Brian West wrote:
> You still didn't answer my question. How are you trying to do this
> from the dialplan.
>
> /b
>
> On Feb 12, 2009, at 8:08 AM, pauld wrote:
>
>
>> Yes I am using 5.1, I haven't done anything special other than
>> followed
>> wiki and
On Thu, Feb 12, 2009 at 3:31 PM, Luis F Urrea wrote:
> Heres pastebin of the A-leg
>
> http://pastebin.com/m6731913d
>
>
> On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins wrote:
>>
>> Pastebin the whole file so that we can see it in context...
>> -MC
>>
>> On Thu, Feb 12, 2009 at 2:50 PM, Luis F
Heres pastebin of the A-leg
http://pastebin.com/m6731913d
On Thu, Feb 12, 2009 at 5:00 PM, Michael Collins wrote:
> Pastebin the whole file so that we can see it in context...
> -MC
>
> On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote:
> > On our test calls we haven't been able to correlat
Done, that was easy, unlike FS :)
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 12 February 2009 23:01
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS
On Thu, Feb 12, 2009 at 2:58 PM, Nik Middleton
wrote:
> Hi,
>
> Not sure who updates the WIKI, but it's wrong on collectinput for the
> example. In the call, dtmf needs quotes, ie "dtmf"
Thanks for the heads up. Actually, YOU can update the wiki. If you
want me to do so I will be happy to.
-MC
YOU DO! ;) Its a user edited content portal.
/b
On Feb 12, 2009, at 4:58 PM, Nik Middleton wrote:
>
> Not sure who updates the WIKI, but it's wrong on collectinput for the
> example. In the call, dtmf needs quotes, ie "dtmf"
___
Freeswitch-users m
Pastebin the whole file so that we can see it in context...
-MC
On Thu, Feb 12, 2009 at 2:50 PM, Luis F Urrea wrote:
> On our test calls we haven't been able to correlate times from the A leg
> with times from the B leg.
>
> I would expect something as A-leg(duration)=
> B-leg1(duration)+B-leg2(d
Hi,
Not sure who updates the WIKI, but it's wrong on collectinput for the
example. In the call, dtmf needs quotes, ie "dtmf"
Correction is session.collectInput( mycb, "dtmf", 8000 );
Without it you get
[ERR] voice.js:70 mod_spidermonkey() ReferenceError: dtmf is not
defined
if ( session.ready
On our test calls we haven't been able to correlate times from the A leg
with times from the B leg.
I would expect something as A-leg(duration)=
B-leg1(duration)+B-leg2(duration)
Also the tag within tag does not seem to be in epoch
microseconds. so it does not seem that's where i should be look
On Thu, Feb 12, 2009 at 2:26 PM, Nik Middleton
wrote:
> Just been chatting to Ken Rice, his view (and he may be mistaken) is
> that it should fire the call back event in much the same way as DTMF
> does, however, it's not working. I used to develop with C/C++ for about
> 10 years, but that was 12
This is prob. why we don't see this crazy stuff on CentOS since the
compiler is 4.1.2
/b
On Feb 12, 2009, at 4:34 PM, Andy Spitzer wrote:
> Possibly. A recent (last year?) GCC change caused some order of
> operations to change, and so code that inadvertently relied on the
> previous behav
Woof!
On Thu, 12 Feb 2009 17:20:18 -0500, Anthony Minessale
wrote:
> So I wonder what about the distro you are using that makes the same exact
> code not work?
> maybe the GCC ?
Possibly. A recent (last year?) GCC change caused some order of operations to
change, and so code that inadverten
Just been chatting to Ken Rice, his view (and he may be mistaken) is
that it should fire the call back event in much the same way as DTMF
does, however, it's not working. I used to develop with C/C++ for about
10 years, but that was 12 years ago. Very rusty. However, I'm going to
look at the star
That's scary
So I wonder what about the distro you are using that makes the same exact
code not work?
maybe the GCC ?
On Thu, Feb 12, 2009 at 2:07 AM, Helmut Kuper wrote:
> Hi Anthony,
>
> hm... on centos5 it works fine. No problems, no warning, no crash.
>
> regards
> Helmut
>
> On 11.02.20
What SVN rev?
/b
On Feb 12, 2009, at 3:41 PM, Ali Al-Rubaie wrote:
> Hi,
>
> How can the default value of "realm" be changed? I had changed the
> command:
>
>
>
> in the file internal.xml but FS still uses the server IP address as
> the challenge realm.
>
> Thanks in advance!
>
__
Hi all,
We are writing a xml_cdr parser to load CDRs in SQLite. We are interested in
logging times for both A leg and B leg so that transfers are reported as
individual calls with accurate timing. eg Inboud call to AA lasted 14
seconds then call to operator 20s and then call to actual extension 5m
On Thu, Feb 12, 2009 at 12:49 PM, Nik Middleton
wrote:
> That makes sense, though could it not have a call back mechanism similar
> to DTMF detect?
>
It probably could but the mod's author was using it exclusively from
event socket. I personally added the channel variable code for the
sake of tes
Hi,
How can the default value of "realm" be changed? I had changed the command:
in the file internal.xml but FS still uses the server IP address as the
challenge realm.
Thanks in advance!
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Freeswitch-user
the entire sdp is available as a variable (route the call to the info app to
see the variables)
so if you have inbound-late-negotiation set to true on the sip profile
then you can use a regex or a script to set absolute_codec string before you
answer.
On Thu, Feb 12, 2009 at 8:06 AM, ivdreg ivdre
transcoding from PCMU (g711) to PCM (raw signed linear) the format that
cepstral speaks.
On Wed, Feb 11, 2009 at 10:38 PM, pauld wrote:
> The issue was resolved by creating symlinks to cepstral libs in FS lib
> directory. I tried that on 1.0.3, but most probably it would work on
> 1.0.2 as wel
deflect takes one arg. and that isn't one. Try a SIP uri... not a
sofia/ string. ie sip:b...@host:5080
/b
On Feb 12, 2009, at 2:44 PM, jonathan augenstine wrote:
I am trying to use the deflect command to transfer an inbound call.
The call is established and the command seems to complete
That makes sense, though could it not have a call back mechanism similar
to DTMF detect?
I'm still not sure how I could use it even in an event socket. I plan
to call my js IVR script using a socket, but that has the originate call
in it which is nice and simple, but I'm unsure how I could abort
I am trying to use the deflect command to transfer an inbound call. The
call is established and the command seems to complete successfully. If I
bump up the sofia logging, I see the command executed in the LUA script and
I see output from the console from sofia that seems to indicate the deflect
Thinak you, William and Brian
I got it now, I didn't know file was a command before because it didn't come
with my CentOS installation. Now I have installed the file package and able
to see the file info.
Thanks again
On Thu, Feb 12, 2009 at 12:31 PM, Brian West wrote:
> Well when I do this:
Well when I do this:
r...@taz [Thu Feb 12 02:20 PM] /usr/src/freeswitch.trunk
<13>:file /usr/local/freeswitch/bin/freeswitch
/usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, AMD
x86-64, version 1 (SYSV), for GNU/Linux 2.6.9, dynamically linked
(uses shared libs), for GNU/Linu
If you run in your shell:
file /usr/local/freeswitch/bin/freeswitch
as Brian suggested it will return something like what I got below:
/usr/local/freeswitch/bin/freeswitch: ELF 64-bit LSB executable, x86-64,
version 1 (SYSV), for GNU/Linux 2.6.8, dynamically linked (uses shared
libs), not stripped
I run /usr/local/freeswitch/bin/freeswitch
but I don't see a place where it says it's 32bit or 64bit.
at the end of the initial script, I do see a version statement though.
FreeSWITCH Version 1.0.trunk (exported) Started.
Is there other ways to check if it's 32bit or 64bit?
On Wed, Feb 11, 2009 at
> I'm trying to get VMD running in js, does anyone have an example of how it's
> called?
http://wiki.freeswitch.org/wiki/Mod_vmd
You need to use the event socket because that is the way VMD is
designed. If called from the dialplan it will set a channel variable
but that isn't of much use in a real
On Wed, Feb 11, 2009 at 8:31 AM, Saeed Ahmed wrote:
> Hi List,
>
> Is there any open source call center tool available which works with FS?
Check this out:
http://opencsm.org/wiki/index.php/Spice_Telephony
-MC
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Freeswit
Hi Guys,
I'm trying to get VMD running in js, does anyone have an example of how
it's called?
If I try
session:execute("vmd");
I get an error
Regards
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http://
>
> Is this running on 64 bit os or 32?
A 64bit , Windows 2008 Server.
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Sorry, should have said this was in js
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 12 February 2009 18:08
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freesw
On Thu, Feb 12, 2009 at 10:07 AM, Brian West wrote:
> Dialplan isn't for writing IVR's... doing so is against the design of
> FreeSWITCH.. you can do simple things in dialplan but more complex
> stuff needs to be in a language.
Or create an IVR and send the call there from the dialplan. You can d
Dialplan isn't for writing IVR's... doing so is against the design of
FreeSWITCH.. you can do simple things in dialplan but more complex
stuff needs to be in a language.
/b
On Feb 12, 2009, at 12:01 PM, Shelby Ramsey wrote:
> Nik,
>
> I'm not sure if this is the right way ... but I use
> a
Nik,
I'm not sure if this is the right way ... but I use application="read"
data="0 1 /path/silence.wav var 1000 #
I'm sure there is a better way ... but this seems to work for me.
SDR
On Thu, Feb 12, 2009 at 11:51 AM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> HI,
>
>
>
> I
Dialplan or language method...btw if you're on IRC its better to ask
there.. faster response... ;)
/b
On Feb 12, 2009, at 11:51 AM, Nik Middleton wrote:
HI,
Is there an equivalent function in FS to waitforexten ? Closest
I’ve seen is collectInput?
Right now I’m using stream file, whic
HI,
Is there an equivalent function in FS to waitforexten ? Closest I've
seen is collectInput?
Right now I'm using stream file, which is ok if they hit a digit before
stream ends, but I want them to have a certain period after the file is
played to hit a button.
Regards,
_
Hi Guys,
I've been experimenting with originate_timeout and progress_timeout as
follows.
However, shouldn't the timeout trigger a 408 Request Timeout instead of 480
Temporary Failure if no Provisional response received?
Just curious, it seems to make sense to me.. but maybe SIP gods see
di
Bang on,
Thanks
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 12 February 2009 01:10
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Call accountin
You can change the config files on disk and then issue reloadxml or
use mod_XML_curl
Mike
On Feb 12, 2009, at 6:05 AM, Rene Pankratz wrote:
> Hello,
> we want to use mod_pa as a softphone, that registers to a
> SIPregistrar.
> But the username and password need to be changed over time witho
On Thu, Feb 12, 2009 at 6:11 AM, Fred wrote:
> Hello
>
> I successfully used Asterisk to build a voice server for our SOHO
> business. I did read the article comparing Asterisk to Freeswitch,
> but I have a couple of questions:
>
> 1. What are the decisive reasons that would justify taking a look
If using gayeway it should already do this.
On Feb 12, 2009, at 3:34 AM, Helmut Kuper
wrote:
> Hi,
>
> any ideas how to get FS's BYEs authenticated ?
>
> On 11.02.2009 13:41, Helmut Kuper wrote:
>> Hello,
>>
>> my FS is connected to my SIP-DDI softswitch, which requires all SIP
>> requests sen
Is this running on 64 bit os or 32?
On Feb 12, 2009, at 4:23 AM, Public Dump wrote:
>> OK does it work now? We have tested this on various windows installs
>> among the team here and not seeing this issue... it was a known issue
>> back in Nov. or Dec. but thats long been fixed.
>
> No, the pro
On Thu, Feb 12, 2009 at 8:11 AM, Fred wrote:
> Hello
>
> I successfully used Asterisk to build a voice server for our SOHO
> business. I did read the article comparing Asterisk to Freeswitch,
> but I have a couple of questions:
>
> 1. What are the decisive reasons that would justify taking a look
You could but I think you want to stream RTP to a multicast it would
be better off building an rtp format mod so you can record rtp://
x.x.x.x:5000 and play from rtp://y.y.y.y:5000
/b
On Feb 12, 2009, at 10:12 AM, Sluschny, Thomas wrote:
Hi Brian,
i thought if i can stream from portaudio
Hi Brian,
i thought if i can stream from portaudio it is almost the same with
streaming from file,
so it should working already now.
Is this not the design idea of channels and media to do so?
regards,
thomas
PS: sry for improper formatted mail, i cant reply at the moment and have
to copy ma
Hello
I successfully used Asterisk to build a voice server for our SOHO
business. I did read the article comparing Asterisk to Freeswitch,
but I have a couple of questions:
1. What are the decisive reasons that would justify taking a look at
Freeswitch? What makes it a better option?
2. I'd l
Hi all,
Can I ask 2 questions about codec negotiation:
1. Is it possible Freeswitch to work negotiate codecs between two phones as
it is described below.
INVITE from A with some codecs in SDP ---> Freeswitch rewrites codec
preference according absolute_codec_string but exclude all codecs not
offe
No, I have not made any changes to reflect anything you asked about.
instant_ringback=true is designed to send artificial ringback to the a leg
while it's executing the bridge app.
it will be meaningless to you if you do not use it with the bridge
application
On Thu, Feb 12, 2009 at 1:39 AM, Denn
Are you calling via a gateway?
/b
On Feb 12, 2009, at 2:34 AM, Helmut Kuper wrote:
> Hi,
>
> any ideas how to get FS's BYEs authenticated ?
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You could store the data in globals and then restart the profiles via
XML PRC. ie global_setvar, reloadxml, sofia profile blah restart.
/b
On Feb 12, 2009, at 5:05 AM, Rene Pankratz wrote:
> Hello,
> we want to use mod_pa as a softphone, that registers to a
> SIPregistrar.
> But the usern
You still didn't answer my question. How are you trying to do this
from the dialplan.
/b
On Feb 12, 2009, at 8:08 AM, pauld wrote:
> Yes I am using 5.1, I haven't done anything special other than
> followed
> wiki and then the advice given here to create symlinks in FS lib dir
> to all
>
esf is for multi cast paging... it currently won't let you play
files... we would have to create a multicast playback application.
/b
On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote:
Hi,
i want to stream a file per IP multicast with mod_esf.
I can stream IP multicast with:
pa call
I didn't mean to touch a nerve there, I think you mis-interpreted 100%
of my original post. You may also have me confused with some other
Nik, as I contribute as much or more than I request . There is a
nother Nik on here that are probably referring to. I am nikko from
#freeswitch. I have made
Yes I am using 5.1, I haven't done anything special other than followed
wiki and then the advice given here to create symlinks in FS lib dir to all
cepstral libs. I have cepstral libs in a standard location /opt/swift/lib.
I have given an example extension I used for testing earlier in this thread
Hi,
i want to stream a file per IP multicast with mod_esf.
I can stream IP multicast with:
pa call stream XML
and in XML dialplan:
and i can also play files with 'playback' app,
BUT: how can put these 2 things together?
May be its trivial, but i cant get it make w
Hi Mike,
at least for incoming calls this shouldn't be too brutal, cause far end
seems to know that the channel should be free otherwise it never would
allocate it. By now the hack works at least for me quite good. Nobody
from AVAYA side moaned about it, yet. But I have to wait one or two
further
Hello,
we want to use mod_pa as a softphone, that registers to a SIPregistrar.
But the username and password need to be changed over time without
restarting freeswitch.
Currently we are using XML/RPC to control the call functions. So it
would be best (if possible) to use it also for changing reg
> OK does it work now? We have tested this on various windows installs
> among the team here and not seeing this issue... it was a known issue
> back in Nov. or Dec. but thats long been fixed.
No, the problem is still there.
I have tested it on a Core AMD 32bit AMD machine = everything is fine.
Hello Brian / Everyone,
Like Nik and Nicolas, I created a openvz box to test things in a 'near
production' environment. The box does only take 'test' calls, ie it
never saw more that a few calls at a time.
The design was 100% openser/opensips/kamailio but I since replaced the
pstn gateways
Hi,
any ideas how to get FS's BYEs authenticated ?
On 11.02.2009 13:41, Helmut Kuper wrote:
> Hello,
>
> my FS is connected to my SIP-DDI softswitch, which requires all SIP
> requests sent by a registered SIP account to be authenticated. I found
> that when FS sends a BYE FreeSWITCH ignores the
Hi Anthony,
hm... on centos5 it works fine. No problems, no warning, no crash.
regards
Helmut
On 11.02.2009 16:29, Anthony Minessale wrote:
> I am highly suspicious of the ubuntu.
> you are using a prerelease of gcc that we have already found at least
> 1 bug.
>
> we tried the file on our box an
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