Sorry, i didn't visited the Jira link you mentioned. Now i know the issue
and I have replied it there.
Thank you.
On Tue, Jun 9, 2009 at 10:45 PM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
Hi all,
there are problems for mod_skypiax in recent centos, with more than a
handful of
Open a Jira.
http://jira.freeswitch.org/
On Tue, Jun 9, 2009 at 4:26 AM, Yuriy Ivzhenko yivzhe...@mksat.net wrote:
Some time ago mod_nibblebill was set variable nibble_total_billed after
hangup.
But after last few updates of module this variable is no more sets.
Somebody else have this
Ciao Muhammad!
What a good news!
Centos is the most stable and performing platform for FS, so I would
really love to test and document on the wiki how to have a stable
centos mod_skypiax installation.
I'll find out your kernel ( Kernel 2.6.18-92.1.22.el5.centos.plusPAE
), and begin to test. In
Thanks. I didn't make any special arrangements for FS or Skypiax to work on
CentOS 5.3. I only enabled CentOS Plus yum repository and then install PAE
kernel with following commands,
root ~# yum update
root ~# yum install kernel-PAE
i installed PAE kernel just because i wanted to increase System
I am not using 64bit CentOS 5.3, I have 32bit CentOS but with PAE enable
kernel. I have heard 64bit ALSA drivers have bad sound issues, but never
used it personally.
As for source code of my modifications, i made those change to develop a
customized commercial solution for large European firm, so
Thanks a lot Muhammad,
and please let that firm know the advantages of having customizations
included into mainstream ;-).
OK, I will try 32 bit too, and see if there is differences.
So, you started fom a fresh install of centos5.2, then you installed
the PAE kernel. Is this right?
Pay
I am glad to share the patch to enable dynamic Skypiax interfaces in FS.
Please do note that however, that i started working on it on May 22, 2009.
So any officaily changes made to mod_skypiax.c since then will not appear in
it and will be lost if you apply this patch blindly.
I request Giovanni
You are welcome.
Let me elaborate my setup here,
I have two machines, one for development, this is basically my lenovo 3000
N200 laptop, it has following specs,
1. Intel 1.6 GHz with 1GB RAM.
2. CentOS 5.3 with Kernel 2.6.18-128.1.6.el5.
3. FS SVN revision Revision ID 13613.
root ~# uname -a
On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote:
Hi All,
I just finished installing freeSwitch and Skypiax. And I'm able to
use skype api directly via the sk command like the following:
freeswi...@localhost.localdomainsk console skypiax1
freeswi...@localhost.localdomainsk CALL userAAA
It
There was no 0.0.0.0 anywhere. I used vi. I'll rotate the logs and restart
FS without nc later today and report back.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 6:05 PM
To:
the default alias was removed from the default configs last week, so
new configs don't have this anymore.
On Jun 10, 2009, at 10:54 AM, Max Bridgewater wrote:
Thanks,
the first variant doesn't work for me. Any idea?
I changed it to:
originate sofia/internal/1...@192.168.10.103 park()
http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
If you're calling a locally registered user... you need to use user/
u...@domain which uses dial-string from the params on the user or
directory. Or sofia/profile/user%domain
/b
On Jun 10, 2009, at 9:54 AM, Max
Thanks Folks; I'm making progress. The following origination string does
make my non-registered SJPhone ring:
{origination_caller_id_number=2000}sofia/external/s...@192.168.50.67
But why isn't it caught by the following extension?
include
extension name=myextension
condition
http://wiki.freeswitch.org/wiki/Dialplan_XML
break=on-true ?
On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote:
Thanks Folks; I'm making progress. The following origination string
does make my non-registered SJPhone ring:
Hello Everybody,
I have a large project coming up. I'm interested in using Freeswitch
instead of SER and Asterisk.
What is the current status of Freeswitch? Can I safely use it in a
large scale commercial environment? How active is the Freeswitch
developer community?
I am concerned that
Hi Paul,
I'll tell you this. We (Teliax) made the decision to use FreeSWITCH
instead of Asterisk in late 2008 and we haven't looked back since. We're
a fairly large SIP/IAX provider with four POPs located throughout the
US, each one running FS.
The dev community surrounding FS is excellent,
Well, i assume break=on-true means, that if this extension is matched
then execute its actions and stop there. That would correspond to what i'm
trying to do. Anyway i removed this attribute and still nothing is being
sent to the socket.
Let me give you more context. This extension is put in a
Hello all,
The documentation has this to say on start_dtmf_generate:
As an example, Placing this in the Dialplan prior to bridging a call,
will allow a phone set to rfc2833 ( info ) to send DTMF tones (in-band )
out to the recipient (IVR) or Auto Attendants. Thus changing the
outgoing routing
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
today I updated to latest trunk. Everything compiled well. FS starts up
without errors. Calls are executed successfully.
But after a few minutes with very few call activity I get this lines on
console:
2009-06-10 18:21:16.52527 [ERR]
Yes.
/b
On Jun 10, 2009, at 12:34 PM, Ben Jones wrote:
Does this mean if a user is set for rfc2833 OR info that FS will
generate inband tones to send out?
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Yep, we tracked it down to the originating pbx ( a cisco call manager) which
had a 12 hour limit on outbound calls, thanks for your help.
D-
- Original Message -
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, June 9, 2009 4:06:01
I will clarify the point on the wiki since it is a bit inaccurate
-MC
Sent from my iPhone
On Jun 10, 2009, at 10:40 AM, Brian West br...@freeswitch.org wrote:
Yes.
/b
On Jun 10, 2009, at 12:34 PM, Ben Jones wrote:
Does this mean if a user is set for rfc2833 OR info that FS will
generate
need to rebootstrap.
/b
On Jun 10, 2009, at 1:30 PM, Nik Middleton wrote:
Hi Guys,
Ran make current today, and am getting the following errors. I ran
bootstrap and configure, but still get these messages.
Any ideas ? Looks like I’m now missing some libraries
Regards,
configure:
Hi Guys,
Ran make current today, and am getting the following errors. I ran
bootstrap and configure, but still get these messages.
Any ideas ? Looks like I'm now missing some libraries
Regards,
configure: configuring in libs/pcre
configure: running /bin/sh './configure.gnu'
your svn update failed,
rm -rf libs/pcre svn update ./bootstrap.sh ./configure
make current
On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote:
Hi Guys,
Ran make current today, and am getting the following errors. I ran
bootstrap and configure, but still get these messages.
Any
Friends,
ClueCon 2009 is fast approaching! We are definitely looking forward to
seeing everyone in Chicago this August. If you haven't finalized your plans
to attend, please do so right away. Time is running out! The early-bird
registration special of $499 per person will expire at the end of
Thanks, that did the trick
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 10 June 2009 19:50
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Is it OK to remove the example.com gateway? I removed the example.xml files
in sip_profiles/external and sip_profiles/internal and changed the
default_provider from example.com to myprovider.com.
But I still see myprovider.com as a gateway in sofia status. How do I get
rid of this, of course,
You can remove it all you want but those settings in vars.xml won't do
anything because its expanded in conf/directory/default/example.com.xml
/b
On Jun 10, 2009, at 4:04 PM, Lars Zeb wrote:
Is it OK to remove the example.com gateway? I removed the
example.xml files in
FYI,
I've added a few new posts on the main FreeSWITCH page:
http://www.freeswitch.org/node/190 - OpenSimulator on EC2
http://www.freeswitch.org/node/191 - Rob Smart FS as-home-PBX (U.K.) how-to
Enjoy!
-MC
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palani vel wants you to join Yaari!
Is palani your friend?
a
href=http://yaari.com/?controller=useraction=mailregisterfriend=1sign=YaariNYX927AYT966XXO323IQT265;Yes,
palani is my friend!/a a
Rupa,
I think the console log has information in it that log/freeswitch.log does
not.
Console:
[r...@fs bin]# ./freeswitch
2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing
Engine.
2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch
thread 0
On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb larc...@yahoo.com wrote:
Rupa,
I think the console log has information in it that log/freeswitch.log does
not.
Console:
[r...@fs bin]# ../freeswitch
2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing
Engine.
To duplicate our old PBX park functionality I need for a user who's
on a call to be able to pick up a second line and dial a number to
park the other call which is on his phone. I have something working,
however am curious if there's a better way to accomplish this.
Specifically I'm curious if
My company is currently investigating a couple of projects that may take
me in the direction of FreeSwitch... In general, our management does
not often consider open source software for projects such as this, but
I've been successful in proving to them recently that open source can
deliver.
Oh, it is probable that logging is initialized after nat so the first
pass won't show up in the filesystem logs. It would should up on subsequent
nat initialization (what I was testing with some new code).
On Wed, Jun 10, 2009 at 6:54 PM, Lars Zeb larc...@yahoo.com wrote:
Rupa,
I
Yes, it's the right way to go!
Thanks, man!
On Wed, Jun 10, 2009 at 10:25 PM, dujinfang dujinf...@gmail.com wrote:
On Jun 10, 2009, at 3:33 PM, Jingwei Yang wrote:
Hi All,
I just finished installing freeSwitch and Skypiax. And I'm able to use
skype api directly via the sk command like
Another vote for Git here :).
On Mon, Jun 8, 2009 at 11:30 PM, Jason White ja...@jasonjgw.net wrote:
Lars Zeb larc...@yahoo.com wrote:
I had a working FS installation which I messed up by doing a fresh
install.
I tried to integrate all my custom changes, but I'm sure I screwed
something
Why not use the presence events to keep that state?
/b
On Jun 10, 2009, at 7:31 PM, John Wehle wrote:
To duplicate our old PBX park functionality I need for a user who's
on a call to be able to pick up a second line and dial a number to
park the other call which is on his phone. I have
Hi,
I am slowly gaining confidence using FreeSWITCH in production, but there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current calls?
So far I have been upgrading in the dead of night, after pausing for 5
Hi,
Create two sip profiles, one per IP.
Math
On 10-Jun-09, at 11:09 PM, lee jason wrote:
Dear All,
I just have a question, How can I use Freeswitch to blind two
IP address for SIP registration at same port(5060 UDP)?
Thanks a lot.
Jason Lee
Dear All,
I just have a question, How can I use Freeswitch to blind two IP
address for SIP registration at same port(5060 UDP)?
Thanks a lot.
Jason Lee
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Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
How are you handling your FS box crashing?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of John
Dalgliesh
Sent: Wednesday, June 10, 2009 9:04 PM
To: freeswitch-users@lists.freeswitch.org
Subject:
By reporting it on Jira so it doesn't crash anymore :D
On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote:
How are you handling your FS box crashing?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org
]
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