Re: [Freeswitch-users] FS Slide deck?

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

> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins 
> 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
 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
>  wrote:
>>
>> Anyone have this issue?
>>
>> On Tue, Sep 29, 2009 at 9:15 PM, Mark Campbell-Smith
>>  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.   |
> |                                        
>                |
> /
>
>
> 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.

  
  
  
  
  
But it prints the same error.
What might be the issue?


On Thu, Oct 8, 2009 at 10:24 PM, Michael Collins  wrote:

>
>
> On Thu, Oct 8, 2009 at 8:02 AM, Moises Silva wrote:
>
>> 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
>> : 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] libpri_span vs prispans

2009-10-09 Thread lakshmanan ganapathy
Hi all,

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.
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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  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  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.
> 
>expression="^(\d{10})$">
>   
>data="openzap/1/1/${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] FreeSWITCH Update: Valet Parking

2009-10-09 Thread Seven Du
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 

> 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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Re: [Freeswitch-users] libpri_span vs prispans

2009-10-09 Thread Russell.Mosemann
lakshmanan ganapathy  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] FreeSWITCH Update: Valet Parking

2009-10-09 Thread Rupa Schomaker
Perhaps a "put in next empty slot" mode and then a say of the slot so
Alice and make a note of the lot #?

On Fri, Oct 9, 2009 at 6:43 AM, Seven Du  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 
>>
>> 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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-- 
-Rupa

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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:
 

 
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  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 
>
>> 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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>
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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
 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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>   


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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  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  > 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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>


-- 
Anthony Minessale II

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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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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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,  wrote:

> lakshmanan ganapathy  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  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
>  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 
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 
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 
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


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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  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] Heartbeat question

2009-10-09 Thread Diego Viola
Thanks Anthony, this solved it. You rock :)

My program now outputs:

Got a SESSION_HEARTBEAT at 17:14:59
Got a SESSION_HEARTBEAT at 17:15:00
Got a SESSION_HEARTBEAT at 17:15:02
Got a SESSION_HEARTBEAT at 17:15:03
Got a SESSION_HEARTBEAT at 17:15:04
Got a SESSION_HEARTBEAT at 17:15:05
Got a SESSION_HEARTBEAT at 17:15:06
Got a SESSION_HEARTBEAT at 17:15:07
Got a SESSION_HEARTBEAT at 17:15:08
Got a SESSION_HEARTBEAT at 17:15:09
Got a SESSION_HEARTBEAT at 17:15:10
Got a SESSION_HEARTBEAT at 17:15:11
Got a SESSION_HEARTBEAT at 17:15:12
Got a SESSION_HEARTBEAT at 17:15:13
Got a SESSION_HEARTBEAT at 17:15:14
Got a SESSION_HEARTBEAT at 17:15:15
Got a SESSION_HEARTBEAT at 17:15:16
Got a SESSION_HEARTBEAT at 17:15:17
Got a SESSION_HEARTBEAT at 17:15:18
Got a SESSION_HEARTBEAT at 17:15:19
Got a SESSION_HEARTBEAT at 17:15:20
Got a SESSION_HEARTBEAT at 17:15:21
Got a SESSION_HEARTBEAT at 17:15:22
Got a SESSION_HEARTBEAT at 17:15:23
Got a SESSION_HEARTBEAT at 17:15:24
Got a SESSION_HEARTBEAT at 17:15:25
Got a SESSION_HEARTBEAT at 17:15:26
Got a SESSION_HEARTBEAT at 17:15:27
Got a SESSION_HEARTBEAT at 17:15:28
Got a SESSION_HEARTBEAT at 17:15:29
Got a SESSION_HEARTBEAT at 17:15:30




On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> 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 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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>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
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>
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Re: [Freeswitch-users] Heartbeat question

2009-10-09 Thread Diego Viola
Here is on two seconds ;)

Got a SESSION_HEARTBEAT at 17:17:13
Got a SESSION_HEARTBEAT at 17:17:15
Got a SESSION_HEARTBEAT at 17:17:17
Got a SESSION_HEARTBEAT at 17:17:19
Got a SESSION_HEARTBEAT at 17:17:21
Got a SESSION_HEARTBEAT at 17:17:23
Got a SESSION_HEARTBEAT at 17:17:25
Got a SESSION_HEARTBEAT at 17:17:27
Got a SESSION_HEARTBEAT at 17:17:29
Got a SESSION_HEARTBEAT at 17:17:31
Got a SESSION_HEARTBEAT at 17:17:33
Got a SESSION_HEARTBEAT at 17:17:35
Got a SESSION_HEARTBEAT at 17:17:37
Got a SESSION_HEARTBEAT at 17:17:39
Got a SESSION_HEARTBEAT at 17:17:41
Got a SESSION_HEARTBEAT at 17:17:43
Got a SESSION_HEARTBEAT at 17:17:45
Got a SESSION_HEARTBEAT at 17:17:47
Got a SESSION_HEARTBEAT at 17:17:49
Got a SESSION_HEARTBEAT at 17:17:51
Got a SESSION_HEARTBEAT at 17:17:53
Got a SESSION_HEARTBEAT at 17:17:55
Got a SESSION_HEARTBEAT at 17:17:57
Got a SESSION_HEARTBEAT at 17:17:59
Got a SESSION_HEARTBEAT at 17:18:01
Got a SESSION_HEARTBEAT at 17:18:03
Got a SESSION_HEARTBEAT at 17:18:05
Got a SESSION_HEARTBEAT at 17:18:07
Got a SESSION_HEARTBEAT at 17:18:09
Got a SESSION_HEARTBEAT at 17:18:11
Got a SESSION_HEARTBEAT at 17:18:13
Got a SESSION_HEARTBEAT at 17:18:15
Got a SESSION_HEARTBEAT at 17:18:17


On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola  wrote:

> Thanks Anthony, this solved it. You rock :)
>
> My program now outputs:
>
> Got a SESSION_HEARTBEAT at 17:14:59
> Got a SESSION_HEARTBEAT at 17:15:00
> Got a SESSION_HEARTBEAT at 17:15:02
> Got a SESSION_HEARTBEAT at 17:15:03
> Got a SESSION_HEARTBEAT at 17:15:04
> Got a SESSION_HEARTBEAT at 17:15:05
> Got a SESSION_HEARTBEAT at 17:15:06
> Got a SESSION_HEARTBEAT at 17:15:07
> Got a SESSION_HEARTBEAT at 17:15:08
> Got a SESSION_HEARTBEAT at 17:15:09
> Got a SESSION_HEARTBEAT at 17:15:10
> Got a SESSION_HEARTBEAT at 17:15:11
> Got a SESSION_HEARTBEAT at 17:15:12
> Got a SESSION_HEARTBEAT at 17:15:13
> Got a SESSION_HEARTBEAT at 17:15:14
> Got a SESSION_HEARTBEAT at 17:15:15
> Got a SESSION_HEARTBEAT at 17:15:16
> Got a SESSION_HEARTBEAT at 17:15:17
> Got a SESSION_HEARTBEAT at 17:15:18
> Got a SESSION_HEARTBEAT at 17:15:19
> Got a SESSION_HEARTBEAT at 17:15:20
> Got a SESSION_HEARTBEAT at 17:15:21
> Got a SESSION_HEARTBEAT at 17:15:22
> Got a SESSION_HEARTBEAT at 17:15:23
> Got a SESSION_HEARTBEAT at 17:15:24
> Got a SESSION_HEARTBEAT at 17:15:25
> Got a SESSION_HEARTBEAT at 17:15:26
> Got a SESSION_HEARTBEAT at 17:15:27
> Got a SESSION_HEARTBEAT at 17:15:28
> Got a SESSION_HEARTBEAT at 17:15:29
> Got a SESSION_HEARTBEAT at 17:15:30
>
>
>
>
>
> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> 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 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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>>
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>>
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Re: [Freeswitch-users] Heartbeat question

2009-10-09 Thread Diego Viola
Here's my heartbeat script now.

#!/usr/bin/env ruby

require 'rubygems'
require 'fsr'
require "fsr/listener/inbound"

def custom_channel_heartbeat_handler(event)
  puts "Got a SESSION_HEARTBEAT at #{Time.now.strftime('%H:%M:%S')}"
end
FSL::Inbound.add_event_hook(:SESSION_HEARTBEAT) {|event|
custom_channel_heartbeat_handler(event) }
FSR.start_ies!(FSL::Inbound, :host => "localhost", :port => 8021)

Thanks again.

Diego

On Fri, Oct 9, 2009 at 9:30 PM, Diego Viola  wrote:

> Here is on two seconds ;)
>
> Got a SESSION_HEARTBEAT at 17:17:13
> Got a SESSION_HEARTBEAT at 17:17:15
> Got a SESSION_HEARTBEAT at 17:17:17
> Got a SESSION_HEARTBEAT at 17:17:19
> Got a SESSION_HEARTBEAT at 17:17:21
> Got a SESSION_HEARTBEAT at 17:17:23
> Got a SESSION_HEARTBEAT at 17:17:25
> Got a SESSION_HEARTBEAT at 17:17:27
> Got a SESSION_HEARTBEAT at 17:17:29
> Got a SESSION_HEARTBEAT at 17:17:31
> Got a SESSION_HEARTBEAT at 17:17:33
> Got a SESSION_HEARTBEAT at 17:17:35
> Got a SESSION_HEARTBEAT at 17:17:37
> Got a SESSION_HEARTBEAT at 17:17:39
> Got a SESSION_HEARTBEAT at 17:17:41
> Got a SESSION_HEARTBEAT at 17:17:43
> Got a SESSION_HEARTBEAT at 17:17:45
> Got a SESSION_HEARTBEAT at 17:17:47
> Got a SESSION_HEARTBEAT at 17:17:49
> Got a SESSION_HEARTBEAT at 17:17:51
> Got a SESSION_HEARTBEAT at 17:17:53
> Got a SESSION_HEARTBEAT at 17:17:55
> Got a SESSION_HEARTBEAT at 17:17:57
> Got a SESSION_HEARTBEAT at 17:17:59
> Got a SESSION_HEARTBEAT at 17:18:01
> Got a SESSION_HEARTBEAT at 17:18:03
> Got a SESSION_HEARTBEAT at 17:18:05
> Got a SESSION_HEARTBEAT at 17:18:07
> Got a SESSION_HEARTBEAT at 17:18:09
> Got a SESSION_HEARTBEAT at 17:18:11
> Got a SESSION_HEARTBEAT at 17:18:13
> Got a SESSION_HEARTBEAT at 17:18:15
> Got a SESSION_HEARTBEAT at 17:18:17
>
>
>
> On Fri, Oct 9, 2009 at 9:27 PM, Diego Viola  wrote:
>
>> Thanks Anthony, this solved it. You rock :)
>>
>> My program now outputs:
>>
>> Got a SESSION_HEARTBEAT at 17:14:59
>> Got a SESSION_HEARTBEAT at 17:15:00
>> Got a SESSION_HEARTBEAT at 17:15:02
>> Got a SESSION_HEARTBEAT at 17:15:03
>> Got a SESSION_HEARTBEAT at 17:15:04
>> Got a SESSION_HEARTBEAT at 17:15:05
>> Got a SESSION_HEARTBEAT at 17:15:06
>> Got a SESSION_HEARTBEAT at 17:15:07
>> Got a SESSION_HEARTBEAT at 17:15:08
>> Got a SESSION_HEARTBEAT at 17:15:09
>> Got a SESSION_HEARTBEAT at 17:15:10
>> Got a SESSION_HEARTBEAT at 17:15:11
>> Got a SESSION_HEARTBEAT at 17:15:12
>> Got a SESSION_HEARTBEAT at 17:15:13
>> Got a SESSION_HEARTBEAT at 17:15:14
>> Got a SESSION_HEARTBEAT at 17:15:15
>> Got a SESSION_HEARTBEAT at 17:15:16
>> Got a SESSION_HEARTBEAT at 17:15:17
>> Got a SESSION_HEARTBEAT at 17:15:18
>> Got a SESSION_HEARTBEAT at 17:15:19
>> Got a SESSION_HEARTBEAT at 17:15:20
>> Got a SESSION_HEARTBEAT at 17:15:21
>> Got a SESSION_HEARTBEAT at 17:15:22
>> Got a SESSION_HEARTBEAT at 17:15:23
>> Got a SESSION_HEARTBEAT at 17:15:24
>> Got a SESSION_HEARTBEAT at 17:15:25
>> Got a SESSION_HEARTBEAT at 17:15:26
>> Got a SESSION_HEARTBEAT at 17:15:27
>> Got a SESSION_HEARTBEAT at 17:15:28
>> Got a SESSION_HEARTBEAT at 17:15:29
>> Got a SESSION_HEARTBEAT at 17:15:30
>>
>>
>>
>>
>>
>> On Fri, Oct 9, 2009 at 4:02 PM, Anthony Minessale <
>> anthony.miness...@gmail.com> wrote:
>>
>>> 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 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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>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>> googl

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
wrote:

>  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_presen

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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-- 
View this message in context: 
http://www.nabble.com/Recorded-file-as-voicemail.-tp25751158p25830785.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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

2009-10-09 Thread Jason White
Nagalenoj  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] libpri_span vs prispans

2009-10-09 Thread lakshmanan ganapathy
Where can I found entire configurations for a libpri spans. Right now I've
only 4 things in my libpri configuration. They are Switchtype, Node,
Contect, Dialplan, which is given as example in wiki.

But I think libprispan has more configuration. Where can I find those
things?



On Fri, Oct 9, 2009 at 11:27 PM, Michael Collins  wrote:

>
>
> On Fri, Oct 9, 2009 at 5:57 AM,  wrote:
>
>> lakshmanan ganapathy  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] 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 

> Nagalenoj  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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