Thanks Michael, fs_eslib sounds the one for Java. I'll give it a try.
On Thu, Jun 25, 2009 at 1:38 PM, Michael Collins wrote:
> FYI,
> Any language that can establish a network socket and send/receive
> information over that socket can be used to control FS. FreeSWITCH comes
> with ESL - the ev
FYI,
Any language that can establish a network socket and send/receive
information over that socket can be used to control FS. FreeSWITCH comes
with ESL - the event socket library - that can abstract away some of the
grunt work, but there isn't a Java one that I'm aware of.
-MC
On Wed, Jun 24, 200
Hi Chris, thanks for your help. Here's my client.xml
On Thu, Jun 25, 2009 at 11:53 AM, Chris Chen wrote:
> Please provide your client.xml detail w
Please provide your client.xml detail with confidential information
crossout, I have gtalk client and server working properly behind the NAT.
I should be able to help you.
Chris
On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang wrote:
> Thanks seven. External IPs have sound echo this time with ext-
Thanks seven. External IPs have sound echo this time with ext-rtp-ip
uncommented and disable-rtp-auto-adjust=true. However, internal IP has no
audio this time no matter what value disable-rtp-auto-adjust is...
On Thu, Jun 25, 2009 at 11:24 AM, seven wrote:
> uncomment ext-rtp-ip
>
> On Jun 25, 2
If then, what bridge i shall call to?
Like this?
dujinfang wrote:
>
> put your extension in dialplan/public.xml instead of sip_profiles/
> external/myprofile.xml
>
>
> On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote:
>
>>
>> Ooops.. Sorry wrong spelling... Same issue
>>
>> Jason White-14
Hi Paul, thanks for your reply. I've give it a try.
On Thu, Jun 25, 2009 at 11:13 AM, paul.d...@gmail.com
wrote:
> You can use FS socket event interface for that. See free Java lib for
> inbound socket event here: http://versafon.com/versafonweb/Software.jsp
>
> Jingwei Yang wrote:
> > Hi Folks,
uncomment ext-rtp-ip
On Jun 25, 2009, at 10:23 AM, Jingwei Yang wrote:
Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and
put "disable-rtp-auto-adjust" inside client.xml. No matter what
value this parameter has (true or false), local IP is able to hear
the echo but extern
put your extension in dialplan/public.xml instead of sip_profiles/
external/myprofile.xml
On Jun 24, 2009, at 6:02 PM, Edmar Cruz wrote:
>
> Ooops.. Sorry wrong spelling... Same issue
>
> Jason White-14 wrote:
>>
>> Edmar Cruz wrote:
>>
>>> Here is my dialplan on sip_profiles/external/myprofil
You can use FS socket event interface for that. See free Java lib for
inbound socket event here: http://versafon.com/versafonweb/Software.jsp
Jingwei Yang wrote:
> Hi Folks,
>
> I understand freeSwitch is supporting a couple of languages for call
> controls like Lua, Javascript, Perl, Java... Ho
Hi Folks,
I understand freeSwitch is supporting a couple of languages for call
controls like Lua, Javascript, Perl, Java... However, after digging into the
detailed wiki pages, I found out the codes written in those languages can
only be executed via the freeswitch console. I was wondering whether
This is on my freeswitch logs...
09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 sofia_glue_tech_set_codec() Set
Codec sofia/internal/1...@116.541.23.11 PCMU/8000 20 ms 160 samples
2009-06-25 10:21:50 [DEBUG] sofia_glue.c:2915 sofia_glue_negotiate_sdp() Set
2833 dtmf payload to 101
2009-06-25 10:21:
Hi seven, thanks for your reply. I've commented out "ext-rtp-ip" and put
"disable-rtp-auto-adjust" inside client.xml. No matter what value this
parameter has (true or false), local IP is able to hear the echo but
external ones still have no audio.
On Wed, Jun 24, 2009 at 6:01 PM, seven wrote:
>
I have a lua script that originates a call:
local s = freeswitch.Session (
"{ignore_early_media=true,origination_caller_id_name=" ..
caller .. "}loopback/" .. destination .. "/default/XML")
s:execute ("sleep", "1000")
...
which works fine if a valid number is supp
I have tried to reproduce this issue but haven't been able too... What
SVN Rev are you on?
/b
On Jun 24, 2009, at 10:29 AM, Richard Lamkin wrote:
I am using the API to manage calls as they arrive at FS from a trunk
I have a very simple Dial plan rule that parks the incoming call.
Turn on debugging and capture the output. Put it in a pastebin and post the
link here. We'll take a look.
-MC
On Wed, Jun 24, 2009 at 2:10 AM, Edmar Cruz wrote:
>
> when a type on the API of freeswitch originate
> sofia/external/8011...@116.541.23.12 1001, 8011104 my extension that I
> want
> to
can do the following:
1) "make current" or do a fresh checkout to make sure you build is clean.
2) try executing the app "ring_ready" with no args in place of respond 180
and see if it makes any difference.
3) clear out your logfile by stopping FS and deleting
/usr/local/freeswitch/log/freeswitch.
I have also observed that the cpu load goes up to 100% when only a
couple of orphaned calls are left without being cleared by "api hupall".
Richard Lamkin
richard.lam...@mettoni.com
From: Richard Lamkin
Sent: 24 June 2009 16:30
To: freeswitch-users@lists.freeswitch.org
Subject: [Fre
I was thinking, for the students machines your could run the command line
program linphonec in auto answer mode ( linphonec -a ). When a teacher calls
one of the thin clients, the student would automatically hear and be able to
speak to the teacher and not have to worry about dealing with a soft
On Tue, Jun 23, 2009 at 7:10 PM, Edmar Cruz wrote:
>
> Where can i find this logs?
>
Please look at this page because it will give you a lot of information about
how to collect information for debugging:
http://wiki.freeswitch.org/wiki/Reporting_Bugs
I recommend setting aside 20 minutes to rea
I am using the API to manage calls as they arrive at FS from a trunk
I have a very simple Dial plan rule that parks the incoming call.
Once the call is parked via the API I first send a ringing (to keep the
originator happy)
sendmsg
call-command: e
try adding
before the bridge and report back results.
Mike
On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote:
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
data="{absolute_codec_string='GSM,G729'}[nibble_rate=0.3]sofia/external/6626734...@203.xxx
Chances are you need to get var us-ring then use that to set the
transfer_ringback
/b
On Jun 24, 2009, at 9:20 AM, Diego Toro wrote:
Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring});
I have message: "[CRIT] switch_channel.c:633 Invalid data ($
{transfer_ringback} contain
Hi Brian, with Session.SetVariable("transfer_ringback", ${us-ring}); I have
message: "[CRIT] switch_channel.c:633 Invalid data (${transfer_ringback}
contains a variable)".
Using from managed code:
string stUsRing = _Session.GetVariable("us-ring");
Session.SetVariable("ringback", stUsRing);
S
Well its the same you use ${us-ring} in both cases.
/b
On Jun 24, 2009, at 8:07 AM, Diego Toro wrote:
Greetings
When I use Session.SetVariable("transfer_ringback", "us-ring") from
managed code the bridge fails with "NO_ANSWER" cause. If I use
from
xml dial plan the call is stablished.
Greetings
When I use Session.SetVariable("transfer_ringback", "us-ring") from managed
code the bridge fails with "NO_ANSWER" cause. If I use from xml dial plan the
call is stablished.
I have FS rev 13750 running on Windows.
This is a issue or I don't use properly transfer_ringback variable
Saeed Ahmed wrote:
> Can we also test dialplan using CLI, like "dial" in asterisk?
Have a look at the originate command.
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Can we also test dialplan using CLI, like "dial" in asterisk?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Cavalera
Claudio Luigi
Sent: Wednesday, June 24, 2009 12:36 PM
To: freeswitch-users@
> From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
> play with it from the cli
> freeswitch> eval ${foo:-4:4}
> API CALL [eval(${foo:-4:4})] output:
> 2345
Thanks anthm!
This way to test dir
search wiki from sth. like disable_rtp_autoajust , I don't remember
the exact var.
On Jun 24, 2009, at 5:40 PM, Jingwei Yang wrote:
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I
commented out "ext-rtp-ip" from client.xml, I'm able to hea
Ooops.. Sorry wrong spelling... Same issue
Jason White-14 wrote:
>
> Edmar Cruz wrote:
>
>> Here is my dialplan on sip_profiles/external/myprofile.xml
>>
>>
>>
>>
>>
> The above should be $1 not @1
>
>
> ___
> Freeswitc
Edmar Cruz wrote:
> Here is my dialplan on sip_profiles/external/myprofile.xml
>
>
>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Hi Guys,
Here's my situation:
The freeswitch server and my machine are behind the same LAN. If I commented
out "ext-rtp-ip" from client.xml, I'm able to hear the echo (by *originate
dingaling/gmail.com/user...@gmail.com &echo*).
However, external calls have no sound at all no matter whether this
when a type on the API of freeswitch originate
sofia/external/8011...@116.541.23.12 1001, 8011104 my extension that I want
to call and its ip is 116.541.23.12. I register on 1001 using a softphone
(X-Lite) and my ip is 116.541.23.11. It works actually.
but when dialing on softphone 1001 account o
Edmar Cruz wrote:
>
> Is there any available complete documentation for Freeswitch with matching
> samples aside from wiki that works. With working samples like dialplans,
> outbounds and prefer codecs etc.
There isn't much besides the wiki. Most of the documentation effort has been
devoted to
HI,
Is there any available complete documentation for Freeswitch with matching
samples aside from wiki that works. With working samples like dialplans,
outbounds and prefer codecs etc.
Thanks...
--
View this message in context:
http://www.nabble.com/Freeswitch-Documentation-tp24180754p241807
Mark Campbell-Smith wrote:
> How can I get more information on this fault to file a bug report?
See the debugging FreeSWITCH page on the wiki, and set
in the FreeSWITCH core configuration (by default in switch.conf.xml), or use a
ulimit -c unlimited command before running FreeSWITCH.
Next tim
Hi!
My call dropped and I saw this error in the syslog:
Jun 24 17:05:04 freeswitch kernel: [157531.309017] freeswitch[4621]:
segfault at c ip b73b2a42 sp b72a3840 error 4 in
mod_sofia.so[b7369000+16c000]
How can I get more information on this fault to file a bug report?
Thanks!
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