There is no way to make the dialplan continue to run when you hang up
the a leg, that is correct.
Mike
On Dec 26, 2008, at 3:20 PM, Frank @ Impact wrote:
> I also tried to add this
>
>
>
> to keep the dialplan process on a-leg hangup. But that did not work
> either.
>
> Svn 10960 is what I a
I also tried to add this
to keep the dialplan process on a-leg hangup. But that did not work
either.
Svn 10960 is what I am testing.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerr
This should work, is there any debug output at hangup that would
indicate why it doesn't run?
Mike
On Dec 26, 2008, at 2:57 PM, Frank @ Impact wrote:
> I also tried this without success. This will not fire at all
> regardless
> of who hangs up.
>
>
>
>
__
Frank,
I'm going to check this out as soon as I can get my test system back on
line. Thanks.
-MC
P.S. - what FS version and OS version are you on? I test with latest trunk
and CentOS 5.2
On Fri, Dec 26, 2008 at 11:57 AM, Frank @ Impact wrote:
> I also tried this without success. This will not
This is correct, if the a leg hangs up it will not continue to run the
dialplan actions.
On Dec 26, 2008, at 11:57 AM, Frank @ Impact wrote:
> I have confirmed that this system call does not fire if the calling
> party hangs up the phone first. Is there a way to get the script to
> fire regar
I also tried this without success. This will not fire at all regardless
of who hangs up.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Frank @ Impact
-Original Message-
From: freesw
Next time we'll need to open a jira for tracking. We have to get into
that habit soon ;)
/b
On Dec 26, 2008, at 11:38 AM, Adnan Barakat wrote:
> Brian West wrote:
>> Please update and try again.
>>
>> Committed revision 10949.
> Thanks Brian, works perfectly.
>
>
> Adnan
>
>
Brian West wrote:
> Please update and try again.
>
> Committed revision 10949.
Thanks Brian, works perfectly.
Adnan
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
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U
Please update and try again.
Committed revision 10949.
/b
On Dec 26, 2008, at 10:19 AM, Adnan Barakat wrote:
> Hi all,
>
> I've just updated from r1 to the latest trunk (as I needed
> mod_http), and group_confirm seems to have broken after the update.
> Now
> the first 2-5 seconds of the
I have confirmed that this system call does not fire if the calling
party hangs up the phone first. Is there a way to get the script to
fire regardless of who hangs up first?
-F
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.
All I am passing into the script is the recording file name.
I tried using the system command right after the bridge command but
before a hangup command. Thusly,
The problem I am seeing is that sometimes this script gets run and
sometimes it does not. I think it has to do maybe with which e
Hi all,
I've just updated from r1 to the latest trunk (as I needed
mod_http), and group_confirm seems to have broken after the update. Now
the first 2-5 seconds of the file is played very quickly at poor
quality, then the end of the file plays fine.
Here is the relevant part of the dialpla
I think what's wrong is that instead of listening to the explanation on the
other thread you already started on this issue, you started a new thread
trying to ask the same question a different way hoping for a different
answer. Once you learn how to configure FreeSWITCH you will be able to stop
us
If anything should be changed it's to add an additional
actual_register_domain header in the cases when it's being forced but it's
not completely necessary.
Typical example is when a client is using the ip address in the domain field
and you want to force it
to point at the domain name in your reg
Hi,
Thanks for the hint. I checked the log again and found out:
2008-12-27 06:01:49 [NOTICE] switch_loadable_module.c:259
switch_loadable_module_process() Adding API Function 'lua'
2008-12-27 06:01:49 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/us
You are indeed using stun. Check vars.xml if your using the default
configs. Also there is config params for the rtp port range,
Mike
On Dec 26, 2008, at 7:23 AM, Juan Backson wrote:
> Hi
>
> The strange thing is that I am not using stun at all. Also, this stun
> error only pops up after th
Your still ignoring early media. Are you trying to detect the tone
before or after answer?
Mike
On Dec 26, 2008, at 6:16 AM, Baskar wrote:
Hi Michael,
I have updated all the changes what u said, But still i did not
get any tone detect in the script
session1 = new Session();
sessio
It looks like mod_openmrcp isn't loaded.
On Dec 26, 2008, at 3:49 AM, mark morreny wrote:
Hi
I tried to setup mod_openmrcp according to wiki, but I am getting
the following error:
2008-12-27 00:40:23 [ERR] switch_core_speech.c:60
switch_core_speech_open() Invalid speech module [openmrcp
Hi
The strange thing is that I am not using stun at all. Also, this stun
error only pops up after the number of sessions reach a certain
amount.
JB
On Thu, Dec 25, 2008 at 2:11 AM, Anthony Minessale
wrote:
> dont load test against channels that must do a stun lookup.
> you are lucky you get 30
Hi,
Is there any hard limit set on the number of RTP sessions for
Freeswitch? I am seeing freeswitch start sending out BYE after the
number of RTP session reaches 3000. This problem happens even when
the machine utilization is still low.
Does anyone know what is wrong?
Thanks,
JB
___
*Hi Michael,
*
* I have updated all the changes what u said, But still i did not get
any tone detect in the script *
*session1 = new Session();
session1.originate(session1, "{ignore_early_media=True}sofia/default/
39841799...@172.20.191.228");
session1.execute("tone_detect", "test 400 r +30 ha
Hi
I tried to setup mod_openmrcp according to wiki, but I am getting the
following error:
2008-12-27 00:40:23 [ERR] switch_core_speech.c:60 switch_core_speech_open()
Invalid speech module [openmrcp]!
2008-12-27 00:40:23 [ERR] switch_ivr_play_say.c:1848 switch_ivr_speak_text()
Invalid TTS module!
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