On Fri, Sep 18, 2009 at 10:41:19AM +0200, Sias Mey wrote:
Indeed the debug info did shed some light.
I know at least one other poster has asked about that before. It seems to be
an error related to accesing mysql InnoDB via odbc. Something about not
decreasing the thread count before
I am a bit confused with what's going on in a following scenario.
I have a public FS server with a public conference, that clients are
connecting to with my softphone. All of this softphones have STUN option
enabled and working, effectively resolving client's public IP address. They
also have ICE
RobertT siniy...@gmail.com wrote:
Where is the problem? Is it NAT, closing RTP port after some silence period
from client?
It could be a time-out, i.e., the nat router isn't keeping the port
translation alive.
I don't like nat at all. As more people migrate to IPv6 the problem will
gradually
Are there ways to escape this timeouts exchanging RTP with FS? Why didn't
waste flag help? Maybe I should flood channel in both directions? Will CNG
on a client side be a good descision? =)
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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
Sorry for the email subject that sounds like a IEEE paper.
I am building IVRs using FS API and sending out audio that is a
combination of TTS and playing WAV files. What is the best way to control
volume levels? I know i might be asking for magic here...
In any case, is there any simple ways to
I commented out the following in our internal profile:
param name=media-option value=bypass-media-after-att-xfer/
Which helped. We don't use bypass media though..
Jan
On Wed, Sep 23, 2009 at 8:59 AM, Michael Jerris m...@jerris.com wrote:
Did you ever resolve this issue? If not, please make
NO, proxy media and bypass media are wildly different behaviors and do
process things a little differently.
/b
On Sep 29, 2009, at 12:40 AM, Mariano de Llano wrote:
it seams very weird to me
that Sofia uses different approach to parse the mappings depending if
it is handling or not the
Type 'sofia loglevel all 9' then 'sofia profile siptrace on'
replace with the profile name then press F8 to turn debug log on
Capture the whole thing and email me the log. I can pretty much tell
you sofia is pissed about something in the SDP but I wanna see the logs.
Thanks,
NAT involved?
/b
On Sep 29, 2009, at 7:27 AM, Jan Kubr wrote:
I commented out the following in our internal profile:
param name=media-option value=bypass-media-after-att-xfer/
Which helped. We don't use bypass media though..
Jan
___
ты все еще наблюдаешь эту проблему?
я думал она уже решена...
эни вей, я уже приехал и сделаю скоро воторой IP нам для собственного
STUN-сервера.
--
Best regards,
Dmitry Kadantsev
http://www.doxwox.com - Best web meeting and online collaboration tool.
On Tue, Sep 29, 2009 at 10:32 AM, RobertT
Hi All,
please let me know implementation of failover in freeswitch.
Thanks
Srinivas
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в том то все и дело что с тобой мы эту проблему вроде как решили, а у Юры ее
никогда не было. и тут на тебе...
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Hi everybody,
I have a problem with Alphanumeric to numeric user mapping
I have done like it's written here :
Alphanumeric to numeric user mapping
Say you want a user's id to be alphanumeric (like an email username), such
as johnsm...@pbx.example.com. These users have alphanumeric usernames in
The best way is to start with normalized sound files, then to use
whatever features are available in your tts engine to send the right
volume matched to the sound files. That being said, a new media bug
was just added in trunk for auto gain control and that might help, but
I would never
Could you test this in svn trunk please.
Mike
On Sep 29, 2009, at 9:33 AM, Jonathan Barou wrote:
Hi everybody,
I have a problem with Alphanumeric to numeric user mapping
I have done like it's written here :
...
But when I want to call my alias-number, FS says No Route, Abording
My
Most likely the client NAT is cutting off the translation due to no
traffic. This could be because the client is not sending any traffic,
regardless of settings you set on FreeSWITCH. Try disabling all vad
and dtx on your soft phone to see if this helps. Also, your email
seems to
I'm sure of that. But I was talking about how is handle the parse of
the packet not the process, and also I was referring to the case when
FS is actually handling the media (proxy-media=false
bypass_media=false) as I said before FS ignores the las parameter in
the rtrpmap when is
On Mon, Sep 28, 2009 at 9:05 PM, Even André Fiskvik grev...@me.com
wrote:
From: Even André Fiskvik grev...@me.com
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 28 Sep 2009 22:52:13 +0200
Subject: Re: [Freeswitch-users] Load-Balance FreeSWITCH with Ultramonkey
I have been working with
I'm sorry but I'm new in the freeswitch communauty, what I have to do to
test this in svn trunk ?
Thanks
2009/9/29 Michael Jerris m...@jerris.com
Could you test this in svn trunk please.
Mike
On Sep 29, 2009, at 9:33 AM, Jonathan Barou wrote:
Hi everybody,
I have a problem with
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code
http://wiki.freeswitch.org/wiki/Installation_Guide#Compiling_the_Source_Code
On Sep 29, 2009, at 10:19 AM, Jonathan Barou wrote:
I'm sorry but I'm new in the freeswitch communauty, what I have to
do to test this in
On 29/09/09 16:37 +0530, srinivasula reddy wrote:
Hi All,
please let me know implementation of failover in freeswitch.
I'm also interested in this topic.
Obviously there are some things which are harder to failover than others,
like event socket connections, or agent queue states, but what
Has anyone given this any thought? Do I need to provide more information?
It's still not making any sense to me, and I'm planning on just
removing all of the default dialplans, but I'd like to make sure this
won't recur in the future.
BB
On Fri, Sep 25, 2009 at 9:33 AM, Bradley Brashier
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
is it possible to detect and avoid loops in dialplan caused by two or
more extensions which create a redirect chain?
regards
helmut
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Say, could someone please direct me to information on registering more than
one DID with a SIP provider?
When I try this using two XML files in ~/conf/sip_profiles/external, I find
only one or the other registers (both work fine when I use them separately).
I don't think the issue is with the
On Tue, Sep 29, Michael Gende wrote:
Say, could someone please direct me to information on registering more than
one DID with a SIP provider?
When I try this using two XML files in ~/conf/sip_profiles/external, I find
only one or the other registers (both work fine when I use them
Sorry for this mundane question, but how do I search mailing archives for
keywords? The following link has no search option?
http://lists.freeswitch.org/pipermail/freeswitch-users/
Thanks And Best Regards,
Jerry
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Here they are, mildly modified:
~/conf/sip_profiles/external/gateway1212.xml (name changed):
include
gateway name=33.44.55.66
param name=username value=8158381212/
param name=password value=bloodyblahbloodyblah/
param name=expire-seconds value=60/
!--/// do not
On Tue, Sep 29, Jerry Richards wrote:
Sorry for this mundane question, but how do I search mailing archives for
keywords? The following link has no search option?
You can use google. site:lists.freeswitch.org yourterm
--FC
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we decrement max forwards across a bridge and on transfer, so they are
supposed to sort themselves out automatically, this of course won't
resolve situations like loops via a provider or pstn that do not pass
along max forwards.
Mike
On Sep 29, 2009, at 11:49 AM, Helmut Kuper wrote:
google site:http://lists.freeswitch.org/pipermail/freeswitch-users/ my
search term here, or try nabble.
Mike
On Sep 29, 2009, at 12:35 PM, Jerry Richards wrote:
Sorry for this mundane question, but how do I search mailing
archives for
keywords? The following link has no search option?
Hello,
I have some problems running a FreeSWITCH with endpoints located in different
subnets. For example a FS is listening at 192.168.50.14/32 and endpoints from
the same (192.168.50.0/24) subnet work as expected. But when I try to receive a
media from an endpoint located at different subnet,
On Tue, Sep 29, Michael Gende wrote:
Here they are, mildly modified:
~/conf/sip_profiles/external/gateway1212.xml (name changed):
include
gateway name=33.44.55.66
param name=username value=8158381212/
param name=password value=bloodyblahbloodyblah/
param
Maybe your router isn't really a router and is doing NAT behind NAT?
Need logs and sip traces because we would only be guessing at this
point.
/b
On Sep 29, 2009, at 10:52 AM, Andrey Nepomnyaschih wrote:
Hello,
I have some problems running a FreeSWITCH with endpoints located in
Hi Mike,
You might want to try putting them in different profiles (maybe one on
port 5080, one on 5082?) so that the provider sees them as coming from
distinct places - that way they should let you use both at once, rather
than just seeing whichever was the last to register.
Cheers --
Dave
On Tue, Sep 29, 2009 at 7:19 AM, Jonathan Barou jba...@sqli.com wrote:
I'm sorry but I'm new in the freeswitch communauty, what I have to do to
test this in svn trunk ?
Thanks
FYI,
If you're running in Linux then this is a handy way to build from scratch
using a somewhat automated process:
Thanks for the advice, Dave. Most appreciated.
Regards,
Mike G.
On Tue, Sep 29, 2009 at 12:11 PM, David Knell d...@3c.co.uk wrote:
Hi Mike,
You might want to try putting them in different profiles (maybe one on
port 5080, one on 5082?) so that the provider sees them as coming from
I'll try it. Many thanks.
On Tue, Sep 29, 2009 at 12:03 PM, Frank Carmickle fr...@carmickle.comwrote:
On Tue, Sep 29, Michael Gende wrote:
Here they are, mildly modified:
~/conf/sip_profiles/external/gateway1212.xml (name changed):
include
gateway name=33.44.55.66
param
This topic has been beaten to death recently. Search the archives for things
like redundancy and failover and you'll see lots of discussions. The
bottom line is that your needs will dictate how much time, effort, and money
you are willing to sink into this. If you want professional assistance then
That did it, Frank. You're good deed for the day.
Mike G.
On Tue, Sep 29, 2009 at 12:20 PM, Michael Gende mge...@gendesign.comwrote:
I'll try it. Many thanks.
On Tue, Sep 29, 2009 at 12:03 PM, Frank Carmickle fr...@carmickle.comwrote:
On Tue, Sep 29, Michael Gende wrote:
Here they are,
Hello Frank,
There is only one interface on FS box, it has an IP address 192.168.50.14/24
and the 192.168.60/24 is accessible through 192.168.50.3. Correct me if I'm
wrong, but if I set the mask to be /16, then 192.168.60/24 will be unreachable
from FS as it will be treated at local connected
http://dir.gmane.org/gmane.comp.telephony.freeswitch.user also works.
Many ways to find what you are after more of matter of preference and
which you remember when you need to find something.
-- W
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On Tue, Sep 29, Andrey Nepomnyaschih wrote:
Hello Frank,
There is only one interface on FS box, it has an IP address 192.168.50.14/24
and the 192.168.60/24 is accessible through 192.168.50.3. Correct me if I'm
wrong, but if I set the mask to be /16, then 192.168.60/24 will be
unreachable
Hello everyone,
I am trying to add a gateway, but after configuring it just like the others
gateways I have, it is failing to register with a message like this:
2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed
with status Operation has no matching challenge [904].
Yeah, I've been working with Network Admin on this issue, and he says that the
router simply routes the packets without doing any translation stuff. Although
he couldn't answer where exactly the RTP stream is going to, so we agreed to
switch from one software to another tomorrow and see if
hi folks, I want to change the default domain name which is by default
$${local_ip_v4} .
In the vars.xml , i have changed the value to the myexample.com but when I
restart the freeswitch server, it still shows domain to be my local ip
address.
X-PRE-PROCESS cmd=set data=domain=myexample.com/
900 level errors are sofia internal errors so probably something is wrong
with your gateway config xml.
if you want to send it with any critical info replaced with XXX maybe we can
see the issue for you.
On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote:
Hello
Anybody tried siptapi with freeswitch?
http://sourceforge.net/projects/siptapi/
This may enable Click2Dial e.g. from Outlook to Freeswitch.
So anybody has experience with that solution?
Best regards
Peter
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On Tue, Sep 29, pankaj anand wrote:
hi folks, I want to change the default domain name which is by default
$${local_ip_v4} .
In the vars.xml , i have changed the value to the myexample.com but when I
restart the freeswitch server, it still shows domain to be my local ip
address.
I had this
Yes, the phones are behind NAT. Freeswitch is on a public IP.
j
On Tue, Sep 29, 2009 at 3:10 PM, Brian West br...@freeswitch.org wrote:
NAT involved?
/b
On Sep 29, 2009, at 7:27 AM, Jan Kubr wrote:
I commented out the following in our internal profile:
param name=media-option
Nope but I would be happy to see someone try and document it on our
wiki.
/b
On Sep 29, 2009, at 1:44 PM, Peter P GMX wrote:
Anybody tried siptapi with freeswitch?
http://sourceforge.net/projects/siptapi/
This may enable Click2Dial e.g. from Outlook to Freeswitch.
So anybody has
I've downloaded 0Kb in 56 minutes. Anyone mind setting up a seeder? I'm on
a pretty fast connection.
While waiting, I found the following discussions:
I found the following threads while searching for redundancy and failover:
Is there a way in FS to selectively deny a BLF presence subscription
request for the sake of privacy? So that groups could be defined that are
allowed to be monitor or be monitored? And others that are not allowed to
monitor or be monitored?
Best Regards,
Jerry
Not at this time... there is nothing in place to do so. Maybe you can
post a bounty on Jira for this feature.
Thanks,
Brian
On Sep 29, 2009, at 2:52 PM, Jerry Richards wrote:
Is there a way in FS to selectively deny a BLF presence subscription
request for the sake of privacy? So that
Those discussions are probably it. It goes something like:
I want hot failover. Can't FS just load state?
No, we think it'd take about $100K of work, and considerable time
Oh. XXX does it. I'd think we just need a database.
No, it'll take a lot of effort. Just use a SIP proxy and redirect traffic
Hello All,
I have an internal extension that needs to send an INVITE without SDP body
(Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable
with reason MANDATORY_IE_MISSING. Would anyone know what I need to do to
enable this?
Best Regards,
Jerry
This is how I went and made the multi-tenant, this config will simply create
two users: 1...@company-a.org and 1...@company-b.org.
It also has two dialplan/context for each domain, so if the
1...@company-a.org user tries to dial a extension, it will go and look in
the company-a.org
Why not start a wiki page on the topic?
/b
On Sep 29, 2009, at 4:33 PM, Diego Viola wrote:
Hope that helps someone.
Diego
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Someone already wikifyed it for me =D
http://wiki.freeswitch.org/wiki/Multiple_Companies
Thanks Brian :)
On Tue, Sep 29, 2009 at 9:49 PM, Brian West br...@freeswitch.org wrote:
Why not start a wiki page on the topic?
/b
On Sep 29, 2009, at 4:33 PM, Diego Viola wrote:
Hope that helps
Anthony, thanks. Below are my config files for the two gateways from the sip
trace. Both files are located in conf/directory/default.
-
redvoiss.xml (the one that works)
include
user id=gateway_redvoiss
gateways
gateway name=redvoiss-pp
param
I've noticed that when using an alphanumeric ID, case sensitivity causes
issues. It doesn't seem to matter what case the userID is in the XML
directory. What does seem to matter is how the user registered their
phone. If the user registered in all upper case (JOHN_SMITH), then
dialing
Hi all,
I have a FS that registers on an Ericsson pabx as gateway under sip_external.
This gateway start registering on the Ericsson ok, but after a while,
around 50mins, it fails with the logs below.
If I hit *sofia profile external restart* on fs_cli then the gateway
returns to register with
I need the sip trace.
/b
On Sep 29, 2009, at 7:28 PM, Fernando Testa wrote:
Hi all,
I have a FS that registers on an Ericsson pabx as gateway under
sip_external.
This gateway start registering on the Ericsson ok, but after a while,
around 50mins, it fails with the logs below.
If I hit
Hi,
I want to use mod_nibble in freeswitch for billing purpose. I've got some
questions, as I'm listing down below;
1- How I create MySQL connection with FS?
2- Is it possible to query the database when billing occurs?
3- Will database credentials require each time when query to database?
4-
mod_nibble uses ODBC. Via ODBC u can connect to mysql.
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
mod_nibble has it's own table design.
http://wiki.freeswitch.org/wiki/Mod_nibblebill
It may be possible to tweak it to use your current tables but I'd
suggest getting the
basics working
Dear all,
I am in the process of implementing IVR server in Perl using
event outbound socket. I want to put a call in music on hold when the perl
is doing something.
I have tried the following steps.
* First using Set application set the hold_music variable
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