wrote:
That usually means they are saying 30 but sending 10 which is broken.. you
can't say hey i'm sending 30 and then send 10... find a new provider or beat
them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use
I tried removing the codec file extension from uuid_record and
session_record but I'm still unable to record a file in native format for a
bridged call.
record WORKS!, but uuid_record and session_record do not want to record in
native format. do uuid_record and session_record work with native
, 2009, at 8:53 AM, Matthew Fong wrote:
record WORKS!, but uuid_record and session_record do not want to
record in native format. do uuid_record and session_record work with
native format? or is it not going to be possible to record a bridged
call in native format?...maybe because there are two
I'm trying to conserve processor power by recording in native file format,
PCMU in my case. It works great with the following line
session:execute(record,
/tmp/my_recording...session:getVariable(read_codec));
however it fails to work with
session:execute(record_session,
I'm trying performing a uuid_record command immediately after a uuid_bridge,
but receive a Can not record session. Media not enabled on channel error.
proxy_media and bypass_media are both set to false.
The uuid_record however works if I use sched_api +1 uuid_record... but if I
do this, I of
it with
execute_on_answer and the record_session application
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 17-Nov-09, at 5:28 AM, Matthew Fong wrote:
I'm trying performing a uuid_record command immediately after
Tina,
How are you originating the calls? from the console? Try bgapi originate...
--matt
Voice Broadcasting - http://www.hellohunter.com/voice_blast.php
On Fri, Nov 13, 2009 at 12:57 AM, t...@a2unlimited.com wrote:
I'm trying to increase the number of calls per second that I can originate
I just tried the webphone with my freeswitch server and it worked fine,
making a call to my echo test w/o any issues...so it's probably a
configuration issue with freeswitch.
--matt
http://www.hellohunter.com
On Tue, Nov 10, 2009 at 4:15 AM, Michael Collins m...@freeswitch.org wrote:
On Mon,
it says.
Mike
On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
:56 AM, Michael Jerris m...@jerris.com wrote:
FreeSWITCH debug level logs should help tell you exactly what is killing
the call.
On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:
I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects
for users utilizing my inbound DID
to send ?
Mike
On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
I'm used to using the onInput callbacks inside lua and javascript to
listen for dtmf and other events and perform a task accordingly. I'm
wondering if there is a way to send an event to a session or channel
that can be caught
I think think this might be a bug, but wanted to post here instead of Jira
in-case I'm overlooking a configuration variable
Dialplan
extension name=1920!--init agent for manual and power dial mode--
condition field=destination_number expression=^1920$
action application=set
it's transferring, just
don't know why it's disconnecting the call instead of playing the .wav and
parking.
On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong mattdf...@gmail.com wrote:
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE
, Matthew Fong mattdf...@gmail.com wrote:
I think think this might be a bug, but wanted to post here instead of Jira
in-case I'm overlooking a configuration variable
Dialplan
extension name=1920!--init agent for manual and power dial
mode--
condition field=destination_number expression
doh! thanks!
On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
because the regex is on 1997 not 1999
On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong mattdf...@gmail.comwrote:
extension name=1999!--DIRECT POWER--
condition field
/sip_1) Ended
2009-10-12 17:39:36.037520 [NOTICE] switch_core_session.c:1089 Close Channel
sofia/internal/sip_1 [CS_DESTROY]
thanks.
--matt
On Mon, Oct 12, 2009 at 10:45 PM, Matthew Fong mattdf...@gmail.com wrote:
doh! thanks!
On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale
AM, Matthew Fong mattdf...@gmail.comwrote:
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
recognized (I think). Is there anyway to get an alloted_timeout to continue
after bridge (failure
AM, Matthew Fong mattdf...@gmail.comwrote:
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is
not recognized (I think). Is there anyway to get an alloted_timeout to
continue after bridge (failure
I'm used to using the onInput callbacks inside lua and javascript to listen
for dtmf and other events and perform a task accordingly. I'm wondering if
there is a way to send an event to a session or channel that can be caught
using the setInputCallback inside lua from outside the session program.
Jingwei,
the dialplan command eavesdrop does this. The person barging in can use key
presses to dynamically turn on/off voice.
--matt
Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php
Predictive
Dialer http://www.hellohunter.com
On Mon, Sep 28, 2009 at 3:38 PM, Jingwei Yang
ESL is probably the way to go tho...if you want to build a dialer.
The Dial Plans can get pretty advanced in FreeSWITCH...and if that is not
enough you might consider using mod_perl or something of that sort.
--matt
Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php
Predictive
I'm having trouble getting the channel variable state in my Lua ivr example.
I have tried both
session:getVariable(state)
session:getVariable(Channel-State)
session:getVariable(answer_state)
session:getVariable(Answer-State)
but lua reports nil for all attempts
I did a uuid_dump and it appears
at 9:36 AM, Matthew Fong mattdf...@gmail.com wrote:
I'm having trouble getting the channel variable state in my Lua ivr
example.
I have tried both
session:getVariable(state)
session:getVariable(Channel-State)
session:getVariable(answer_state)
session:getVariable(Answer-State)
but lua
Whats the best way to record only one leg of a call?
uuid_record records both channels
session_record does the same (but has a stereo option)
is there any way to record only an a-leg of the call? Thanks so much.
--matt
http://www.hellohunter.com
hosted dialer voice broadcasting
='bind_meta_app' data='2 a s
record_session::$${base_dir}/recordings/${strftime(%Y-%m-%d_%H-%M-%S)}.${caller_id_number}.wav'/
the person would have to press *2 during the call to start the recording.
2009/9/7 Matthew Fong mattdf...@gmail.com
Whats the best way to record only one leg of a call
this sound like a bug that should
be submitted to JIRA?
--matt
http://www.hellohunter.com
On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong mattdf...@gmail.com wrote:
originate {ignore_early_media=true}sofia/gateway/epik.com/914159927717
bridge(sofia/gateway/epik.com/914154650027)
is the string
function are you
using)?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
I'm trying to get FreeSWITCH to bridge two channels together through the
same external gateway
I changed
/*! Minimum time for a beep. */
#define MIN_TIME 8000
to 6500 and it seemed to work, but I'm not sure how many false positives I
will get in a real-world environment. at 4000 it fired the event like 5
times in a session, but 6500 only once. Do you think I should expect a lot
of false
I'm interested in doing some testing on the accuracy of mod_vmd (and
mod_amd) but wanted to see if anyone could provide some guidelines on the
maximum number of concurrent sessions I can record audio files to disk with
a typical EIDE drive under 64-bit linux without overloading my system.
Also,
I tried emailed Eric, seeking advice on this, but his email (the one in the
source code) is bouncing email (invalid user), so thought I would ask here
instead. If anyone has eric's new email address, I'd be interesting in it.
I did some tests with mod_vmd this afternoon, but I'm only finding
Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho, that
ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
false. Just a guess tho.
Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes
Does the file exist at /usr/local/freeswitch/conf/freeswitch.xml? does the
user you are executing freeswitch as have permission to read the file?
--matt
hello hunter - hosted predictive dialer voice broadcasting
http://www.hellohunter.com
On Wed, Aug 5, 2009 at 11:46 AM, tom tomabr...@gmail.com
), 483 (too many hops) or 484 (address
incomplete).
Do a SIP trace to sched more light on what's happening.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong
can't you just have your python script execute the sched_hangup command, and
then finish the remainder of the python script?
On Wed, Jul 29, 2009 at 12:15 AM, Apostolos Pantsiopoulos
r...@kinetix.grwrote:
Hi,
Is there a way to execute more than 1 commands in the execute_on_answer
variable? I
is there a way to have the freeswitch_http.log, log what command is being
executed across the webapi? For most requests the log looks like
127.0.0.1:17093 - freeswi...@127.0.0.1 - [29/Jul/2009:00:58:29 +] POST
200 422
but it would be useful to know more precisely what is being executed across
Hi,
I'm trying to build an application that provides statistics of calls and
call recording. Someone told me this could be done out of band with a SPAN
(?) port that would replicate SIP and media packets to a separate NIC
without having to actually pass the real-calls thru FreeSWITCH. It was
bkw,
you said Downgrading. I suspect its an issue with your lua sql module
not linking to the thread safe client. in the Jira ticket. I'm
curious how one would go about doing this. I use luasql (the default
ubuntu apt-get install) and have a similar memory problem. I suppose I
would need to
there's a reference on the wiki to a three_way dial plan command. What does
that do?
What's the best way to put 2 bridged callers into a new conference? Must I
park both uuid's first, and then transfer both to an extension that will add
them to a new conference? Is there a way to do this without
I'm trying to use xml_rpc to initiate an att_xfer on channel A (which is
already bridged to channel B), but I'm running into some issues.
I know the uuid's from both channel A and B, but the documentation I found
on att_xfer only seems to indicate a way to do this from DMTF
presses occurring on
, Matthew Fong wrote:
I'm trying to use xml_rpc to initiate an att_xfer on channel A
(which is already bridged to channel B), but I'm running into some
issues.
I know the uuid's from both channel A and B, but the documentation I
found on att_xfer only seems to indicate a way to do
Does the log show anything? if the lua script fails to execute it should
appear in freeswitch.log
On Tue, Jun 23, 2009 at 9:45 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
Scratching my head on this one, under load FS is not playing an audio file,
OR and lua
I have two providers and want to first try to originate the call with
provider A, and if that fails on certain failure causes attempt to originate
the same call with provider B.
Normally I would do this using an | in the dial string like originate
sofia/gatewayA/123456|sofia/gatewayB/123456
but I
the script is not part of a session or dial plan. :(
On Thu, Jun 18, 2009 at 11:31 PM, Jason White ja...@jasonjgw.net wrote:
Mathieu Rene mrene_li...@avgs.ca wrote:
action application=set
data=failure_causes=user_busy,recovery_on_timer_expire / and then
originate it.
Or if you're
, 2009, at 1:38 AM, Matthew Fong wrote:
the script is not part of a session or dial plan. :(
On Thu, Jun 18, 2009 at 11:31 PM, Jason White ja...@jasonjgw.net wrote:
Mathieu Rene mrene_li...@avgs.ca wrote:
action application=set
data=failure_causes=user_busy,recovery_on_timer_expire
I upgraded to 13857 today, but noticed that the channel_hangup event no
longer contain the variable_billsec header.
Is this correct, or am I crazy? Thanks.
--matt
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
with a specific
thread.
--matt
On Fri, Jun 19, 2009 at 1:38 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
are you connecting to a db with the lua?
On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong mattdf...@gmail.com wrote:
With yesterday's trunk and also a release from 2 weeks ago, I noticed
:44 PM, Matthew Fong mattdf...@gmail.com wrote:
I'm using a lua script to control an IVR, and would like to know how I can
tell if a
session:execute(bridge,sofia/gateway/blahblah);
was successful or not
it seems the response from session:execute is nil regardless if the bridge
was successful
grr...continue_on_fail...ignore my ignorance ;)
but it would still be nice getting a response back from the session:execute
bridge
--matt
On Thu, May 21, 2009 at 11:09 PM, Matthew Fong mattdf...@gmail.com wrote:
hrm...it's also seems to be that if my lua script looks like
session:execute
I'm using a lua script to control an IVR, and would like to know how I can
tell if a
session:execute(bridge,sofia/gateway/blahblah);
was successful or not
it seems the response from session:execute is nil regardless if the bridge
was successful or not
whats the best way? Thanks
--matt
You can always have your lua script fire a custom event on api_hangup...this
will only send the data you specify in your event.
On Sat, May 2, 2009 at 1:36 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi Guys,
Is there an alternative to the hang-up event that doesn’t send
There's a reference in the wiki to the sched_del API command, but it doesn't
give an example, and the console doesn't give a syntax either. Does anyone
know it? I'm interested to know how it relates to the group_name
(task-group) set in the sched_api command. If I want to delete a specific
.
--matt
On Fri, Apr 24, 2009 at 12:43 AM, Michael Collins m...@freeswitch.orgwrote:
On Thu, Apr 23, 2009 at 4:53 AM, Matthew Fong mattdf...@gmail.com wrote:
There's a reference in the wiki to the sched_del API command, but it
doesn't give an example, and the console doesn't give a syntax
Thanks I'll check it out.
One more quick but related question.
Is there ever an instance when the audio is BRIDGED before the BRIDGE event
is fired. Could this fifo issue have bridged audio immediately, but somehow
withheld the bridge event from being fired for 5 seconds? A few of my
callers were
to put the system back in a live environment
since the fix and diagnosis aren't 100% compatible. As always tho, thanks
for the really quick fix and reply. Awesome telephone framework.
--matt
On Tue, Apr 21, 2009 at 9:53 AM, Matthew Fong mattdf...@gmail.com wrote:
Thanks I'll check it out.
One
I tried using fifo in an environment with about 9 agents last week, but ran
into some issues, that I'm trying to piece together. The system is setup on
a new ubuntu 64-bit machine and it should be plenty fast to handle this
load. The delay does not occur when testing with a single agent...so it's
Hi Nicolas,
Just off the top of my head, but I think couchDB is rather large compared to
sqlite, and I think it's also geared more towards
storing dynamic datasets...rather ones that can be structured...like FS
calling data can.
But I might be wrong :)
your buddy.
--matt
On Mon, Apr 13, 2009 at
I'm doing some outbound dialing, and want to use mod_vmd to detect if a live
person picks up or a voicemail picks up. I've read the wiki, and have been
playing around with the dialplan implementation and the lua implementation,
along with capturing the mod_vmdvmd::beep event.
Using the examples
Got a few more questions about running LUA scripts, please forgive me, I'm
an absolute newbie with LUA.
If I want to subscribe to a custom event, and I use
con = freeswitch.EventConsumer(CUSTOM my::event);
I get an error. Is this because I must subscribe to the CUSTOM (only) event,
and then
the same way you do
console_log, it takes milli seconds as its arg. Note this should NOT be
used when you have a script running as a session, only when you are running
an api script.
Mike
On Mar 31, 2009, at 11:15 AM, Matthew Fong wrote:
Got a few more questions about running LUA scripts
in the syntax of the subscribe?
Thanks Michael for your help...
--matt
2009/3/31 Brian West br...@freeswitch.org
Lua has no sleep or pause ... if you read thru the lua wiki they show you
various ways to accomplish that.
On Mar 31, 2009, at 10:50 AM, Matthew Fong wrote:
I know before I asked
I'm trying to get rubymod, working to experiment with it, but I'm getting
the following error when I try to make on my Ubuntu system.
r...@ubuntu:/usr/src/freeswitch/libs/esl# make rubymod
make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x
CFLAGS=-I/usr/src/freeswitch/libs/esl/src/include
I've been playing around with using freeswitch.EventConsumer in a lua
process that starts-up when FS boots, and stays in the background. I've
setup the example on the wiki, but the example uses
session:execute(sleep,1000), and essentially loops every second until an
event is fired. I'm wondering
both work like the regular import but one is a list of vars to copy from
caller to consumer and one is a list to copy from consumer to caller.
2009/3/23 Matthew Fong re...@matthewfong.com
Thanks Anthony, for creating the transfer_after_bridge feature for me.
Your rapid actions, are the reason
.
in latest trunk
2009/3/26 Matthew Fong mattdf...@gmail.com
Hi Anthony,
So it's been 2 days since my last request, so I'm due for another one ;)
It would be nice if there was a way to execute a script (lua) on fifo
bridge. I currently rely on the channel_bridge event, but I'm worried
event based on say my fifo name, this way my event_socket only has to
read events for a specific fifo, rather than all fifos.
it's not to make more work for u :)...although it's sort of amazing
how efficient of a coder you are.
--matt
On Thu, Mar 26, 2009 at 10:20 PM, Matthew Fong mattdf
Hi Brian,
Thanks for the link...I saw that, but i'm a newbie to lua (only use it cause
of FS), and I'm a little confused how the example works.
It consumes all events? Then subscribes to a session? and then, every second
checks to see if an event has been fired for that session?
Would it be
, Mar 26, 2009 at 10:53 PM, Matthew Fong mattdf...@gmail.com wrote:
Woops, my double identity of my marketing alias isn't subscribed
correctly...-
O, then this is an error because bridge_pre_execute_aleg is not firing
on fifo bridge. I'm using
FreeSWITCH Version 1.0.trunk
I'm wondering if there's any features that allow the cron-like execution of
code inside of Freeswitch, preferably with lua--or if I am stuck using the
api interface and running the cron outside of freeswitch.
--matt
___
Freeswitch-users mailing list
Thanks Anthony, for creating the transfer_after_bridge feature for me. Your
rapid actions, are the reason I'm positive I made the right decision switch
to to FS.
I got another challenge to throw your way. In the current fifo
implementation, there's no way to identify which fifo consumer, consumes
Thanks Anthony, for creating the transfer_after_bridge feature for me. Your
rapid actions, are the reason I'm positive I made the right decision switch
to to FS.
I got another challenge to throw your way. In the current fifo
implementation, there's no way to identify which fifo consumer, consumes
, 2009 at 8:04 PM, Matthew Fong mattdf...@gmail.com wrote:
Hi Anthony,
I'm trying to use fifo in a different sense. Instead of using it for
inbound call queing, I'd like to use it for outbound call making. So
instead, my agents are waiting in the que, and once an outbound call is
connected
to make
*N (where N = 0-9) to trigger a software blind transfer.
2009/3/20 Matthew Fong mattdf...@gmail.com
Also, I would not be able to have a hang-up script do it, because in this
scenario, the fifo consumer could hang-up at any time without any prior
warning--otherwise, I could just
confused and missing something obvious, please correct me... Thanks
--matt
2009/3/19 Anthony Minessale anthony.miness...@gmail.com
This is the patch
http://jira.freeswitch.org/browse/MODAPP-237
it's not added yet.
2009/3/18 Matthew Fong mattdf...@gmail.com
I upgraded to
FreeSWITCH Version
I upgraded to
FreeSWITCH Version 1.0.trunk (12654M)
but caller is still being hungup (and not continuing on with dialplan) after
agent disconnect with hangup_after_bridge=false
Is there a separate patch I need to apply? Thanks.
--matt
On Wed, Mar 18, 2009 at 11:39 AM, Matthew Fong mattdf
I apologize if this is a double post to -dev. I'm not sure why I don't see
my message appearing, so I'm going to try again in the -user list (first
timer posting here ;).
I have a situation where it would be useful for a caller not to be hungup,
after finishing the fifo in execution (when the
2009/3/17 Anthony Minessale anthony.miness...@gmail.com
there is a patch in jira that will implement this feature about to be added
2009/3/17 Matthew Fong mattdf...@gmail.com
I apologize if this is a double post to -dev. I'm not sure why I don't see
my message appearing, so I'm going to try
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