] 
Sent: Friday, October 02, 2009 11:28 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


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


  _  

From: Anthony Minessale [mailto:anthony.miness...@gmail.com] 
Sent: Friday, October 02, 2009 10:52 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay
PresencePUBLISHToSubscribing Phones


connect to sqlite directly with sqlite3 app and try that sql stmt and see
why it doesn't match anything.

sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

select
sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_hos
t,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscripti
ons.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscripti
ons.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subsc
riptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,
'Away','away','192.168.72.38',sip_presence.status,sip_presence.rpid from
sip_subscriptions left join sip_presence on
(sip_subscriptions.sub_to_user=sip_presence.sip_user and
sip_subscriptions.sub_to_host=sip_presence.sip_host and
sip_subscriptions.profile_name=sip_presence.profile_name) where
(event='presence' or event='presence') and sub_to_user='1001' and
(sub_to_host='192.168.72.38' or presence_hosts like '%192.168.72.38%') and
(sip_subscriptions.profile_name = 'external' or
sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)



On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards jerry.richa...@teotech.com
wrote:


Okay, I put a log up on the pastebin that shows the PUBLISH event coming
from a CounterPath Bria Professional phone.  For some reason, FS is getting
an error and not relaying the presence status to the subscriber.
 
Best Regards,
Jerry


  _  

From: João Mesquita [mailto:jmesqu...@freeswitch.org] 
Sent: Thursday, October 01, 2009 8:14 PM 

To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Does Not Relay Presence
PUBLISHToSubscribing Phones


Piece of advice, don't ask, just do it. ;)

jmesquita


On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards jerry.richa...@teotech.com
wrote:


If you have time to take a look, I could put a trace in the pastebin?
 
Jerry


  _  


From: Jerry Richards [mailto:jerry.richa...@teotech.com] 

Sent: Thursday, October 01, 2009 10:29 AM 

To: 'freeswitch-users@lists.freeswitch.org'

Subject: RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


I am using two Bria Professional Version 2.5.4 Build 54835 softphones.
 
Thanks,
Jerry


  _  


From: Anthony Minessale [mailto:anthony.miness...@gmail.com] 

Sent: Thursday, October 01, 2009 9:36 AM 

To: freeswitch-users@lists.freeswitch.org

Subject: Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
ToSubscribing Phones


which phone is it,
we tested it with eyebeam and it appears to work for us.



On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards jerry.richa...@teotech.com
wrote:



By the way, I see the following lines at the FS console, which might be a
clue as to why this is happening.  Could someone point me toward what might
cause this?  I set the manage-presence parameter to true in each XML
file where I saw it defined.

[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
[ERR] sofia_presence.c:611  DUMP PRESENCE SQL
  ...
[WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


Best Regards,
Jerry



-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Wednesday, September 30, 2009 9:12 AM
To: 'freeswitch-users@lists.freeswitch.org'
Subject: FS Does Not Relay Presence PUBLISH To Subscribing Phones

I have two phones configured to subscribe to each other's presence status.
When I change the presence status in one phone, I see the SIP PUBLISH
message going to FS, but I don't see FS relaying that presence status to the
subscribing phone.  Does anyone know why?

Best Regards,
Jerry



Re: [Freeswitch-users] FreeSWITCH Update: Valet Parking

2009-10-09 Thread Andrew Thompson
On Fri, Oct 09, 2009 at 11:01:34AM -0700, Michael Collins wrote:
 On Fri, Oct 9, 2009 at 7:50 AM, Rupa Schomaker r...@rupa.com wrote:
 
  Oh, definitely.  I think both use cases (existing, and park in next
  avail slot and read back slot #) are useful.
 
 
 They do have some usefulness. The most important reason for this feature,
 though, is for the person who handles calls all day and moves them around.
 That person will most likely have a pretty good idea of which parking stalls
 are available. However, I do like the idea of a webby interface showing what
 stalls are in use...

mod_snom and the programmable LEDs would be a handy way to do it too. I
implemented a poor-man's version of a key system using that, but I think
I'll replace the parking portion with the valet parking stuff when I
get a chance.

Andrew

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Re: [Freeswitch-users] FW: FS Does Not Relay PresencePUBLISHToSubscribing Phones

2009-10-09 Thread Anthony Minessale
yah what i was getting at was between the sql statement and what was
actually in the table you should be able to tell what's wrong.


On Fri, Oct 9, 2009 at 3:06 PM, Jerry Richards
jerry.richa...@teotech.comwrote:

  I put the sqlite3 select query in the paste bin again, and prior to that,
 I entered the .dump command.  The select command came back with the
 sqlite3 prompt, which I guess means it didn't find an entry.  How do I go
 about isolating this problem?  I'm using CounterPath's Bria Professional
 softphone.  They are the same company that make the Eyebeam.

 Any ideas?

 Best Regards,
 Jerry


  --
 *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
 *Sent:* Friday, October 02, 2009 11:28 AM
 *To:* 'freeswitch-users@lists.freeswitch.org'
 *Subject:* RE: [Freeswitch-users] FS Does Not Relay
 PresencePUBLISHToSubscribing Phones

  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

  --
 *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
 *Sent:* Friday, October 02, 2009 10:52 AM
 *To:* freeswitch-users@lists.freeswitch.org
 *Subject:* Re: [Freeswitch-users] FS Does Not Relay
 PresencePUBLISHToSubscribing Phones

 connect to sqlite directly with sqlite3 app and try that sql stmt and see
 why it doesn't match anything.

 sqlite3 /usr/local/freeswitch/db/sofia_reg_internal.db

 select
 sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Away','away','
 192.168.72.38',sip_presence.status,sip_presence.rpid from
 sip_subscriptions left join sip_presence on 
 (sip_subscriptions.sub_to_user=sip_presence.sip_user
 and sip_subscriptions.sub_to_host=sip_presence.sip_host and
 sip_subscriptions.profile_name=sip_presence.profile_name) where 
 (event='presence'
 or event='presence') and sub_to_user='1001' and (sub_to_host='192.168.72
 .38' or presence_hosts like '%192.168.72.38%') and 
 (sip_subscriptions.profile_name
 = 'external' or sip_subscriptions.presence_hosts !=
 sip_subscriptions.sub_to_host)


 On Fri, Oct 2, 2009 at 12:12 PM, Jerry Richards 
 jerry.richa...@teotech.com wrote:

  Okay, I put a log up on the pastebin that shows the PUBLISH event coming
 from a CounterPath Bria Professional phone.  For some reason, FS is getting
 an error and not relaying the presence status to the subscriber.

 Best Regards,
 Jerry

  --
 *From:* João Mesquita [mailto:jmesqu...@freeswitch.org]
 *Sent:* Thursday, October 01, 2009 8:14 PM

 *To:* freeswitch-users@lists.freeswitch.org
 *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence
 PUBLISHToSubscribing Phones

   Piece of advice, don't ask, just do it. ;)

 jmesquita

 On Thu, Oct 1, 2009 at 3:29 PM, Jerry Richards 
 jerry.richa...@teotech.com wrote:

  If you have time to take a look, I could put a trace in the pastebin?

 Jerry

  --
  *From:* Jerry Richards [mailto:jerry.richa...@teotech.com]
 *Sent:* Thursday, October 01, 2009 10:29 AM
 *To:* 'freeswitch-users@lists.freeswitch.org'
 *Subject:* RE: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
 ToSubscribing Phones

  I am using two Bria Professional Version 2.5.4 Build 54835 softphones.

 Thanks,
 Jerry

  --
  *From:* Anthony Minessale [mailto:anthony.miness...@gmail.com]
 *Sent:* Thursday, October 01, 2009 9:36 AM
 *To:* freeswitch-users@lists.freeswitch.org
 *Subject:* Re: [Freeswitch-users] FS Does Not Relay Presence PUBLISH
 ToSubscribing Phones

   which phone is it,
 we tested it with eyebeam and it appears to work for us.


 On Thu, Oct 1, 2009 at 9:57 AM, Jerry Richards 
 jerry.richa...@teotech.com wrote:


 By the way, I see the following lines at the FS console, which might be
 a
 clue as to why this is happening.  Could someone point me toward what
 might
 cause this?  I set the manage-presence parameter to true in each XML
 file where I saw it defined.

 [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal)
 [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
 [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (internal-ipv6)
 [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
 [INFO] sofia_presence.c:603 IN START_PRESENCE_SQL (external)
 [ERR] sofia_presence.c:611  DUMP PRESENCE SQL
   ...
 [WARNING] sofia_presence.c:565  192.168.72.38 is an alias, skipping


 Best Regards,
 Jerry


 -Original Message-
 From: Jerry Richards [mailto:jerry.richa...@teotech.com]
 Sent: Wednesday, 

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

2009-10-09 Thread Nagalenoj

No, When I do voicemail_inject it is reporting the following error.

'Stereo is currently not supported, please downsample to mono.'

Any other way to do this.??


Michael Jerris wrote:
 
 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] Re corded file as voicemail.

2009-10-09 Thread Jason White
Nagalenoj nagale...@gmail.com wrote:
 
 No, When I do voicemail_inject and check through voicemail, it is not playing
 the file instead reporting the following error.
 
 'Stereo is currently not supported, please downsample to mono.'

This has been discussed on the list before. I think the solution was to use
sox to convert the files to mono. You might need a script to do this before
injecting them into the voicemail.


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

2009-10-09 Thread Seven Du
Yes, it's discussed before.

http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO

set that var to false before you record.


2009/10/10 Jason White ja...@jasonjgw.net

 Nagalenoj nagale...@gmail.com wrote:
 
  No, When I do voicemail_inject and check through voicemail, it is not
 playing
  the file instead reporting the following error.
 
  'Stereo is currently not supported, please downsample to mono.'

 This has been discussed on the list before. I think the solution was to use
 sox to convert the files to mono. You might need a script to do this before
 injecting them into the voicemail.


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