Sent from my iPhone
On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy
lakindi...@gmail.com wrote:
No. In the dial plan I said, application=perl data=The perl
script.
I also checked $session-execute(bridge,user/1010). This is
working fine.
But originate is not working as I expected.
Bkw, I would recommend charging a fee from callcentric for the
consultancy. This consultant thing can get you going someday! LOL
Jmesquita funny joke
On 8/28/09, Carlos S. Antunes c...@nowthor.com wrote:
Brian,
You've been vindicated. Callcentric is now advertising zero weighted SRV
records!
The following is the requirement.
Let say I have 3 extensions in my freeswitch. (777,1000,1010).
When 1000 made a call to 777, I'll execute a perl application.
In that, I'll get the DTMF (1010) from the caller. Then I'll make a call to
the entered digit and I've to bridge 1000 and 1010.
This is
Hello Michael!
#Does this Python code helps:
new_api_obj = API()
result = new_api_obj.executeString(show channels)) # Replace
show channels with a command of your choice
# By the way, I use the command below to connect an existing and answered
leg to a second one:
Hi,
In a new environment, I am getting the following error when building the
latest freeswitch from svn. Does anyone know how to resolve it?
gcc -I/usr/src/freeswitch-snapshot/src/include
-I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden
Hello,
the trace seems good.
If you check the answer from Kamailio, you'll see that Kamailio answers
with 302 PEER_01.
As Michael Collins stated before, you can get the variable containing
PEER_01, then this variable is stored in a custom variable.
In your dialplan, may you please add:
Hey I'm sorry. I've solved this as follows
my $sess=new freeswitch::Session(user/1000);
$sess-answer();
#$sess-waitForAnswer($session);
if($sess-ready())
{
freeswitch::consoleLog(INFO,Session is answered\n);
Hello,
we have a strange problem, sometimes calls are hung up with
DESTINATION_OUT_OF_ORDER cause.
I added some extra log lines, and it turned out that
switch_ivr_bridge.c:140 a timeout occurs while waiting for CF_BRIDGED flag
switch_channel_wait_for_flag(chan_b, CF_BRIDGED, SWITCH_TRUE, 1,
Cool, feel free to report these bugs in Jira. If they are indeed FreeSWITCH
bugs.
On Fri, Aug 28, 2009 at 12:10 AM, Jason White ja...@jasonjgw.net wrote:
Jay Binks jaybi...@gmail.com wrote:
Reason I ask ... I personally only have a preference for debian,
but others may have policy
The wiki page is updated (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
it is understandable :)
Anatoliy Kounitskiy
On Thu, Aug 27, 2009 at 9:30 PM, Michael Collinsm...@freeswitch.org wrote:
Thanks! I'll be in touch to help you if you need any wiki editing
assistance.
i've also got a list of voicemail-related prompts that i'm going to
use callie for. i'm willing to throw my stuff into the pot, too, if
we can have this turned around in two weeks. otherwise i'll probably
have to press on, as i have a deadline to make. lemme know...
On Aug 27, 2009, at
Dear friends,
I am using freeswitch for the last one month and trying to write perl
scripts to do some tasks.
I am trying a new thing, which is something that I don't how to do it and I
don't whether it can be done with freeswitch. my requirement is as
follows,
A calls B[This
Thats still debatable The RFC's on these things read like chinese
stereo instructions.
/b
On Aug 28, 2009, at 12:13 AM, Carlos S. Antunes wrote:
Brian,
You've been vindicated. Callcentric is now advertising zero weighted
SRV records! :)
I've re-enabled SRV lookups for the
Yes you own the recordings. Its work for hire. Do you want to
include the recordings in our sound repo?
/b
On Aug 27, 2009, at 3:13 PM, Carlos S. Antunes wrote:
Mike,
Sure! I am planning on doing a session soon, maybe in a couple of
weeks or so. My only question is whether GM Voices
10 seconds is a pretty long time for the 2nd half of the bridge to start?
You can try increasing it but you probably have a more profound problem if
you are hitting that code.
When you start the bridge the code has 2 session pointers, it sets up state
handler on the b leg and changes state to
get
Nagalenoj,
This is definitely possible to do with FreeSWITCH. However, it is not
exactly simple. You have some learning to do. My recommendation is to learn
about event socket. Most likely you'll need to use outbound event socket.
(Read about event socket and ESL, the event socket library, on the
Doh! I just noticed you already attached the log. My bad. Let the experts
check it out.
-MC
On Fri, Aug 28, 2009 at 9:10 AM, Michael Collins m...@freeswitch.org wrote:
On Fri, Aug 28, 2009 at 2:15 AM, Mathieu Parent math.par...@gmail.comwrote:
Hi,
I just started testing Freeswitch, this
Thanks! I made very minor changes. We appreciate you helping out with
documenting new features.
-MC
On Fri, Aug 28, 2009 at 4:13 AM, Anatoliy Kounitskiy
anato...@kounitskiy.com wrote:
The wiki page is updated (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
it is
Hello Rod,
I did the change.
Here is extract of console:
-
variable_continue_on_fail: [true]
variable_sip_h_X-ROUTE: [LOOKUP]
variable_export_vars: [sip_h_X-ROUTE]
variable_signal_bond:
Hi Mathieu,
On 08/28/2009 05:15 PM, Mathieu Parent wrote:
Hi,
I just started testing Freeswitch, this works well. This is powerfull
and easy to administrate.
I just have one problem and one request. The request is BOUNTY-5: IMAP
integration of voicemail (I will try to propose a patch if
Julian,
It might be better if you gave us the run-down on what your current Asterisk
extension does. Most likely there is an elegant way of doing it in
FreeSWITCH. Part of the unlearing of the Asterisk way is coming to grips
with the fact that the FS dialplan is not a programming language in and
Now you have to actually catch this in your dialplan on FreeSWITCH and
execute the bridge application to the dialstring provided
/b
On Aug 28, 2009, at 11:42 AM, Hristo Benev wrote:
variable_sip_redirect_dialstring_0: [sofia/internal/
sip:fra...@peer_01]
variable_sip_redirect_dialstring:
As a background, I ran an asterisk consulting company for about 3 years that I
gave up on 2 years ago after repeatedly failing to achieve any sort of
stability on any sort install over about 30 phones, I gave up.
Maybe that was wrong, I am open to the possibility that I just didn't know
On Tue, Aug 25, 2009 at 10:15 AM, Shawn L. Djernes sdjer...@gmail.comwrote:
Woops,
That line was an attempt to try something from an older diaplan example.
Copied and pasted from a file on my computer not the server. The actual
line is:
action application=bridge
Hello List,
I have read the current thread about scalability and I would need some
advice about a callcenter setup:
First where I come from:
I have lot's of problems with an asterisk solution. I have regular
crash's and lock-ups, with downgrading and other stuff i got it
somewhat stable,
Tom,
Welcome! Sadly, your experience is not unique...
On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom paveraw...@hotmail.comwrote:
As a background, I ran an asterisk consulting company for about 3 years
that I gave up on 2 years ago after repeatedly failing to achieve any sort
of stability
Hi Tom,
best is to try it for yourself, you cannot expect from the FS mailing
list an answer like: you know, fs is nor a marked improvement on
anything, we just like to spend time together :-)
-gm
On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tompaveraw...@hotmail.com wrote:
As a background, I
On Fri, Aug 28, 2009 at 11:55 AM, Giovanni Maruzzelli
gmar...@celliax.orgwrote:
Hi Tom,
best is to try it for yourself, you cannot expect from the FS mailing
list an answer like: you know, fs is nor a marked improvement on
anything, we just like to spend time together :-)
Although, in
Usually you don't need to worry about stability issues with FS.
For scalability, peoples tend to use openser or some other sip
loadbalancer in fron of fs, but you probably would not need that.
Live migration of calls is not yet possible, tough.
-giovanni
On Fri, Aug 28, 2009 at 8:52 PM,
http://www.freeswitch.org/node/117
That's essentially the story of why I wrote FS.
On Fri, Aug 28, 2009 at 1:54 PM, Michael Collins m...@freeswitch.org wrote:
Tom,
Welcome! Sadly, your experience is not unique...
On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom
-users
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Hi folks,What linux distro would you suggest to deploy FS with django:
Centos 5.3 or Ubuntu LTS?
Thanks
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please use CentOS 5
On Fri, Aug 28, 2009 at 2:33 PM, Fernando Testa
te...@voicetechnology.com.br wrote:
Hi folks,What linux distro would you suggest to deploy FS with django:
Centos 5.3 or Ubuntu LTS?
Thanks
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On Fri, Aug 28, 2009 at 12:11 PM, Giovanni Maruzzelli
gmar...@celliax.orgwrote:
Usually you don't need to worry about stability issues with FS.
For scalability, peoples tend to use openser or some other sip
loadbalancer in fron of fs, but you probably would not need that.
Live migration of
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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Debian :)
--
Stopping junk mailers is good for the environment
On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa
te...@voicetechnology.com.br wrote:
Hi folks,What linux distro would you suggest to deploy FS with django:
Centos 5.3 or Ubuntu LTS?
CentOS 5.3 FTW!
-MC
I am totally fine without a slick GUI interface. The first 2 years of asterisk
stuff I did was all in on the CLI in editor of your choice (I use vim most of
the time, but not for religious reasons...).
Anyway, thanks for the info, I'll be setting up a freeswitch system this
weekend expect to
OpenSolaris :P
Am 28.08.2009 um 21:50 schrieb Alberto Escudero:
Debian :)
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On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa
te...@voicetechnology.com.br wrote:
Hi folks,What linux distro would you suggest to deploy FS with
django:
Welcome on board Tom! And sorry for being witty before ;-)
-gm
On Fri, Aug 28, 2009 at 9:54 PM, Christensen Tompaveraw...@hotmail.com wrote:
I am totally fine without a slick GUI interface. The first 2 years of
asterisk stuff I did was all in on the CLI in editor of your choice (I use
vim
Thanks Michael. What the extension currently does is :
A) Add 1 to the current list of Queue calls (we have a blended system
where I need to dynamically allocate agents from outbound to inbound)
This counter is maintained in the phone system
B) call my application via a web call (CURL) to tell it
Centos 5.3 / Solaris . of course
And I am looking at http://moblin.org/ for tiny embedded system.
Best regards,
-E
Gpro.ws
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Gladly using FreeBSD 64 bit for a 35 ext PBX
On Fri, Aug 28, 2009 at 2:07 PM, EdPimentl edpime...@gmail.com wrote:
Centos 5.3 / Solaris . of course
And I am looking at http://moblin.org/ for tiny embedded system.
Best regards,
-E
Gpro.ws
no seriously, use CentOS
None of these other guys have to support FS ;)
If you plan on doing it all without asking for help, then you are welcome to
any OS that FS compiles on. =D
On Fri, Aug 28, 2009 at 3:07 PM, EdPimentl edpime...@gmail.com wrote:
Centos 5.3 / Solaris . of course
And
Hi Christensen,
Welcome, you made a good choice on FreeSWITCH, and FS is much better at
those things than Asterisk.
Good luck, we are here to help you and tell us your experience later :).
Diego
On Fri, Aug 28, 2009 at 6:14 PM, Christensen Tom paveraw...@hotmail.comwrote:
As a background, I
the L16 codec is used because the wav files is raw PCM audio
which is being encoded to PCMU
Have you tried doing this to a locally registered phone?
On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones pjinthe...@gmail.com wrote:
thanks for the reply.
Tried this and exactly the same result. The
nevermind,
I didn't look closely at your code.
how did you end up with blegSession ?
Are you running this script from the CLI or via the application interface.
On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
the L16 codec is used because the wav
Hi Raimund,
One FreeSWITCH box will be quite enough to handle the call volumes that
you're talking about, and it ought to be much more stable than the
Asterisk solution which you've outlined below.
It's probably best to forget about live failover without calls dropping
- this isn't something
Taking over the session while it's parked like that from your code and
asking it to play a file is making it do 2 things at once.
The session's thread is already busy in the park loop so you would have to
change it's state to something passive like soft_execute so
it's thread was not doing
I currently use FS for a managed PBX services for many companies. It's so
configurable and extensible, I don't think I would change it for anything. I
have Cisco CME and SCCP phones in my office all interconnected to FS doing
logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does
windows 95 :P
- Original Message -
From: Giovanni Maruzzelli gmar...@celliax.org
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, August 28, 2009 10:02 PM
Subject: Re: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
Vista!
On Fri, Aug 28, 2009 at 10:00 PM, Michal
You guys should put your experiences here :).
http://wiki.freeswitch.org/wiki/Testimonials
Diego
On Fri, Aug 28, 2009 at 10:03 PM, Ognjen Seslija osesl...@gmail.com wrote:
I currently use FS for a managed PBX services for many companies. It's so
configurable and extensible, I don't think I
asterisk anymore.
-Tom
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hi,
with the java-asterisk bridge its possible to to AMI-stuff. is thi as well
possible with freeswitch? if so, are there any tutorials, docu etc. out
there?
thx
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On Fri, Aug 28, 2009 at 4:35 PM, tom tomabr...@gmail.com wrote:
hi,
with the java-asterisk bridge its possible to to AMI-stuff. is thi as well
possible with freeswitch? if so, are there any tutorials, docu etc. out
there?
thx
The AMI in Asterisk is analogous to the event socket in
Here is a copy of the data from the console at log 7
freeswi...@internal group_call
sdjerne...@ewr.djernes.netsdjernes%...@ewr.djernes.net
-ERR no reply
freeswi...@internal group_call sdjer...@ewr.djernes.net
error/NO_ROUTE_DESTINATION
freeswi...@internal group_call
Hi David,
What have you used on FS for call center, mod_fifo?
Can you describe your experience with that, I'm currently interested in call
center + FS scenario.
Diego
On Fri, Aug 28, 2009 at 9:01 PM, David Knell d...@3c.co.uk wrote:
Hi Raimund,
One FreeSWITCH box will be quite enough to
Yes, FreeSWITCH is a system that you can trust 100%. I have switched my
Asterisk servers to FreeSWITCH and have peace now.
If I were you I would get rid of Asterisk and use FreeSWITCH, FS will handle
all what you want very well.
And I agree with David, fail-over is kinda irrelevant since the FS
Hi Diego,
We didn't use mod_fifo; we built our own queues using an application
hanging off the event socket. This was partly down to typical
programmer hubris, and partly to allow us to do things that mod_fifo
might not without either (a) trawling through mod_fifo, or (b) pestering
Anthony.
As
Thanks for your help. Is it the only possible way.?
Else, will it be done with the functions listed in the below url,
http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes
I was trying to do with these classes and functions.
mercutioviz wrote:
Nagalenoj,
This is definitely
Ok,
I found what's happening. I probably did some change on the wiki to
start reflect the new configuration, without having time enough to check
the configuration. There is a mistake in your dialplan configuration.
You should put this instead:
wrong line:
action application=set
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