I have a dummy question. Say, you have an outbound call to the demo
IVR as below:
originate sofia/gateway/myvoip/19876543210 5000
How do I delay the IVR response until the recipient at 19876543210
picks up the call? I tried ignore_early_media=true, which had no
effect.
Many thanks in advance.
I would have to say its YOUR system and not ours.
/b
On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:
Resolving files.freeswitch.org... failed: Temporary failure in name
resolution.
Again...
On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola diego.vi...@gmail.com
wrote:
Nope.
Is your provider answering the call before its connected? If so then
they should be shot. I can't imagine other way the call would be
answered unless you're using ignore_early_media wrong can you show
me who you're doing this?
/b
On Aug 12, 2009, at 12:58 AM, Paul Li wrote:
I have
I am actually doing a lua script for IVR as follows
-- answer the call
session:answer();
while session:ready() == true do
-- sleep a second
session:sleep(1000);
-- play a file
session:streamFile(/path/to/blah.wav);
-- hangup
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by typing
$ sed s/1000/1050/g 1000.xml 1050.xml
rescan and reload
Aww, ok.
Bad luck to me :).
On Wed, Aug 12, 2009 at 1:57 AM, Brian West br...@freeswitch.org wrote:
I would have to say its YOUR system and not ours.
/b
On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:
Resolving files.freeswitch.org... failed: Temporary failure in name
resolution.
I would say there are no changes in gender for dialects...but with so many
languages around I can't assure it 100% ;)
Samuel.
2009/8/11 Michael Collins m...@freeswitch.org
2009/8/11 João Mesquita jmesqu...@gmail.com
Mike, the gender thing will eventually have to change code, I guess. I
Please post a bug for this on jira.freeswitch.org.
Mike
On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
This is an integral part of my application. I need to have
FreeSWITCH outside of the media path as well as be able to do
multiple bridges for the same A leg.
/*WORKS*/
action
Greetings,
I have the following LUA script (at end of email) in a fresh FS 1.0.4 install.
I originally did an upgrade from one of the 1.0.4preX versions but when I came
across this issue I went fresh just to make sure there wasn't an
incompatibility with my previous config.
What I'm seeing
Not sure, but they do certainly have a reasonably large server farm for
doing processing :)
I note that sphinx4 I believe has a java example for doing dictation
transcription from an audio file (saw something on a sphinx forum or mailing
list while trawling the net).
I'm still investigating
Hi All,
I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and
1.0.4.
I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
The build fails with:
--- snip ---
make: Fatal error: Command failed for target `all-recursive'
Current working directory
Hello,
I've just had the same problem. Solved it by adding the new extension to
the default group.
i.e. In the /usr/local/freeswitch/conf/directory/default.xml file you
need to add user id=1050 type=pointer with one of the group
blocks (e.g. after the line group name=default
Kevin
Tzury Bar
I've tried to use answer command from outbound event socket and it's
working, but
the problem is that FS answering the call, but SIP Client (we tried this
with EyeBeam and CISCO 7960)
doesn't know that call was answered. So, as long as FS doesn't know what to
do with this number it then
If your seeing a segfault, please report it to jira.freeswitch.org
with a backtrace and details of how to reproduce.
Mike
On Aug 12, 2009, at 2:37 AM, Charles Boening wrote:
Greetings,
I have the following LUA script (at end of email) in a fresh FS
1.0.4 install. I originally did an
still not working, I mean, I can initiate a call from 1060 to 1000 but
not from 1000 to 1060.
1060 is just an example. This applies to all new extension I have
added (beyond to the default 1000-1019).
as you can see below I added them all to group name=support
This is how the confs look like
Edit the line below as shown (in the dialplan/default.xml file)
Original line (about line 206)
condition field=desination_number expression=^(10[01][0-9])$
Replacement line
condition field=desination_number expression=^(10[0-9][0-9])$
This will allow extensions number 1000 to 1099.
Kevin
Thanks allot Kevin.
I felt it is about a missing configuration parameter
On Wed, Aug 12, 2009 at 1:31 PM, Kevin Goldingke...@kgolding.co.uk wrote:
Edit the line below as shown (in the dialplan/default.xml file)
Original line (about line 206)
condition field=desination_number
On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by
typing
It's not Eyebeam but FS hung up the call because it have nothing to do
after answer.
You should either playback a sound, do the echo command, record, hold
the call, bridge to another channel or transfer somewhere else.
On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
I've tried to use
Hi,
In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
every 10 s. The following lines of code does not show any effect at all.
switch_scheduler_task_thread_start();
switch_scheduler_add_task(switch_epoch_time_now(NULL),
data_flush_callback,
Does anyone know how to take the epoch time in switch_event_t and convert it
into a format such as Sat Jul 5 02:44:33 2009?
Is there any existing facility that I can use for this purpose?
br,
JB
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I posted it yesterday evening: http://jira.freeswitch.org/browse/FSCORE-417
On Tue, Aug 11, 2009 at 9:43 PM, Michael Jerris m...@jerris.com wrote:
Please post a bug for this on jira.freeswitch.org.
Mike
On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
This is an integral part of my
David / Michael - thanks for your your replies. The SoftIVR example is
particularly useful. Must admit though - I was hoping not to have to
do any custom stuff at this stage.
It does appear there is no method to do this by staking bridge lines
so I will put an issue in jira to try and get
perhaps we need to add some syntax + logic to originate:
application=originate
data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)
This would acomplish the equiv of
loopback/bar,loopback/yum where bar and yum are then further expanded
in the dialplan as
Hello Peter,
I'd appreciate if you can keep the discussion going in the freeswitch-users
mailing list, there are other people there that will benefit of the
discussion or even can help. Read my comments below.
On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson
peter.ols...@visionutveckling.se wrote:
Hi Bruce,
I am having similar issues trying build freeswitch 1.0.4 on Solaris
x86 as well. I sent some information over the mailing list, and I
received a response from Michal Bielicki (attached), stating he'd test
this and direct me to the steps to successfully build freeswitch.
Just an FYI in
Of course – no problem!
I’m not using libpri support now, I don’t think it’s ported for Windows (yet)?
I’ll try it out some more, and try to detect what’s going wrong...
/Peter
Från: Moises Silva [mailto:moises.si...@gmail.com]
Skickat: den 12 augusti 2009 16:17
Till:
Hi,
The standard C function is strftime.
FreeSWITCH has some wrapped ones:
switch_apr.h:SWITCH_DECLARE(switch_status_t) switch_strftime(char *s,
switch_size_t *retsize, switch_size_t max, const char *format,
switch_time_exp_t *tm);
switch_apr.h:SWITCH_DECLARE(switch_status_t)
Hi,
I did the same thing on my side
API CALL [load(mod_skel)] output:
+OK
2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
data_flush (core) to run at 1250089698
2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
Successfully Loaded [mod_skel]
Hi there,
application=originate
data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)
I agree. However, perhaps the ideal is not to specify the carriers at
this level, as carriers are added and removed fairly often as costings
change. So it would be nice to have some sort of proxy
On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jonespjinthe...@gmail.com wrote:
Hi there,
application=originate
data=(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)
I agree. However, perhaps the ideal is not to specify the carriers at
this level, as carriers are added and removed
I will try to paraphrase my question.
Is there any possibility to answer call from CTI application and
synchronise answer with answer in SIP client?Maybe we can use SIP functions
in our CTI application instead of FS api commands?
I'm trying to find the way to make prototype of lineAnswer command
Well you can only truly answer an inbound call to FS... you can't
force answer an outbound call.
/b
On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
I will try to paraphrase my question.
Is there any possibility to answer call from CTI application and
synchronise answer with answer in
Hi,
I would like to implement a random route selection based on some arbitrary
percentage.
Does anyone know if there is any good way of doing that within freeswitch?
If there isn't any api that I can use, does freeswitch has any random
generator that I can be used for this purpose?
br,
JB
If I have two FS extensions A and B. I'm calling from A to B and want to
answer from B-side in my CTI application and to make SIP phone to be
synchronised to my CTI application. Is it possible to do it?
Brian West-3 wrote:
Well you can only truly answer an inbound call to FS... you can't
mod_lcr will do random route selection if the rates are the same. But
that gives an equal distribution. There is no weighting/percentage
supported.
On Wed, Aug 12, 2009 at 12:21 PM, Juan Backsonjuanback...@gmail.com wrote:
Hi,
I would like to implement a random route selection based on some
Sip does not support this functionality. The called device would have
to support this via some other mechanism such as ctsa which I have
seen recently someone was looking at for freeswitch. So the first
issue you must resolve is the called device needs to support some way
to do this.
I have been reading all the docs about conferences I can find and am
getting somewhat confused. What I am trying to do is set up a dialplan
where I have subscribers with extensions in the 1xx range, and then to
set an ability to have a series of conference rooms for each subscriber
in the
I just did a rebootstrap on a fs box, it turned out the new revision has
this at the end of mod_sofia.h:
char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char
*prefix);
.mine
void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const
*sip, const char
You have a merge conflict. svn revert sofia_glue.c
/b
On Aug 12, 2009, at 12:59 PM, Milena wrote:
I just did a rebootstrap on a fs box, it turned out the new revision
has this at the end of mod_sofia.h:
char *sofia_glue_get_extra_headers(switch_channel_t *channel, const
char *prefix);
I was trying to explore the documentation for the application hash
which is in the default dialplan. Its vaguely obvious what its doing,
but I wanted to be sure.
It appears to be listed as a application under dp_tools, but when I
click on it I get taken to a page that talks about limit_hash
Write you a small C app to randomly return them based on the percentages...
Currently we do something similar to this but use a random round robin based
thing using a simple sql backend and doing a select order by random sort of
thing...
Contact me off list if you need some profession help
It's modified because it wouldn't compile with those at the end of
the file
2009/8/12 Michael Jerris m...@jerris.com
The M in the version number means modified. You had local code
changes that conflicted when you updated trunk. Revert the changes to
that file and it should be fine.
Mike
hash is just like db but its all in memory.. you can interchange db
and hash.
/b
On Aug 12, 2009, at 1:25 PM, Alan Chandler wrote:
I was trying to explore the documentation for the application hash
which is in the default dialplan. Its vaguely obvious what its doing,
but I wanted to be
Ok, done and fixed, thank you very much :)
2009/8/12 Milena testeado...@gmail.com
It's modified because it wouldn't compile with those at the end of
the file
2009/8/12 Michael Jerris m...@jerris.com
The M in the version number means modified. You had local code
changes that conflicted
Hi there,
Does anyone have the URL for where I might find all the electronic
versions of the presentations made at ClueCon last week?
Thanks!
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FreeSWITCH-users@lists.freeswitch.org
They are all getting gathered up and put online...
files.freeswitch.org/cluecon_2009 just keep an eye there some of the
videos are up also.
/b
On Aug 12, 2009, at 2:18 PM, Christian Jensen wrote:
Hi there,
Does anyone have the URL for where I might find all the electronic
versions of
I to would like to put my thanks on the table. I have been going to
conferences for a very long time and often question the value of taking time
off to attend these venues. When I was asked to attend by a client I was
very hesitant. I am very pleased that I decided to attend.
Now the skeptical
Hello,
anybody has a clue what this message means?
[WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
PROGRESS_MEDIA to PROGRESS
What does VETO mean here?
Best regards
Peter
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Dave,
Thanks, Hope to see you there next year...
/b
On Aug 7, 2009, at 5:54 PM, David Knell wrote:
Just a quick note to say thanks to Cluecon's organisers for putting
together such a useful, informative and packed three days. I've come
away with a head full of ideas, a bunch of new
Remember next year we'll have more Mac Book's to give away and iPod
Touches with engraved sponsor logos on them. :)
/b
On Aug 12, 2009, at 2:57 PM, jonathan augenstine wrote:
I to would like to put my thanks on the table. I have been going to
conferences for a very long time and often
Macbook ... that's nothing, I got $1500 worth of coffee :-)
On Wed, Aug 12, 2009 at 12:57 PM, jonathan
augenstinejaugenst...@gmail.com wrote:
I to would like to put my thanks on the table. I have been going to
conferences for a very long time and often question the value of taking time
off
Hello,
I have spent the past couple of weeks toying around with FS to
evaluate the possibility of using it for a large scale conference
server for our organization. The plan is to have several FS servers
initiate calls to participants and connect them together, but not have
to transfer all of
On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX prometheus...@gmx.net wrote:
Hello,
anybody has a clue what this message means?
[WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
PROGRESS_MEDIA to PROGRESS
What does VETO mean here?
Best regards
Peter
Means that state transition
So you're the one that drank 16 gallons of coffee! Good luck sleeping!
/b
On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote:
Macbook ... that's nothing, I got $1500 worth of coffee :-)
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Isn't progress_media already past progress in the state machine? so
the state machine can't move backwards in states right?
/b
On Aug 12, 2009, at 3:33 PM, Moises Silva wrote:
On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX prometheus...@gmx.net
wrote:
Hello,
anybody has a clue what this
Correct, so the question is why ozmod_libpri attempting to move from
progress_media to progress ... may be a delayed libpri event? or some crap
along those lines.
On Wed, Aug 12, 2009 at 4:42 PM, Brian West br...@freeswitch.org wrote:
Isn't progress_media already past progress in the state
Yes you can get a progress after you get a progress with media ... I
have seen it.
/b
On Aug 12, 2009, at 3:57 PM, Moises Silva wrote:
Correct, so the question is why ozmod_libpri attempting to move from
progress_media to progress ... may be a delayed libpri event? or
some crap along
On Wed, Aug 12, 2009 at 2:25 PM, Brian West br...@freeswitch.org wrote:
They are all getting gathered up and put online...
files.freeswitch.org/cluecon_2009 just keep an eye there some of the
videos are up also.
/b
FYI,
I've uploaded the first batch and they should get synched up on
On Wed, Aug 12, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote:
Yes you can get a progress after you get a progress with media ... I
have seen it.
Yes, you definitely can and I believe that some of the PRI specs suggest
that this is totally legal, even though it's kind of silly.
-MC
That's the trouble with a 8am conference in a town where the bars
close at 4am :-)
On Wed, Aug 12, 2009 at 1:42 PM, Brian Westbr...@freeswitch.org wrote:
So you're the one that drank 16 gallons of coffee! Good luck sleeping!
/b
On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote:
Macbook
And it didn't help we had an open bar two of the nights!
/b
On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:
That's the trouble with a 8am conference in a town where the bars
close at 4am :-)
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it helped me!
oh... well, I helped myself!
-giovanni
On Wed, Aug 12, 2009 at 11:30 PM, Brian Westbr...@freeswitch.org wrote:
And it didn't help we had an open bar two of the nights!
/b
On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:
That's the trouble with a 8am conference in a town
On Wed, Aug 12, 2009 at 2:55 PM, Tina Martinez t...@a2unlimited.com wrote:
Hello,
I have spent the past couple of weeks toying around with FS to evaluate the
possibility of using it for a large scale conference server for our
organization. The plan is to have several FS servers initiate
You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls (unless you're
controlling the conference yourself from outside, using, say, the event
socket).
Before I go further, I want to make sure I understand what you're
Whoops. All of my parens () should be curly braces {}. Wasn't paying
attention.
BB
On Wed, Aug 12, 2009 at 2:40 PM, Bradley Brashier bjbrash...@gmail.comwrote:
You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls
then probably we should check the current state and ignore the libpri event
when already in progress with media.
On Wed, Aug 12, 2009 at 5:10 PM, Michael Collins m...@freeswitch.org wrote:
On Wed, Aug 12, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote:
Yes you can get a progress
Bradley Brashier wrote:
...
Before I go further, I want to make sure I understand what you're
proposing. What you're essentially saying is that when the command to
kick someone is pressed the person should be transferred out of the
conference, checked for moderator status, asked whom to
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the second
part of the command. Am I correct
Well you really can't ignore it... it happens with our ISDN stack
too. Thats what the VETO handles.
/b
On Aug 12, 2009, at 5:28 PM, Moises Silva wrote:
then probably we should check the current state and ignore the
libpri event when already in progress with media.
On Wed, Aug 12, 2009 at 6:10 PM, Tina Martinez t...@a2unlimited.com wrote:
Michael,
Thanks for the welcome, and for the response to my question.
The call control and dynamic setup of conferences I have working (pretty
cool stuff).
The tricky part, as you said, is linking the servers
Is there a way to limit the number of calls a UA can receive in the FS
configs?
I'm doing some testing with XLite as the UA, and can not figure out
how to keep line 2 from answering if line 1 is in use.
THanks.
-str
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Check out mod_limit... Other wise you'll have to look specifically at the UA
you are trying to use, some like polycom and sipura offer a way to disable
call waiting
Remember with SIP there is no such thing as a line, its a SESSION and you
can have as many sessions as the software allows (and most
I have been testing analog support under Windows with good results so far. I
am waiting on another driver fix to solve some small problems with the api.
I have been running this code for 6 weeks or so now under a home test
environment(light call traffic) and the reliabilty has been fine - no
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