Kristian,
from your experience, supposed we go for net5501 + a 4 - 8 FXS card, what is
the maximum simultaneous calls that this box can handle of course using g729
codec?
I used blackgin (IP08), alix2d3... and all of them were giving up on 6-7
simultaneous calls.
To be honest, i didnt run
An intermediate report:
*Audiocodes*: TLS works only on outgoing requests, incoming ones are
ignored. I am waiting for Audiocodes' help in order to debug it.
SRTP: worked when no TLS is active. When TLS is active the call is
disconnected when the remote party answers. Still debugging it.
Sometime next week I hopefully am going to start a document that
follows my progress in setting up a FS system from scratch, with all
the pitfalls and successes. A kinds of warts and all story.
Alongside this blog (for want of a better word) I will also then
document the steps needed to get it
No publisher, although uploading and selling books (deadtree or online) is
easy with companies like www.lulu.com
I was just thinking of some way to learn FS gradually and effectively. The
frequent problem with wiki's, is that the quality of articles is uneven and
they don't have a good layout.
Frank Carmickle wrote:
A board with an atom 330 on it would probably do the trick for you. There
are a few made by Intel and Supermicro that look pretty nice. There were
some other people on the list looking to use them. Maybe we can get a
report from someone.
Intel came up with the
I use a dectop by Data Evolution... Its cheap at ~$100. I have it
running debian lenny and FS... works well for me.
http://www.dataevolution.com/dectop%20info%202.htm
http://www.gadgettastic.com/2007/08/18/dectop-the-100-pc/
On Thu, Dec 10, 2009 at 10:45 PM, Fred-145 codecompl...@free.fr
Mark Campbell-Smith wrote:
I use a dectop by Data Evolution... Its cheap at ~$100. I have it
running debian lenny and FS... works well for me.
Thanks for the tip, although this type of box doesn't have a PCI slot, so
the only way to connect FS to the PSTN is through a VoIP provider (or a
Hello
I'm going through the various XML files, and noticed this first line in
vars.xml.
X-PRE-PROCESS cmd=set data=default_password=1234/
What is this password used for?
Thank you
--
View this message in context:
I have confirmed it works with Polycom, Snom and a few others
polycom is the hardest to set due to having to put the ca cert into
the phone... but other than that its good.
/b
On Dec 10, 2009, at 3:11 AM, Yehavi Bourvine wrote:
An intermediate report:
Audiocodes: TLS works only on
please look in conf/directory/default/*.xml
/b
On Dec 10, 2009, at 7:40 AM, Fred-145 wrote:
Hello
I'm going through the various XML files, and noticed this first line
in
vars.xml.
X-PRE-PROCESS cmd=set data=default_password=1234/
What is this password used for?
Thank you
Hello
I wanted to check if my ADSL modem worked with STUN, so I left its UPNP
activity option unchecked, ran FreeSwitch, and used eg. Shields Up
(www.grc.com) to check if UDP5080 (and possibly UDP5060) were opened...
which SU says no.
Does it mean that...
- by default, FS doesn't use STUN
- or
Ah, makes sense:
conf/directory/default/1000.xml:
user id=1000
params
Thanks for the tip.
--
View this message in context:
http://old.nabble.com/-vars.xml--default_password%3D1234--tp26727371p26727835.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
Hello,
I try to invite a user into a conference by
loopback/255 8000 Conference
255 is the user, I invite the user via loopback as that way I can also
invite external numbers.
It processes the user's local dialplan correctly (as if the user was
normally dialled), however it only offers L16
On Thu, Dec 10, Fred-145 wrote:
Frank Carmickle wrote:
A board with an atom 330 on it would probably do the trick for you. There
are a few made by Intel and Supermicro that look pretty nice. There were
some other people on the list looking to use them. Maybe we can get a
report
STUN is not a way to open ports in a manner in which sheilds up would
detect.
http://en.wikipedia.org/wiki/STUN
http://en.wikipedia.org/wiki/STUNUPNP is what you want if you want to open
ports.
STUN is just a method for figuring out how to do nat traversal. STUN
method is initiated by the
On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle fr...@carmickle.com wrote:
The 330 boards are a little more power hungry but you get a dual core 64 bit
processor. As far as I'm concerned the performance increase is well worth
the extra money. You still well below the power consumption of
Don't worry.
I was an asterisk developer/volunteer in 2003. I still managed to figure it
out. ;)
On Thu, Dec 10, 2009 at 4:13 AM, Julian Lyndon-Smith aster...@dotr.comwrote:
Sometime next week I hopefully am going to start a document that
follows my progress in setting up a FS system from
Otis wrote:
div class=moz-text-flowed style=font-family: -moz-fixedI have 2
FS servers FS1 (aka medion) and FS3 (callweaver). These are set as
gateways and register with each other. I wanted all users on FS1 to
dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed
as 331001 on
Hi,
I am currently creating IVR using the functions provided in the XML dialplan
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
Using functions like this
entry action=menu-play-sound digits=1
param=$${base_dir}/1255549537_Welcome.wav/
I can play files, etc.
I wonder what is the
Thanks for the clarification. So it's either UPnP or STUN/port-mapping.
--
View this message in context:
http://old.nabble.com/Does-FS-support-STUN-by-default--tp26727762p26731188.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
On Thu, Dec 10, 2009 at 2:13 AM, Julian Lyndon-Smith aster...@dotr.comwrote:
Sometime next week I hopefully am going to start a document that
follows my progress in setting up a FS system from scratch, with all
the pitfalls and successes. A kinds of warts and all story.
Alongside this blog
On Thu, Dec 10, 2009 at 3:40 AM, Fred-145 codecompl...@free.fr wrote:
No publisher, although uploading and selling books (deadtree or online) is
easy with companies like www.lulu.com
I was just thinking of some way to learn FS gradually and effectively. The
frequent problem with wiki's, is
On Thu, Dec 10, 2009 at 8:42 AM, Otis ab...@greatiam.com wrote:
I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set
as gateways and register with each other. I wanted all users on FS1 to
dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as
331001 on
On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero aep.li...@it46.se wrote:
Hi,
I am currently creating IVR using the functions provided in the XML
dialplan
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
Using functions like this
entry action=menu-play-sound digits=1
Use the BKW method... three to four word sentences to describe what to
do... its very poetic! Or is that haiku?
/b
On Dec 10, 2009, at 11:53 AM, Michael Collins wrote:
I was just thinking of some way to learn FS gradually and
effectively. The
frequent problem with wiki's, is that the
we also support natpmp and static ip setting.
Mike
On Dec 10, 2009, at 12:21 PM, Fred-145 wrote:
Thanks for the clarification. So it's either UPnP or STUN/port-mapping.
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On Thu, Dec 10, 2009 at 10:17 AM, Brian West br...@freeswitch.org wrote:
Use the BKW method... three to four word sentences to describe what to
do... its very poetic! Or is that haiku?
/b
Update to latest
Did you type make current yet?
Tony hates build skew
-MC
I have an application where I would like to route both calls and other
requests through the same queue to the same agents, for example the same
agent might take a call and then right after that take a chat. But, the chat
server we use is separate from our phone system.
What I would like to do is
I want to trigger CUSTOM events via ESL as they navigate inside of the IVR.
The XML IVRs are generated from a GUI.
The CUSTOM events need to carry
- what IVR the user is navigating
- what option has been selected
- ideally how long they stayed listening (this can be calculated)
- and when they
ok, but how much smultaneous calls did you get on an alix board using
astlinux... for istnace, this is the question?
T.
On Thu, Dec 10, 2009 at 5:12 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
On Thu, Dec 10, 2009 at 9:26 AM, Frank Carmickle fr...@carmickle.com
wrote:
On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:
Update to latest
Did you type make current yet?
Tony hates build skew
Brilliant.
Michael Collins-san
Shrinks all usual advice
Into one Haiku.
--Dave
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On Thu, Dec 10, 2009 at 1:11 PM, David Knell d...@3c.co.uk wrote:
On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:
Update to latest
Did you type make current yet?
Tony hates build skew
Brilliant.
Michael Collins-san
Shrinks all usual advice
Into one Haiku.
--Dave
I
Ok. The journey begins.
http://makingfs.blogspot.com/
Don't know if you want to add this link to the website or wiki.
Julian
2009/12/10 Michael Collins m...@freeswitch.org:
On Thu, Dec 10, 2009 at 1:11 PM, David Knell d...@3c.co.uk wrote:
On Thu, 2009-12-10 at 10:26 -0800, Michael Collins
On Thu, Dec 10, 2009 at 10:26:39AM -0800, Shaun Clark wrote:
I have an application where I would like to route both calls and other
requests through the same queue to the same agents, for example the same
agent might take a call and then right after that take a chat. But, the chat
server we
On Thu, Dec 10, 2009 at 2:16 PM, Julian Lyndon-Smith aster...@dotr.comwrote:
Ok. The journey begins.
http://makingfs.blogspot.com/
Don't know if you want to add this link to the website or wiki.
Julian
Excellent work! Thanks,
-MC
Asterisk deadlocked
Why does it suck so badly?
Use
I downloaded yesterdays latest pre compiled and seems to works great, but I get
invalid Asr module when trying to run pizza app.
It seemed to come pre configured with pocketsphynx, anything I should know
before I spend a boat load of time on it?
Rest seems real good,. thatks!!!
Sorry, I now see it wasn't loaded, so must not come with the pre compiled.
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FYI,
Here's the agenda for tomorrow's conference call:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11
Please be ready to join at 11AM CST! :) Don't forget to bring your agenda
items, questions, and a willingness to help out with our various janitor
projects.
Thanks!
-MC
On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote:
Lack of OpenZAP support might be an issue, I assume that would be
required to connect to an onboard analogue port... I assume I could just
install Debian or another distribution instead though.
This is another distribution I found:
As a note, we are pretty aggressive about making sure all this stuff works
right out of svn without any patches so it should be easy to port freeswitch to
most platforms now.
Mike
On Dec 10, 2009, at 8:57 PM, Brian May wrote:
On Thu, Dec 10, 2009 at 03:53:32PM +1100, Brian May wrote:
Lack
On Thu, Dec 10, 2009 at 09:20:39PM -0500, Michael Jerris wrote:
As a note, we are pretty aggressive about making sure all this stuff works
right out of svn without any patches so it should be easy to port freeswitch
to most platforms now.
Thats good to hear.
I am guessing this means I should
Anyway I sent an email to Yawarra to ask them if the net5501 computer
http://www.yawarra.com.au/product.php?productCode=HW-NT55 is
compatible with the TDM400 cards.
It is, people have been doing this for a while w/ astlinux:
Hi!
My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to execute a 'send sms' script.
I don't want a separate statement in the dialplan after the voicemail
statement because I only want to send sms's when a voicemail is
actually left.
The
That wont work.
I'm not sure if there is a way, I cant think of one off the top of my head.
On Thu, Dec 10, 2009 at 10:10 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
My voip provider provides a SOAP interface to be able to send SMS's,
so after a voicemail is left, I want to
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