Hi Anthony,
well currently both ip addresses and ports (of proxy and registrar) are
the same. And it works good as it is now. :)
regards
Helmut
On 10.02.2009 15:13, Anthony Minessale wrote:
register-proxy is for where it actually sends the packet but it will
still say the name of proxy in
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 authentication
challenge (SIP/2.0 407) received from proxy and simply terminates the
session.
Is
Hi Brian,
of course. Will do it as soon as I have a second FS plattform for
testing SVN trunks.
thx and regards
Helmut
On 11.02.2009 13:16, Brian West wrote:
The proper location to post this is on Jira... please in the future
report ALL issues via Jira. They'll get lost if not done this
Have you seen this? and how will impact FS in the future?
From Seth:
(Urgent. Send your note TODAY and CONFIRM the automatic reply from
IETF. You can cc campai...@fsf.org . Three links below, FSF's
Why do you need to wait?
jira is just a website, just go there and file the bug and attach the file
that causes the issue.
On Wed, Feb 11, 2009 at 6:43 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
Hi Brian,
of course. Will do it as soon as I have a second FS plattform for
testing SVN
Hi Anthony,
On 11.02.2009 14:58, Anthony Minessale wrote:
Why do you need to wait?
jira is just a website, just go there and file the bug and attach the
file that causes the issue.
Well, there is a question on jira, which makes sure that I have
reproduced the bug on SVN trunk ... but I'm not
if the alternative is to post it to the mailing list, you have our
permission this one time to answer not yet so you have somewhere to attach
the bad file so we can reproduce it.
On Wed, Feb 11, 2009 at 8:12 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
Hi Anthony,
On 11.02.2009 14:58,
Hi Anthony,
I quickly have setup a test server with current trunk. So I can now
enter there a YES into that field. Current trunk crashed as well. But
thx for stretching the jira rules a bit :)
I attached the file on jira in
http://jira.freeswitch.org/browse/MODFORM-24
Can you delete it asap
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 and it doesn't even say anything about the file
being bad etc.. it plays and hangs up fine even 4 times at once.
It would be a big help if you could
Hi Mike,
I removed the apr dependency for timestamp and use the function openzap
delivers. Works good so far.
For unstabilities in openzap and Q931: On my side main problem seems to
be, that channels for inbound traffic sometimes not be freed during
runtime. Maybe our remote TDM end (AVAYA)
Hi Anthony,
hm ... have to dig for a centos5 machine here ... I have one somewhere
... I will test it there as well.
Concerning the prerelease of gcc ... My svn trunk FS was compiled by
gcc version 4.1.2 (Ubuntu 4.1.2-0ubuntu4). In Jira I entered the gcc
version of FS in production.
regards
Hi List,
Is there any open source call center tool available which works with FS?
Kind Regards
Saeed Ahmed Tariq
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that is interesting. we are receiving the dtmf digits over 2833. might
it be possible, that we receive 2833 AND inband (we asked our carrier
for 2833, because we had problems with inband and fs - and we got it)?
is there something we can setup in fs or is it a problem wich only our
carrier can
Well if they are sending both they are broken. I would call and yell
at them and beat them with a cluebat.
/b
On Feb 11, 2009, at 10:42 AM, Dennis wrote:
that is interesting. we are receiving the dtmf digits over 2833. might
it be possible, that we receive 2833 AND inband (we asked our
i can't tell, if they are sending both, but it seems so. we get 2833
for sure. they were kind enough to give it to us, because inband seems
to be quite unreliable over sip.
how can in find out, if both are coming and is there a way to block
inband to test?
perhaps we need both: if we bridge an
turn on the start_dtmf app and dial digits from the outside.. if you
get duplicate digits then they are sending both.
/b
On Feb 11, 2009, at 11:14 AM, Dennis wrote:
i can't tell, if they are sending both, but it seems so. we get 2833
for sure. they were kind enough to give it to us, because
ok, i will try this, but how can it be possible, that inband tones are
audible in conference, when we do not even have start_dtmf activated?
i just don't understand, why it must be dtmf inband, if the tones are
audible and how they can be audible, if start_dtmf is not set.
is it, because the
anthony, did you make the changes with add {instant_ringback=true} to
make ringback not wait for indication to generate ringback for the
described problem?
we read something like this out of it, but we can not test it, because
we get errors with the latest fs version (switch_ivr.c:674
On Feb 11, 2009, at 12:23 PM, Dennis wrote:
ok, i will try this, but how can it be possible, that inband tones are
audible in conference, when we do not even have start_dtmf activated?
They aren't really sending 2833.
i just don't understand, why it must be dtmf inband, if the tones are
I have a situation where FS aborts
I'm running an lua script with mysql statements
First time it runs, on hangup I get
[CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim()
Returning 4 recycled memory pool(s)
If I run it again, FS exits.
Should there be an error log
Can you show us what you're doing?
/b
On Feb 11, 2009, at 1:15 PM, Nik Middleton wrote:
I have a situation where FS aborts
I’m running an lua script with mysql statements
First time it runs, on hangup I get
[CONSOLE] switch_core_memory.c:374 switch_core_memory_reclaim()
Returning 4
I was running in a screen session, so going back to the console it shows
it's a seg fault
2009-02-11 19:27:53 [NOTICE] sofia.c:3090 sofia_handle_sip_i_state()
Hangup sofia/internal/1...@192.168.3.206 [CS_EXECUTE] [NORMAL_CLEARING]
Segmentation fault (core dumped)
Seg fault occurs on
... did anybody succeed with this ? The solution for VS2008 does not seem to
have a valid 64bit configuration.
Regards
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I don't have a 64 bit windows box/os to get this working. Someone
with access to such a box would have to set this up and submit a patch.
Mike
On Feb 11, 2009, at 2:49 PM, Public Dump wrote:
… did anybody succeed with this ? The solution for VS2008 does not
seem to have a valid 64bit
this does not help. we are using socket outbound and everything worked
before the changes yesterday.
we have the same error with other dialplans.
2009/2/11 Brian West br...@freeswitch.org:
Try answer or pre_answer before park.
/b
On Feb 11, 2009, at 12:37 PM, Dennis wrote:
anthony, did
Please collect the backtrace and report it on Jira.
/b
On Feb 11, 2009, at 2:11 PM, Dennis wrote:
this does not help. we are using socket outbound and everything worked
before the changes yesterday.
we have the same error with other dialplans.
Where is the core dump written?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 11 February 2009 19:38
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
Forget my last, followed the link
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 11 February 2009 19:38
To: freeswitch-users@lists.freeswitch.org
Subject:
Try starting it from your /usr/local/freeswitch/bin... ./freeswitch
it'll dump in the same folder.
/b
On Feb 11, 2009, at 2:20 PM, Nik Middleton wrote:
Where is the core dump written?
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Hi all, I¹m just starting playing around with FS and I¹ve searched for the
answer to what I think is an easy question but I can¹t find it. I have FS
running, 2 X-lite clients on 2 different computers connected using the
preconfigured 1000 and 1001 extenstions. Both can call each other and
show me sofia status, Try changing the @ to a % but I really need
to see the sofia status output.
/b
On Feb 11, 2009, at 2:32 PM, Chris Elam wrote:
$cmd = api originate sofia/mydomain.com/1...@192.168.15.50 bridge(sofia/mydomain.com/1...@192.168.15.50
);
The result I get is : -ERR
The % gives the same error. Here is the sofia status output:
API CALL [sofia(status)] output:
Name Type Data
State
=
That's it, worked perfectly, thanks a bunch!
On 2/11/09 3:59 PM, Brian West br...@freeswitch.org wrote:
try sofia/myinsideip/1000 and sofia/myinsideip/1001
I sure hope it doesn't say myinsideip on there and you only tried to
hide your IP.
/b
On Feb 11, 2009, at 2:54 PM, Chris Elam
After reading you suggestions I deployed the version from SVN today, the
problem persists.
Regards
Von: Public Dump
Gesendet: Dienstag, 10. Februar 2009 19:42
An: 'freeswitch-users@lists.freeswitch.org'
Betreff: High CPU load after starting
After starting FreeSwitch (1.0.2) on a 4 core
Are you sure you rebuilt it clean? Are you doing anything special?
Changing any configs?
/b
On Feb 11, 2009, at 3:11 PM, Public Dump wrote:
After reading you suggestions I deployed the version from SVN today,
the problem persists.
Regards
For compiling x64 code you shouldn't need a 64bit system, but you couldn't run
it of course.
I don't have a 64 bit windows box/os to get this working. Someone with access
to such a box would have to set this up and submit a patch.
Mike
___
and make sure it's svn trunk or at least a daily snapshot and not 1.0.2
On Wed, Feb 11, 2009 at 2:23 PM, Brian West br...@freeswitch.org wrote:
Try starting it from your /usr/local/freeswitch/bin... ./freeswitch it'll
dump in the same folder.
/b
On Feb 11, 2009, at 2:20 PM, Nik Middleton
Figured out the phone was sending packets that were too large, and the
receiving system was not reassembling the fragmented packet. This can be
fixed on the Aastra by enabling basic codecs:
Go to the phone web-UI -- global SIP -- Codec Preference List -- Codec 1 --
change all to basic, save
Message: 3
Date: Wed, 11 Feb 2009 15:16:16 -0600
From: Brian West br...@freeswitch.org
Subject: Re: [Freeswitch-users] High CPU load after starting
To: freeswitch-users@lists.freeswitch.org
Message-ID: 1a18cbf4-81f9-4e57-af9d-54b6103b2...@freeswitch.org
Content-Type: text/plain;
If any one needs a FreeSWITCH box with a public, static IP, I can
provide them for you at a reasonable cost. I'm building a
Virtualization platform for FreeSWITCH hosting, and have the first
node complete. These are OpenVZ Virtual Engines with Centos 5.2, a
full build environment, and the latest
Quick note make sure you're 100% 64 bit.. if you need help with that I
can show you how on CentOS 5.2
/b
On Feb 11, 2009, at 4:34 PM, Nik Martin wrote:
If any one needs a FreeSWITCH box with a public, static IP, I can
provide them for you at a reasonable cost. I'm building a
Be sure to make the Virt nodes 64bit too... FS works 100% better w/ 64bit!
From: Nik Martin freeswi...@servercorps.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Wed, 11 Feb 2009 16:47:23 -0600
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH
I've had a VE in (light) production for about 2 weeks, with no issues
so far. I'm going to build a pure 64 bit VE container though, and
will run in that for a while too. Brian, you sid you have a readme on
that?
On Wed, Feb 11, 2009 at 4:55 PM, Nicolas Brenner nico...@medularis.com wrote:
http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora
and
http://linux.carreira.com.pt/ovzutils/setx86_64-0.3.tar.gz
Will set it up for 64bit containers and patch everything to work
correctly...
/b
On Feb 11, 2009, at 5:03 PM, Nik Martin wrote:
I've had a VE in (light)
Hi Guys
I'm trying to set the outbound caller-id in js. The params seem to be
acceptable, except I'm getting the default +0 caller-ID sent.
Should the below work with js?
session.originate(session,'{accountcode=54321,ignore_early_media=true,or
Lua has known issues with MySQL you must use latest SVN builds of the
luasql driver for that to avoid it.. and still its not stellar.. the
unixODBC one on the other hand works fine.
/b
On Feb 11, 2009, at 5:36 PM, Nik Middleton wrote:
I’ve abandoned LUA.
All sorts of problems (DTMF etc).
Great, thanks!
Nik
On Wed, Feb 11, 2009 at 5:08 PM, Brian West br...@freeswitch.org wrote:
http://wiki.openvz.org/Install_OpenVZ_on_a_x86_64_system_Centos-Fedora
and
http://linux.carreira.com.pt/ovzutils/setx86_64-0.3.tar.gz
Will set it up for 64bit containers and patch everything to
I'm having an issue with call accounting
If I initiate a call, and it is then transferred to an IVR menu.
Person selects 1 to talk to someone.
In js
else if (data.digit == 5) {
if (session.ready()) {
var new_session = new Session();
You can run a small SOHO operation on 256 megs
/b
On Feb 11, 2009, at 6:08 PM, EdPimentl wrote:
256 MB Ram . is this correct?... Does any VoIP provider to use
this?
-E
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first off don't use the session.originate
var new_session = new Session({var=val}sofia/blah/blah);
will do it all for you in one step. Also can you point me to where on
the wiki that keeps talking about session.originate? I need to clean
them off there.
/b
On Feb 11, 2009, at 6:09
Please refer to the extension name=unloop extension in public.xml
and default.xml both will cause a deflect to be done so the 3 leg
call gets turned back into a 2 leg call. In some cases it might be
desired to do a 3 leg call so you can bill the party that caused the
302 and the original
Thank you Brian,
is there any way to inspect what exactly is sent in 302 message and if
possible to replace it or remove it.
Regards
Chav
- Original Message -
From: Brian West br...@freeswitch.org
Date: Wednesday, February 11, 2009 4:17 pm
Subject: Re: [Freeswitch-users] How i can
Soho,,, yes of course...
Voip (soho)Service Provider not convinced is possible to provide
reliable QoS.
My .02 cents
-E
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session.originate(session,'{accountcode=54321,ignore_early_media=true,origination_caller_id_number=0763060,originate_timeout=25}sofia/gateway/mygw/01XXX');
(this works using lua BTW)
h... how about using effective_caller_id_number instead? I think
the JavaScript paradigm is a
Nope its on auto pilot... we don't get passed the 302 from sofia. So
what you have there is all you can get at.
/b
On Feb 11, 2009, at 6:21 PM, Tchavdar Paskov wrote:
Thank you Brian,
is there any way to inspect what exactly is sent in 302 message and
if possible to replace it or
Actually you can if you don't overload the machine like most VPS
providers do... The advantage with OpenVZ in this case is that you can
migrate the running FreeSWITCH instance between hardware nodes and not
drop calls at this size.
/b
On Feb 11, 2009, at 6:24 PM, EdPimentl wrote:
Soho,,,
This Second call leg is not accounted for in either CSV or xml logs
Am I doing something wrong? In the XML record is shows that I've diverted
to the new number, but the time is all bundled with the initial call.
This is exactly the same issue in Asterisk, which I was hoping to avoid.
Good question,
Does anybody have any contacts at Skype to open a discussion with them?
Should we just call them anyway?
They have chosen to interop directly with asterisk which has not completed
it's attempt at wideband support.
Maybe they are more interested in connecting to the PSTN but it's
On Wed, Feb 11, 2009 at 4:31 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Good question,
Does anybody have any contacts at Skype to open a discussion with them?
Should we just call them anyway?
They have chosen to interop directly with asterisk which has not completed
it's
However, in the original originate, any ideas why {var=val} is not being
processed?
I think Brian's suggestion is the way to go:
first off don't use the session.originate
var new_session = new Session({var=val}sofia/blah/blah);
The above syntax is the clean way to do it.
will do it
It is kind of being processed, the account code is being set, XML cdr's are
created and are correct, but csv cdr's for the account code are not
Caller ID is not being set in the A leg but is in the B Leg
DING DING DING!!! We have a weener!
Okay, that was the key piece of info. Most likely
If you have outbound gateways registering make sure you set the
context and extension param on the gateway so it'll go to the right
spot. Recent changes made it work much smoother.
/b
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My goal is obviously not to provide carrier grade VOIP switch service,
but a platform for sandbox testing of configurations, SOHO VOIP
Switches, development of FS addons, backup switch capability, etc.
Doing this stuff at home behind NAT and a consumer grade router is one
reason Brian, Anthony,
http://files.freeswitch.org/freeswitch-snapshot.tar.gz
Extracted into empty directory, SVN update, compile.
it was a completely clean build?
as in compleletely new and/or clean solution?
Which tarball was it ?
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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.
/b
On Feb 11, 2009, at 7:59 PM, Public Dump wrote:
Brian:
I am also running my freeswitch on my own openVZ containers. Just how do you
verify if the freeswitch is compiled as 64bit? I would assume if I compile
it under a 64bit container, I would automatically get a 64bit freeswitch
right?
On Wed, Feb 11, 2009 at 2:34 PM, Nik Martin
ding ding ding .. yep!
file /usr/local/freeswitch/bin/freeswitch will also confirm
/b
On Feb 11, 2009, at 6:37 PM, Henry Huang wrote:
Brian:
I am also running my freeswitch on my own openVZ containers. Just
how do you verify if the freeswitch is compiled as 64bit? I would
assume if I
There is always C, it's actually considered a high level language by many ;)
On Wed, Feb 11, 2009 at 5:50 PM, Brian West br...@freeswitch.org wrote:
Lua has known issues with MySQL you must use latest SVN builds of the
luasql driver for that to avoid it.. and still its not stellar.. the
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 well. Thanks for help.
BTW, without that FS would do a core dump (seg fault) on shutdown after
TTS was invoked at least once.
Looking at FS logs I
This is normal. Are you using 5.0? can you include examples of how
you're doing this?
/b
On Feb 11, 2009, at 10:38 PM, pauld wrote:
TRANSCODING_NECESSARY
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the problem is fixed in the latest version of fs - at least it is
working as before without any errors.
but there is still the question, if the changes where made because of
our problem with the not starting socket!?
we can see in the cli, that the var is set, but it does not change
anything
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