I didn't say i have a working FS on blackfin... i just said i've ported a
lot of software to blackfin and it was always floating point, fork vs
vfork ... main issues... but why do you think it cannot be done?
T.
On Mon, Sep 21, 2009 at 6:08 AM, Hadley Rich h...@nice.net.nz wrote:
On Mon,
In that case you should turn on sip trace for profile where your callcentric
peer is configured. By default FS comes with two profiles namely internal
and external. If you haven't created any new profile and configured your
users and peers in these two profiles then you should try turning on sip
Or as a more affordable solution... is it possible to connect an entry-level
GSM phone to a PC running Freeswitch and use this as a poor man's gateway?
--
View this message in context:
http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530241.html
Sent from the
With help from Pakistan Software Export Board (PSEB), we formed Asterisk
Pakistan community forum in early 2008. This forum is still active and we
arranged many workshops during last 18 months in all major cities of
Pakistan. It was a great success and we effectively introduced Asterisk in
so many
How far do you want things to scale?
/b
On Sep 19, 2009, at 4:34 AM, Nagalenoj wrote:
Dear friends,
I want to know which is the better way to do route calls and
control
calls. I've did a experiment which can be done in both ways,
Mod_perl and
ESL. I don't know which one is better
If you refer to the latest internal.xml in the default config for sip
profiles you'll see an example of how to use a single profile for
phones inside and outside of NAT. So you no longer have to have two
profiles thus cutting the confusion level to almost zero when you
setup FreeSWITCH to
You can't put the users directly into a db with FreeSWITCH you'll have
to serve up the XML document via XML CURL or write your own module to
do so via the module interfaces provided.
/b
On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote:
Yes use odbc in fs
Thanks Regards,
Mitul Limbani,
On Mon, Sep 21, Brian West wrote:
No you no longer have to do this. Please refer to the internal.xml
profile in the default config. If you set the local-network-acl and
then set ext-sip-ip and ext-rtp-ip then the profile will figure out
which IP to use based on the destination or
--
С уважением, Кривушин Михаил
г. Томск сот. +7 913 865 78 66
icq: 218 744 127
xmpp: krivushi...@jabber.ru
skype: mkrivushin
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Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong
registrations (bad username or password) in less then 50 seconds (49496 ms
to be exact) and it processed all of them and gave correct responses using
XML CURL.
I am willing to do this test again soon, with correct registration data
I searched my sent emails and found the results, copying it below (after
removing some sensitive info),
1,000 Calls
==
Total 1000 REGISTER calls sent in 890 ms at rate of 1123/sec
Total 1000 responses receieved in 4516 ms at rate of 221/sec:
its a waste of time ... i doubt it can be done.
T.
On Mon, Sep 21, 2009 at 10:56 AM, Fred-145 codecompl...@free.fr wrote:
Or as a more affordable solution... is it possible to connect an
entry-level
GSM phone to a PC running Freeswitch and use this as a poor man's gateway?
--
View this
Check out this range
http://www.noblesolutions.co.uk/shop/index.php?main_page=indexmanufactu
rers_id=16
You should be able to find a local supplier
We've used them for a couple of years now. They just plug into your
network.
Regards,
-Original Message-
From:
Whoah what a term - ONCE BOMBED FS...
Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong
registrations (bad username or password) in less then 50 seconds (49496 ms
to be exact) and it processed all of them and gave correct responses using
XML CURL.
I am willing to do this
well first off you would setup a gateway and set the param 'from-
domain' to what you wish it to be.
/b
On Sep 21, 2009, at 5:44 AM, Mikhail Krivushin wrote:
I see that call_id_number placed in From:, but with wrong realm. I
need a way for change realm in From:. Is any ability to do that?
yes but you can lie about IP's in the via/to and from if you set the
local-network-acl ... I'm not talking two physical interfaces on
FreeSWITCH... because that is one of the harder scenarios to setup...
I'm talking single interface on FS sitting behind a nat router which
is the most
Fred-145 wrote:
Hello
I'm selling a basic solution for SOHO customers (FS is installed on their
work computer running Windows or Macs) to handle an analog phone line.
When they're on the road, in addition or instead of getting a notification
by e-mail when someone calls their office,
On Sun, Sep 20, 2009 at 08:42:40PM +0200, Remko Kloosterman wrote:
This actually sounds very good Andrew. You even have an agent interface.
Do you have plans for a outbound campaign dialer? I know of a commercial
dialer that is good in it's predictive algotithm, but very bad when it
comes to
I' ve get the same error with a fresh tree
Thanks in advance
De: Brian West br...@freeswitch.org
Para: freeswitch-users@lists.freeswitch.org
Enviado: jueves 17 de septiembre de 2009, 10:12:36
Asunto: Re: [Freeswitch-users] Compile error
NO you must not. The
Hi All,
I am new to Freeswitch. So please bear with me if I ask any silly questions.
* Can anyone of you please tell me how to display the extension name which
has matched an incoming/outgoing call.
* And can you please elaborate what does 'action application=info/'
mean.
* Suppose we have set
Hello Anil
On Mon, Sep 21, Anil Kumar S. R. wrote:
* Can anyone of you please tell me how to display the extension name which
has matched an incoming/outgoing call.
In the log you will find something like this
2009-09-21 14:36:15.574827 [INFO] mod_dialplan_xml.c:315 Processing
fs-03977304 in
Hi,
I have trouble recording incoming calls with FreeSwitch.
I have followed the instruction from Misc. Dialplan Tools record session
(http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session)
It works well for outgoing calls, but I have the problem with incoming
calls.
The person
On Mon, Sep 21, 2009 at 11:27 AM, Anil Kumar S. R. sra...@gmail.com wrote:
Hi All,
I am new to Freeswitch. So please bear with me if I ask any silly
questions.
* Can anyone of you please tell me how to display the extension name which
has matched an incoming/outgoing call.
* And can you
set ringback before record_session and also set transfer_ringback
because record_session causes an pre-answer.
/b
On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote:
Hi,
I have trouble recording incoming calls with FreeSwitch.
I have followed the instruction from Misc. Dialplan Tools record
At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the
FreeSWITCH team have been working together to create a new PBX appliance.
Dubbed the CudaTel Communications Server, this new communications platform
is both feature-rich and easy-to-use. We are pleased to announce that
Have not found anything usable in the wiki/mail list archives.
I'm trying to setup BLF (busy lamp field) for Grandstream GXP-2000
phone. It offers BLF/eventlist BLF modes. Does Freeswitch supports
both including the latter (RFC4662)? How to setup BLF on Freeswitch
side? Are there any examples?
Hi Guys,
I have an issue running FS... it crashes apparently without leaving any log
... not even a core dump is left.
The machine is dual AMD opteron quad core with 8 GB RAM and i'm running 75
simultaneous calls (with media) with a rate of 5 calls per second.
As i was not able to reproduce
We invite you to visit the CudaTel http://www.cudatel.com/ website or
call 989-720-4000 for more information or to request evaluation
units.
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URL???
On 21/09/2009, Michael Collins m...@freeswitch.org wrote:
At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the
FreeSWITCH team have been working together to create a new PBX appliance.
Dubbed the CudaTel Communications Server, this new communications platform
is
On Tue, 22 Sep 2009 11:13:13 Gavin Henry wrote:
URL???
On 21/09/2009, Michael Collins m...@freeswitch.org wrote:
We invite you to visit the CudaTel http://www.cudatel.com/ website or
call 989-720-4000 for more information or to request evaluation units.
--
https://nicegear.co.nz
VoIP and
On Mon, Sep 21, 2009 at 4:19 PM, William Suffill
william.suff...@gmail.comwrote:
We invite you to visit the CudaTel http://www.cudatel.com/ website or
call 989-720-4000 for more information or to request evaluation
units.
Hehe, thanks for pointing that out. Also, I said TMD hardware option
Hi,
Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT
hangup cause on the second leg. I looked for info on the Wiki and Google,
but I couldn't find a detailed explanation. Does anybody know what does it
mean exactly?
Thanks!
Nicolas
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