Hi João, thanks for the reply. But I don't quite get you.. Could you please
elaborate a little bit? I tried installing libtiff and upgrading FS to the
latest revision, but still the same error.
Here's how I normally update FreeSwitch: *make clean svn up
./bootstrap.sh ./configure make install
Problem solved. It's due to the lack of definition in tones.conf. In case
anyone else need it, here's the tone plan for Singapore.
[sg]
generate-dial = v=-7;%(1000,0,425)
detect-dial = 425
generate-ring = v=-7;%(2000,4000,425)
detect-ring = 425
generate-busy = v=-7;%(750,750,425)
detect-busy =
Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding...
On Tue, Dec 8, 2009 at 2:50 AM, Brian West br...@freeswitch.org wrote:
We can ONLY hope someone will do this and BSD/MIT the library and NOT
GPL it... if they GPL it then we'll have to have someone write it all
over
Hello,
I have some black hole understading how to debug Freeswitch. In fs_cli I
do sofia debug all 7 and indeed get a lot of debugging messages on the
console; however, the logfiles get only Critical messages. Where do I define
which messages go to the logfile?
And in a related topic: I've
I got the combination Lua with direct access to the core Sqlite database to
work. Hurray, maybe I'm not as stupid as A.M II hints...
The problem was that Lua did not like:
require luasql.sqlite
env = luasql.sqlite()
con = assert(env:connect(/usr/local/freeswitch/db/core.db))
After
Hello
I'd like to install OpenZAP so I can use a TDM card with Freeswitch, but I'm
getting a software error althought the TDM card seems detected (lspci -v
OK). Dahdi was successfully compiled from source code.
Is it OK to just install Dahdi 2.2.0 without Asterisk before going ahead
with
Fred-145 codecompl...@free.fr said:
Has someone succesfully installed Dahdi without Asterisk?
Of course, and it's working like a charm. DAHDI is a driver. It doesn't
care what software uses it. We're using DAHDI with a TE110P PRI T1 card.
What is in /proc/dahdi? If it shows 1, what do you see if
That would binary only, not 64 bit Linux .
On Dec 8, 2009, at 9:17 AM, Kevin Green ke...@johnnyvoip.com wrote:
It seems you can get a copy of either the binaries or the source by
doing the following:
Review execute SILK Agreement - attached. NOTE - please add your
Skype login to this
Thanks Russel for the tip. After more googling, I ended up figuring that
/etc/dahdi/modules had to contain the list of drivers to load.
For those interested, here's how to compile and install Dahdi (which doesn't
need Asterisk at all, unlike some docs on the Net seem to imply due to
references
And you didn't open a Jira about this? These are the kinds of issues
that you should report so we can fix them... sitting on them and NOT
reporting them only delays the 1.0.5 release.
/b
On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote:
Changing the core db into a MySQL via ODBC caused some
Their site (https://developer.skype.com/silk) specifies that they will
provide the source, which as you say may not be 64-Bit compatible but could
likely be tweaked to work. I think you just need to be specific in that you
want a source copy not a binary copy of the codec.
Regards,
Kevin Green
Fred-145 codecompl...@free.fr said:
5. vi modules:
wcfxo
wctdm
dahdi
You only need one of the modules above, if you have one card. I don't see
a dahdi module listed in the file here.
8. ls -la /proc/dahdi/
You should be able to cat the file in that directory for more information.
--
Fred-145 codecompl...@free.fr said:
For those interested, here's how to compile and install Dahdi
It would be helpful to others if you add the results of your efforts to
the wiki.
--
Russell Mosemann
Concordia University, Nebraska
Russell.Mosemann wrote:
You only need one of the modules above, if you have one card. I don't see
a dahdi module listed in the file here.
Yup, turns out wcfxo is needed for the X10xP card, while wctdm is needed for
Digium cards. As for dahdi, maybe wcfxo/wctdm loads the dahdi module
They provide you with a 32 bit library, with the header files to link
with it.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 8-Dec-09, at 9:39 AM, Kevin Green wrote:
Their site (https://developer.skype.com/silk)
We have as of yet been unable to obtain source and we have been in
very close contact with skype all the way up to the lead technical and
business people on this project. We would of course welcome access to
the source but we have as of yet not been able to get a copy
Mike
On Dec 8,
Hello,
is there a way to manually force a presence status update?
In our scenario we have a Freeswitch cluster. As phones sometimes
register on one and one time on another machine via the load balancer,
we cannot dial via user/exten. Instead we dial each phone by it's
register string via
Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes.
The Asterisk boxes are individual hosted PBXs but they are configured
with identical software. This a x86_64 CentOS 5.4 system. I've tried
1.0.4 and the latest svn with the same results. Basically Freeswitch
registers
I have resubmitted our request for the source.
/b
On Dec 8, 2009, at 9:58 AM, Michael Jerris wrote:
We have as of yet been unable to obtain source and we have been in
very close contact with skype all the way up to the lead technical
and business people on this project. We would of
The fun part comes when you try to link that 32bit .a file into a
64bit so file.
:P
/b
On Dec 8, 2009, at 9:49 AM, Mathieu Rene wrote:
They provide you with a 32 bit library, with the header files to
link with it.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
You definitely need to use the settings in combination for what you are trying
to do. Can you explain a bit more what you want to do in what conditions and
maybe we can suggest how to accomplish this. NORMAL_CLEARING is not a failure,
so it can continue on after the bridge unless you specify
set hangup_after_bridge=true
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 7-Dec-09, at 1:31 PM, Peter P GMX wrote:
I have a Problem with continue_on_fail.
I have setup a hunt group
action application=set
If this issue continues after another update and re bootstrap/configure, please
open up a bug on jira.freeswitch.org under build system, assign to me, and
attach the config.log and config.status file from the root of your freeswitch
src dir.
Mike
On Dec 7, 2009, at 2:39 PM, Anthony Minessale
Please re-test this with svn trunk of freeswitch and if it is still the case
open up a bug on jira.freeswitch.org in the build system catagory assigned to
me and attach the config.log and config.status from the libs/esl dir to the bug.
Mike
On Dec 7, 2009, at 1:34 PM, Kendall Stauffer wrote:
I'm using FreeSWITCH in front of Asterisk without any issue.
Stick with the latest trunk. Can you set your loglevel to debug and
pastebin your log?
Here are some additional tips to help us help you :)
http://wiki.freeswitch.org/wiki/Reporting_Bugs
Rob
On Mon, Dec 7, 2009 at 3:34 PM, Spencer
We will really need debug logs and sip traces to be able to figure out what
exactly is going on here.
Mike
On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:
Sorry no, apart from the fact that I was seeing the hangup.
I’m wondering if this a bandwidth congestion issue. Is there anyway
If you can off list provide me with remote login information to this box I can
troubleshot the issue.
Mike
On Dec 8, 2009, at 4:09 AM, Jingwei Yang wrote:
Hi João, thanks for the reply. But I don't quite get you.. Could you please
elaborate a little bit? I tried installing libtiff and
We could check it out for you if you want to contact me and give me ssh
access.
Or I can provide the instructions
get it into the 100% cpu usage state then do the following without stopping
FS.
1) run top -H and sort so all the FS threads are at the top and screen cap
it so we can see which
One last bit of free consulting advice for you:
You are again being rude because you want us to work for you for free.
The code is free sir, the support here is voluntary and based on our
willingness to help and comments like that are all it takes to get us to
ignore you completely.
On Tue, Dec
I changed the name of key to ikey in trunk.
Mike
Changing the core db into a MySQL via ODBC caused some problems even after it
seemed to work. For instance, console help caused an error with an error
description indicating that a SQL SELECT query including the reserved word
key has been
The best way to solve this is probably to share the db for presence and
registration between those boxes. If you take a look at the default configs
the settings should be commented there.
Mike
On Dec 8, 2009, at 11:02 AM, Peter P GMX wrote:
Hello,
is there a way to manually force a
On Tue, Dec 8, 2009 at 3:46 AM, Jon Bruel j...@consiglia.dk wrote:
I got the combination Lua with direct access to the core Sqlite database
to work. Hurray, maybe I’m not as stupid as A.M II hints…
Tsk tsk! He didn't actually hint that you were stupid - all he said was
that doing ODBC and
Otis wrote:
div class=moz-text-flowed style=font-family: -moz-fixedMichael
Collins wrote:
On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah
ab...@greatiam.com mailto:ab...@greatiam.com wrote:
Pardon me if this has been addressed already.
How does one go about having in the
Brian West br...@freeswitch.org said:
The fun part comes when you try to link that 32bit .a file into a
64bit so file.
That would require a dual-core processor. One core would be 32 bit and
the other core would be 64 bit. ;-)
--
Russell Mosemann
Well the fun part is you can't link them. :P
/b
On Dec 8, 2009, at 10:38 AM, russell.mosem...@cune.org
russell.mosem...@cune.org
wrote:
That would require a dual-core processor. One core would be 32 bit and
the other core would be 64 bit. ;-)
--
Russell Mosemann
Point taken Anthony. Naturally you are not going to work for me for free.
But I'm a bit confused about the statement that I'm rude. That's not my
purpose to be. And I certainly do hope that this is not just a question of a
cultural clash between an elderly man with a Phd in black holes from a
Hello,
I tune in the presence freeswitch and linksys spa962 932 I had a few
qusteions:
1) if $ PROXY specified domain name and not ip the phone records. But
all the buttons on spa932 blinking orange indicating that no subscriptions.
phone logs like this:
Call-ID: 76e0f816-9617a...@192.168.0.100
Greetings,
The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH
1.0.5 pre-release version. Please check out the release
announcementhttp://www.freeswitch.org/node/220.
Let's all get updated as soon as possible. Also, please report bugs right
away and follow up when the
would not be able to even guess without some data to examine.
On Tue, Dec 8, 2009 at 12:14 PM, Spencer Thomason
spen...@5ninesolutions.com wrote:
Hmm.. It doesn't seem to be a problem with Asterisk 1.6.0.13. Asterisk
1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this
Let's see if we can beat Duke Nukem Forever!
On Tue, Dec 8, 2009 at 12:19 PM, Michael Collins m...@freeswitch.org wrote:
Greetings,
The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH
1.0.5 pre-release version. Please check out the release
For those interested, here's how to compile and install Dahdi (which doesn't
need Asterisk at all, unlike some docs on the Net seem to imply due to
references to /etc/asterisk/*.conf):
I understand that Some Debian based distro's have Dahdi in their repo's making
it
simple, but not many know
For reference, here is the AstLinux kernel config for the ALIX:
http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/target/device/alix/linux.config?view=markup
We've got what I consider to be excellent support for the ALIX - most
of the developers use them and they are very popular in the
Dear list,
Some Nec phones sends DTMF RFC2833 with payload 101 during the call, but have
negotiated a different one on SDP.
When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 we notice
this phone sends the following INVITE packet and RTP packets:
Best option for you is to use 96 in the sofia profile you're using to
talk to these broken devices.
/b
On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote:
Dear list,
Some Nec phones sends DTMF RFC2833 with payload 101 during the call,
but have negotiated a different one on SDP.
Hey you guys, I know this isn't the right place for this, but I have been
working with asterisk for 5 years now, and just got freeswitch working (on
windows, not os x yet).
All I can say is AWESOME --- thanks so much
From: freeswitch-users-boun...@lists.freeswitch.org
Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I don't
know how to fix it properly.
Looking in the Dialplan directory I see the following:
default (dir)
default.xml
features.xml
public (dir)
public.xml
Under the default dir the webinterface has created the
Our plan for 1.0.5 is that we will also have rpm and deb packages for many
distros on our own repo. Stay tuned. This has been another major reason for
the delay in 1.0.5.
Mike
On Dec 8, 2009, at 1:26 PM, Joseph L. Casale wrote:
For those interested, here's how to compile and install Dahdi
Anthony and Michael,
I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs
that Anthony told me to turn on. I put the results up in the PasteBin.
Best Regards,
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Monday, December 07, 2009
Here is the Pastebin Link: http://pastebin.freeswitch.org/11432
Thanks,
Jerry
_
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Tuesday, December 08, 2009 12:35 PM
To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org'
Subject: RE: [Freeswitch-users] FS Machine
Can you copy the address of the pastebin so that people can see it? After
you hit the Send button, the address is posted back at the top of your
browser, like:
http://pastebin.freeswitch.org/11441
From: freeswitch-users-boun...@lists.freeswitch.org
are you using more than one profile here?
if so you have to repeat the siptrace on command for each one.
This trace makes little sense to me because I think half of it is missing.
but you can see several packets coming in like 20 times each which means you
have some kind of nat or network problem
I dont think there are any supported hw for bsd, there are legacy sangoma
and zaptel drivers floating around but they are not supported by the
vendors.
On Tue, Dec 8, 2009 at 3:25 PM, Orien Love or...@tx.rr.com wrote:
I am looking for a 4 port FXO card to use with my PfSense installation
of
I have 2 Rhino cards for sale if anyone needs one. They are both Best Offer. I
have a R2T1-EC and a R24FXX-EC with 12 dual FXS modules. Both have never been
used more than a few times for testing purposes. Both cards work fine and are
guaranteed not to be DOA.
Reece Savage
Information
have you created Extension 1002?
-nandy
On Wed, Dec 9, 2009 at 3:20 AM, mailinglist mailingl...@fribert.dk wrote:
Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I
don't know how to fix it properly.
Looking in the Dialplan directory I see the following:
On Wed, Dec 9, 2009 at 7:56 AM, Kendall Stauffer k...@ksac.com wrote:
Hey you guys, I know this isn’t the right place for this, but I have been
working with asterisk for 5 years now, and just got freeswitch working (on
windows, not os x yet).
All I can say is AWESOME --- thanks so much
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