try installing php-devel
-Ray
On Nov 20, 2009, at 12:51 PM, Costa Zikalala wrote:
I've been trying to make phpmod without any success. I've even tried
to ./configure --with-php and it did't help.
I've just upgraded to latest svn and am running FS on FC11.
I keep getting this error
you need additional libs... editline-devel or something similar.
alternatively, you can remove them in the Makefile
-Ray
On Nov 20, 2009, at 1:50 PM, Costa Zikalala wrote:
Thanks for quick responses guys, yes I was doing it under the php
directory.
I've now also installed php-devel, and
yes you can, and yes it should go to -users not -dev
just make sure you have multiple-registrations turned on in the
profile you want to allow that on
Raymond Chandler
http://freeswitchsolutions.com
http://cluecon.com
http://cudatel.com
On Sep 3, 2009, at 12:21 PM, Jerry
Could you load freeswitch with a couple hundred calls then run the
test again.. and do the same to asterisk and see how the numbers stack
up then? I'm just curious to see what happens at that point.
-Ray
On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
Hi Everyone,
I'm working on a PBX
a proper load balancer.
Raymond Chandler
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UNSUBSCRIBE:http
to which the gateway to the faulty carrier is
attached should fix the problem. You shouldn't actually need to hard
code any IPs into your hosts file.
Raymond Chandler
http://freeswitchsolutions.com
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is as defined in [1].
Raymond Chandler
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I think i remember needing to send the callerid in the From: when
using them, try throwing
param name=caller-id-in-from value=true/
on the gateway and restarting the profile to see if it works after that
Raymond Chandler
http://freeswitchsolutions.com
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http
it.
Check out the deflect app
Raymond Chandler
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Gu Sh wrote:
What is the current status of IAX support on freeswitch?
Basically unsupported, the module still builds and sort of works, but
there isn't much interest in maintaining it. It would definitely be a
lot better for IAX lovers if someone wanted to take on maintaining it,
or fund its
you could hang on the event socket and catch the conference events, then
play the sounds via the conference api commands
-Ray
Rudolf Denert wrote:
Hallo everybody!
I would like to play soundfiles in a existing conference.
The procedure is this:
Someone calls the number of the conference.
Eli Hayun wrote:
Is there is a way to initiate a call without making any dial manually?
i think the api command originate is what you're looking for
-Ray
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Brian West wrote:
Kaiden Anthony Chandler arrived Friday July 3rd at 1411 EDT 7lbs 10oz
YAY... Congrats mr Lanman!
/b
THANKS BRIAN
And the few others that sent money. The hospital is a little over an
hour drive, so the money definitely helped out with gas, etc... You
really are a
On 07/01/2009 08:29 AM, Muhammad Danish Moosa wrote:
Hi
Freeswitch is being used in a scenario where two endpoints are running
traffic with bypass media mode. Performance is good and all things are
smooth.
But as the time goes after starting freeswitch, it starts consuming
almost whole
There doesn't seem to be any direct link between FreeSWITCH and Howler.
How does this benefit the project? Can we assume that Howler is giving
something back?
-Ray
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On 06/25/2009 01:04 PM, paul.degt wrote:
I will have to ask my boss about that, most probably he will ask same in
That doesn't really make sense... FreeSWITCH isn't using or benefitting
from your software... but yours is from FreeSWITCH
-Ray
On 06/25/2009 11:24 AM, paul.degt wrote:
You can use FS XML Curl - FS sends XML CDRs to a web server of your
choice, and there you do whatever you want with these CDRs, like store
in a database.
There are also pre-built solutions available, check here:
windmills
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Peter P GMX wrote:
Hello Brian,
this is too easy :-).
This is for a small callcenter app and I only want the user to pickup
the call once (to accept the call in X-Lite (or a Snom phone) and to
start the workflow on the web application). I do not want him to accept
the call on the phone and
Bradley Brashier wrote:
How much power do I have with DTMF conference controls? The wiki
doesn't have much information on this. For example, one of the things
I'd like to do is take the currently existing lock and unlock
actions and merge them into a lock toggle action. Preferably in XML
Saeed Ahmed wrote:
No idea at all,
It's a commercial SBC.
I wish if we can have same functionality in FS.
You could accomplish parts of this with hearbeat and ldirectord the
in-session calls aren't going to go anywhere, but if the server crashes,
the second one can take over the ip of
jcro...@gmail.com wrote:
Thoughts anyone?
You can't please all of the people all of the time
-Ray
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Klaus Teller wrote:
Hi Team,
I'm still in need of a way to reject a call without answering it. I very much
appreciate your help.
Klaus.
action application=respond data=503 Unanswered/
-Ray
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Durk de Beer wrote:
Hello all,
I have observed an issue on using Freeswitch and some SIP-phones. Ok
the problem is this. Some phones
which phones?
-Ray
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I think you're more likely to get more help if you write it in PHP or
ruby. PHP seems to be the open-source web language of choice by most,
but ruby/rails has been gaining a lot of ground, so either would
probably be a safe bet.
-Ray
Matthew Lockwood wrote:
KeeblerElfMatt ... I'll be there
we created a #freeswitch-gui channel for talks such as these... I'm the
only one idling in it most of the time, though :-(
-Ray
Michael Collins wrote:
On Thu, Jun 4, 2009 at 2:39 PM, Matthew Lockwood
matthew.lockw...@gmail.com mailto:matthew.lockw...@gmail.com wrote:
I'm about to
bakko wrote:
Ok.
But why i receive a 407 response if no call in progress or active.
REMEMBER: ngrep is your friend.
-Ray
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Matthew Lockwood wrote:
No, FS is installed on a VPS. I'm just connecting through my cable
connection.
SPI or SIP ALG on the router?
-Ray
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Matthew Lockwood wrote:
Is that stable enough to use in production?
it's more stable than not working
-Ray
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Michael Collins wrote:
On Mon, Jun 1, 2009 at 10:11 AM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
It's always august =D
it's ok. it counts as work!
If you need a letter to your boss we can do that on ClueCon letterhead. :)
-MC
my
- one for my wife
bring her too
- one for my bank (asking more credit)
we can write it... but i won't guarantee that they give you what we ask :-P
-Ray
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speak for yourself... the Dr. Pepper PJ's are bad enough ;-)
-Ray
SP wrote:
I think we'd like to see the outfit!
On Tue, May 19, 2009 at 14:53, Brian West br...@freeswitch.org wrote:
FreeSWITCHers,
Is it too much to ask for the community to help promote the project when
possible? Now I
chenexyee wrote:
1. user A is in conversation with user B, and at this time, a incoming
call from user C comes to A, in this case, I want freeswitch to play
busytone to C, how to configure?
you could use the limit app (mod_limit) to limit A's number of calls to
1, then play the busy sound with
Meftah Tayeb wrote:
Hello,
please anyone here a implemented CURL for freeswitch using any Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL but
is not working.
not working is relatively loose. Do you have any more information than
that?
Bipin Patel wrote:
hi,
will the below work if all the registration that is to be accepted come
from different public ip addresses, i mean, clients from all ip ranges
and addresses rather than a single ip
yeah, that's kinda why its called blind ... you don't have to know
where its
seven wrote:
I know that. And I'd like to read code. Developers written great code
and also plenty of comments(which is documentation) in code. However,
there are sth. don't need to comment in code but should be available
on wiki. E.g. I followed the svn commit log, and found
mszla...@aol.com wrote:
Pardon me, but you speak only for yourself. I think Janitor is not an
appropriate word.
you're welcome to your opinions, no matter how wrong they are
Second, 'marketing and sales' does not only mean making money. It also
means 'selling' someone on the idea of trying
mszla...@aol.com wrote:
You missed the point again. But suffer fools to long.
No, I think you missed the point... several times. The point that most
of us are trying to make is if you're not going to help, you have no
room to talk. Although, I guess your approach works for you. If you're
HarryK wrote:
I have Cepstral working.
Can someone please tell me how to go about having it read RSS feeds? I can
have the dialplan direct it np. But I really dont have a clue how to point
it at an RSS. Any help would be great, ddint find anything in the wiki.
have you tried mod_rss?
Tamas wrote:
My guess is: pbm = problem :)
sure, but is it really that hard to spell all the way out?
-Ray
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Richard Lamkin wrote:
Dear All,
Does anyone have any load test results using XP(SP3) -- I have checked
the wiki but could not find any recorded.
Regards
Richard Lamkin
for the most part, we try to steer clear of posting load test results
since my testing may be different than
Harry FSwitch wrote:
If it takes off and provides a friendly, helpful entryway for new
FreeSWITCH users then I will be happy, if it flops I'll be said. But I
will maintain it and do what I can to help FreeSWITCH grow. This is
the only time I'll overtly mention it on this list, I'll have
SP wrote:
On Wed, Mar 11, 2009 at 11:32, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
On Mon, Mar 9, 2009 at 11:09 PM, Will Smith willbel...@yahoo.com wrote:
Thank you Brian, it works like a champ.
Yes, west philadelfia born and raised?
On a playground is where I
Anthony Minessale wrote:
try now using sip_from_display variable to control the display field
in the from url
It would be awesome if someone from the community would document all of
these nice new variables on the wiki.
-Ray
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and it will probably be a good idea to do
make phpmod-install
so that the .so and .php files gets into the correct place to be included
-Ray
Mathieu Rene wrote:
You need your distro's php dev pakage.
On 26-Feb-09, at 6:25 AM, Rex_Alex wrote:
Hi All, I tried svn up ./bootstrap.sh
Ali Al-Rubaie wrote:
Thanks Brian,
Sorry if the question looks primitive but in which file I can find the
rvn?
you can find the revision from typing version at the fs_cli
Is there any tarball with the latest revisions?
the 1.0.3 tarball was just rolled yesterday, give it a shot
-Ray
did you ./configure --enable-core-odbc-suport... those errors reek of
that flag with no unixODBC-devel package installed
-Ray
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Brian West wrote:
Don't they cryptographically sign the config also?
it's an option in the device... some providers do, some don't.
but it shouldn't matter too much if they're using https or not, as long
as the ata doesn't authenticate via certificate or something.
-Ray
Nik Middleton wrote:
I'll shortly post some docs on the php fs_sock.
don't waste your time... There's a php .so for ESL now, and i'll
probably be removing the fs_sock from tree sometime very soon... maybe
replacing it with some specific api classes... i'm not sure on that part
yet.
-Ray
pauld wrote:
The libs are there with correct symlinks, see below. I tested both
voices directly via swift command, works fine.
Any other ideas?
It's Cepstral 5.1, FS 1.0.2.
Unpredictable issues have been reported using cepstral 5 with
FreeSWITCH. I'd suggest using their 4.x release. If
kokoska rokoska wrote:
Just my 2c:
If you use curl with lighttpd and custom built fastcgi C responder (it
is really simple with fcgi libs - even I can do it :-) performance could
be not that bad.
hmmm, mod_xml_curl using C, interesting thought.. all of the
complexities of writing your own
Its the firewall... some fw's will give you port 5060 if there's nothing
else in the network using 5060, but others will randomize it always.
-Ray
Jonas Gauffin wrote:
I'm behind NAT.
Is it FS that picks the random port, or the FW?
I've mapped port 5060 to the freeswitch ip in my FW.
On
Apostolos Pantsiopoulos wrote:
When I am using the following method to place a call from the dialplan :
sofia/gateway/gateway/number_to_dial
how do I tell FS which profile to use (as in the
sofia/profile/extension method?)
I am asking that because all my calls to my declared gateways use
kokoska rokoska wrote:
Hi all,
I have just post two bounties
Where did you post these bounties? We've started moving bounties away
from the wiki and adding them to jira instead. ( so that progress can be
followed more closely )
-Ray
begin:vcard
fn:intralanman
n:Chandler;Raymond
adr:;;630
kokoska rokoska wrote:
Raymond Chandler napsal(a):
kokoska rokoska wrote:
Hi all,
I have just post two bounties
Where did you post these bounties?
I have posted them to the Boutny wiki page (at the bottom of the page):
http://wiki.freeswitch.org/wiki/Bounty
fidibus83 wrote:
Hello,
I need your help again!
I want to use ODBC in javascripts.
First I installed unixodbc and I did all the steps what are mentioned
in wiki: mod spidermonkey odbc.
But when I test my ODBC Setup by running isql I get this error:
[ISQL]ERROR: Could not
...@lists.freeswitch.org] *Im Auftrag von
*Raymond Chandler
*Gesendet:* Dienstag, 6. Januar 2009 16:13
*An:* freeswitch-users@lists.freeswitch.org
*Betreff:* Re: [Freeswitch-users] Problems with ODBC
fidibus83 wrote:
Hello,
I need your help again!
I want to use ODBC in javascripts
Jason White wrote:
This patch implements the /help idea, but I'm still not sure that it's the
right solution.
jira would be a better place to put this
-Ray
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FERNANDO VILLARROEL wrote:
Dear All,
I am new on this list, i am chilean.
I come from Asterisk and my company is dedicated to wholesale, i am provider
for chile mobile.
We will need implement FreeSwictch before Asterisk for Retail and Wholesale
How i will can starting, please recomended
can_...@gmx.de wrote:
Hello,
thank you for your answers. I am a step further now, it seems that just the
condition tags as described in the wiki aren't enough. After sending the
following xml I think I get stuck at the point Micheal mentioned:
The $$ substitutions are only done in the
Gabriel Kuri wrote:
I've tried to do the same and in my own experience, most carriers don't
accept 302 redirects. What I've seen is they take the 302 as a failure
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to
Carole O. wrote:
Hello,
Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it
posting relevant pieces of your dialplan and sofia configs would
probably help a bit.
-Ray
Jonas Gauffin wrote:
Yeah I know. That's what I'm trying to do, but with the domain name
(sip-corporate2.tele2.se http://sip-corporate2.tele2.se) instead of
the ip.
I'm not sure that the gateway
I think you misunderstand.
It's these files:
freeswitch.history
freeswitch.pid
freeswitch.xml.fsxml
That I feel would be better off in the db folder. They are not logs, and
should not be rotated.
if freeswitch.history isn't a log, what is it? seems to me taht it's a
the audio going one way, but not the other. So, I
can
hear the person that I just transferred to, but they can't hear me.
Anyone have any other ideas?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Raymond
Chandler
Sent: Wednesday, December 10, 2008 3:41 PM
Alexandru Nedelcu wrote:
Hi,
I'm a newbie trying to configure freeswitch.
Our needs are simple: for starters we need to call external phones from
freeswitch and play a wav file.
Now, we have the voice provider's IP (no user/password authentication),
and we have the codec used (called
use the start_dtmf app to get inband dtmf
-Ray
Alexandru Nedelcu wrote:
On Thu, 2008-12-11 at 17:25 +0200, Alexandru Nedelcu wrote:
Hi,
I've tried a simple example from here:
http://wiki.freeswitch.org/wiki/Session_streamFile
You can view my code here:
http://paste.scsys.co.uk/paste
try blocking ICMP packets TO the MSS i had this exact same problem a
few months ago MSS starts sending RTP to FS before FS is ready to
accept so the OS catches the port not open and returns an ICMP 3:3
back to the MSS which in turn chokes on the queued up RTP and
refuses to
i saw this in the first email
If you wish to donate please paypal [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] that'll help
-Ray
[EMAIL PROTECTED] wrote:
How do I donate?
-Original Message-
From: Brian West [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Cc: [EMAIL
if you built from source, make sure you use the
--enable-core-odbc-support option on your ./configure
like so
./configure --enable-core-odbc-support make make install
-Ray
preetha Ayyappan wrote:
You mean i have to uncomment the line load
module=mod_spidermonkey_odbc/.?
yes.I have
Peter P GMX wrote:
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : Invalid profile
My assumptions for a right xml answer back to FS are as follows, but I
think at least one of it is false:
1. )
This definitely works. I've set up mod_xml_curl in a number of places,
managing anywhere from 10-1 gateways... as long as you have the url
correct, and you spelled configuration correctly in the xml_curl.conf,
then everything should work fine.
-Ray
Peter P GMX wrote:
Nobody has an idea
/freeswitch.xml.fsxml
Am I wrong with my expectation?
Best regards
Peter
Raymond Chandler schrieb:
This definitely works. I've set up mod_xml_curl in a number of places,
managing anywhere from 10-1 gateways... as long as you have the
url correct, and you spelled configuration correctly
Matt Darnell wrote:
On Tue, Jun 10, 2008 at 4:58 PM, Raymond Chandler
[EMAIL PROTECTED] wrote:
Here's the agenda for this week's conference call. We hope to see a larger
turnout than last week. Looking forward to hearing from everyone at 1PM
CDT... 2PM EDT ... 6PM GMT
Here's the agenda for this week's conference call. We hope to see a
larger turnout than last week. Looking forward to hearing from everyone
at 1PM CDT... 2PM EDT ... 6PM GMT
* 1 Brief review of progress on dead end page cleanup from last
week's call
Rizwan Hisham wrote:
Hi all,
Can i use asterisk's extensions.conf with freeswitch?
yes
If yes, How?
http://wiki.freeswitch.org/wiki/Mod_dialplan_asterisk
--
Best Regards
Rizwan Hisham
Everyone remember the call that's coming up in about two hours. The
agenda was sent out yesterday and is attached below for your reference.
We look forward to hearing from everyone then.
Thanks,
-Ray
* 1 we still need to finish adopting the orphaned pages
is it still using wikipbx?
andrew wrote:
I released the latest version of trixswitch today. Its
a fast way to get FreeSWITCH 1.0 running.
trixswitch.org
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whoops, probably have to wait till 0.5 for that one ;-)
Brian West wrote:
scripts/socket/fs.pl Needs the FreeSWITCH::Client package in that
directory installed.
/b
On Jun 3, 2008, at 4:08 PM, andrew wrote:
That's a good question. I have not figured out how to
open a CLI on FreeSWITCH.
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