Hi Mike,
I'm just trying to send it an event with some custom event headers, just so
an external program can communicate with a session without having to
transfer the session to a different program. I'm curious what uuid_display
does...the wiki only gives a brief description and my Google'ing
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
2009/10/12 Brian West br...@freeswitch.org
It was possible but we have a regression in the code that isn't
letting that happen right now... hence the reason i said Open a jira
so we could fix it.
IS THAT
FreeSWITCH 1.0.2?
That's more than a year old I think, you should really update to 1.0.4 or
latest SVN trunk.
Diego
On Mon, Oct 12, 2009 at 4:14 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
Hi Mike,
Thanks for your valuable reply,
when i install freeswitch1.0.2 in my
Any one please help me to solve the mentioned problem
On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu velu.techni...@gmail.comwrote:
Dear All,
I am using ESL.pm module to control the dial plan application. I want
to play some music while executing the some external scripts. I
Yes, I confirmed that with Wireshark (filter rtp and ip.src == device ip).
RTP packets are sent every 20ms.
MAniserowicz
- Original Message -
From: Michael Jerris (via Nabble)
To: Maciej Aniserowicz
Sent: Monday, October 12, 2009 12:21 AM
Subject: Re: [Freeswitch-users]
try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open them.
not sure how to do that on windows, but on linux:
# sqlite3 xx.db
sqlite select * from sip_registration;
2009/10/12 srinivasula reddy srinivas.ksvre...@gmail.com
Hi Mike,
Thanks for your valuable reply,
when i
Is it possible to keep a list of registered phone, and when FS will
start it will register them all automatically?
On Mon, 2009-10-12 at 13:19 +0200, Seven Du wrote:
try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open
them. not sure how to do that on windows, but on linux:
Ya ok. Here is the required stuff
Configuration: (PRI span)
configuration name=openzap.conf description=OpenZAP Configuration
settings
param name=debug value=7/
/settings
pri_spans
span id=1 name=PRI_1
param name=q921loglevel value=alert/
param name=q931loglevel value=alert/
param name=mode
Perfect! Thanks.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
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Did you open a jira and attach all the info?
/b
On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter rtp and ip.src ==
device ip). RTP packets are sent every 20ms.
MAniserowicz
___
On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TCHi Yuriy,
TC
TCcan you share what you have so far, I'm sure we can help with RTP part...
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems it
work, but should be buggy,
to build need libpt
it works in either case with or without media
the syntax for setting the frequency was answered above.
On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov shiya...@gmail.com wrote:
Michael, Diego,
thanks for the rapid answers!
As far as I understand, enable_heartbeat app is launching
2009/10/12 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TCHi Yuriy,
TC
TCcan you share what you have so far, I'm sure we can help with RTP
part...
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code,
Hi,
I have implemented the solution for tone detection in wiki, and also answer
the channel before detecting the tone:
condition field=destination_number expression=^(055138419992)$
action application=answer/
action application=tone_detect data=fax 1100 r +15000 transfer fax
XML default/
We can host this in our SVN if you wish?
/b
On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
seems it work, but should be buggy,
to build need libpt 2.6.5 and h323plus cvs version, i test it now on
fs 1.0.4.
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
On 2009-10-12 09:43 -0500, Brian West wrote freeswitch-us...@lists.freeswit...:
BWWe can host this in our SVN if you wish?
If in fs svn i think yes. But i think may be little time later?
i don't known is it builds on trunk because i develop it on 1.0.4.
BW/b
BW
BWOn Oct 12, 2009, at 8:31 AM,
which line is hanging up your A (inbound) leg?
look for a blue line that says Hangup xyz that matches it so i can
see.
I think what is happening is you are getting early media so the bridge is
actually working then when nobody answers it dies but technically the bridge
worked.
On Mon, Oct
extension name=1999!--DIRECT POWER--
condition field=destination_number expression=^1997$
action application=playback
data=hh/hh-unable_to_connect_contact.wav/
action application=park/
/condition
/extension
my extn 1999... since it looks from the output like
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
might be the line..or the entire output is below
freeswi...@matthew-laptop originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 15:21:44.029517 [NOTICE]
You are asking how you can do asynchronous actions park then play a sound
before park exits without enabling async mode. Think about that.
you could use new valet_parking that parks with music i guess
or tell the channel execute playback instead of park since playback an park
are almost the same
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
hi,
can't make it...
subZero:~/freeswitch-trunk$ make mod_h323
making all mod_h323
Compiling mod_h323.cpp...
quiet_libtool: compile: g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/home/tculjaga/freeswitch-trunk/src/include
Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
JIRA before regarding some other matter and it turned out to be my mistake,
so I decided to try mailing list first this time.
MA
Brian West wrote:
Did you open a jira and attach all the info?
/b
On Oct 12,
Fixed... svn up.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
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Hi, all
I used sipp to test to forward an un-implemented SIP method, but I got 501
Not Implemented.
Can freeswitch be a SIP proxy server?
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.10.59.161:5061
From: sip:1...@10.10.59.69:5060
To: sip:1...@10.10.59.69:5060;tag=gBpay50SD2SUD
Call-ID:
On Sun, Oct 11, 2009 at 2:40 PM, Diego Viola diego.vi...@gmail.com wrote:
I'd like to add this for the next weekly conference.
I have added a few events to the event list, as you can see here:
http://wiki.freeswitch.org/wiki/Event_list
But I need more help from the community to complete
This is because FS is a B2BUA ... not a proxy. You should consider
OpenSER/SIPS/Kaemillio for this type of application.
SDR
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On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
i already write this, ptlib 2.6.5, tou can find link to it on oplalvoip.org,
h32plus latest CVS version,
you can find it on www.h323plus.org.
TChi,
TC
TCcan't make it...
TC
TCsubZero:~/freeswitch-trunk$ make
On Sun, Oct 11, 2009 at 4:49 PM, Brian West br...@freeswitch.org wrote:
I have tried to police the wiki when things like this appear.. its one
thing to crack a joke in fun from time to time... but to put stuff
like that on the wiki isn't acceptable.
/b
It seemed appropriate to do so,
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)?
revised dialplan cmd output
extension
On Mon, Oct 12, 2009 at 4:01 AM, homqua ngay01042...@gmail.com wrote:
Hi,
I have implemented the solution for tone detection in wiki, and also answer
the channel before detecting the tone:
condition field=destination_number expression=^(055138419992)$
action application=answer/
On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong mattdf...@gmail.com wrote:
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
still no luck...
extension name=1920!--DEBUG--
condition field=destination_number expression=^1920$
action application=set data=hangup_after_bridge=false/
action application=bridge
data={leg_timeout=10,ignore_early_media=true}sofia/gateway/
debug.com/14159927717/
On Sun, Oct 11, 2009 at 10:48 AM, EdPimentl edpime...@gmail.com wrote:
http://www.assistivetech.net/search/productDisplay.php?product_id=18854
-E
Let's keep an eye on this but I doubt that an assistive technology
application is going to create any real confusion.
-MC
Turn on debug, make another test call, and pastebin the output.
-MC
On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins m...@freeswitch.orgwrote:
On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong mattdf...@gmail.comwrote:
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
On Tue, Oct 6, 2009 at 8:41 AM, Lars Zeb larc...@yahoo.com wrote:
http://pastebin.freeswitch.org/10612
I having been running v14996 OK for a while. I have upgraded a couple of
times after, but every time, an inbound call is hung up on. The only thing
that has changed is the upgrade. This
Lak,
Okay I will need a little bit of time to dig into the IE's and what they
contain. In the meantime can you tell me who the carrier is? I'd like to
find out if they have some specific requirements. The fact that it doesn't
work with libpri surprises me because that would mean that Asterisk
http://pastebin.freeswitch.org/10656
On Tue, Oct 13, 2009 at 1:34 AM, Michael Collins m...@freeswitch.org wrote:
Turn on debug, make another test call, and pastebin the output.
-MC
On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins m...@freeswitch.orgwrote:
On Mon, Oct 12, 2009 at 10:42
Michael Collins m...@freeswitch.org said:
On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
mayamatake...@gmail.comwrote:
On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
m...@freeswitch.orgwrote:
FYI,
The FreeSWITCH devs have added valet parking! Check it out:
I would say that the parking meter is a good idea and it is the default
behavior of parking on legacy PBXs. Since we always do _more_, what do you
think about having the option to transfer to any extension instead of just
the one that transfered the call?
Regards,
jm
On Mon, Oct 12, 2009 at
I don't know if this was mentioned yet. It would be useful to have a way
to have the parking lot automatically find the next available spot and
tts it to the person parking the call.
Then the auto unpark would pop off the lowest numbered lot, or return
fail if there is nobody in the parking lots
Hi,
Does Freeswitch detect all of these hangup cases mentioned here [
http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
Trunk provider?
If not, should I put in tone_detect application in the dialplan for
detecting the SITs?
Won't freeswitch have to depend on the SIP
On Mon, Oct 12, 2009 at 12:55 PM, William King quentus...@gmail.com wrote:
I don't know if this was mentioned yet. It would be useful to have a way
to have the parking lot automatically find the next available spot and
tts it to the person parking the call.
Then the auto unpark would pop off
2009/10/12 João Mesquita jmesqu...@freeswitch.org
I would say that the parking meter is a good idea and it is the default
behavior of parking on legacy PBXs. Since we always do _more_, what do you
think about having the option to transfer to any extension instead of just
the one that
On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur
vinuth.madi...@gmail.comwrote:
Hi,
Does Freeswitch detect all of these hangup cases mentioned here [
http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
Trunk provider?
If not, should I put in tone_detect application in
Thanks Michael. I'll go through the resources you mentioned.
Thanks,
Vinuth.
On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur vinuth.madi...@gmail.com
wrote:
Hi,
Does Freeswitch detect all of these hangup cases
On Mon, Oct 12, 2009 at 12:00 PM, Matthew Fong mattdf...@gmail.com wrote:
http://pastebin.freeswitch.org/10656
Matthew,
Try continue_on_fail=true instead of hangup_after_bridge=false.
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
I think it will do what you want.
-MC
Hello,
I have been doing some work recently on the FreeSWITCH wiki, to improve
things.
You can see some of my work here:
http://wiki.freeswitch.org/index.php?title=Special:Contributionslimit=500target=Diego.viola
I am trying to polish the wiki and give it a more professional and clean
look,
if you are using ringback variable you now must also use
ignore_early_media=true either exported from A leg or in {} on the b leg
dial string to get the original behavior.
On Tue, Oct 6, 2009 at 10:41 AM, Lars Zeb larc...@yahoo.com wrote:
http://pastebin.freeswitch.org/10612
I having
On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins m...@freeswitch.org wrote:
It seemed appropriate to do so, therefore I added a small snippet on the
documentation guidelines:
http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
Good guidelines...
Just in time to
On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola diego.vi...@gmail.com wrote:
Hello,
I have been doing some work recently on the FreeSWITCH wiki, to improve
things.
You can see some of my work here:
http://wiki.freeswitch.org/index.php?title=Special:Contributionslimit=500target=Diego.viola
On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson g...@gundy.org wrote:
On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins m...@freeswitch.org
wrote:
It seemed appropriate to do so, therefore I added a small snippet on the
documentation guidelines:
Hello
I would like to watch these files. Can someone seed please. Thank you.
--FC
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Opps. didn't realize it was closed. Here you go. Moving it to my seedbox
for permanent seeding.
-William King
Frank Carmickle wrote:
Hello
I would like to watch these files. Can someone seed please. Thank you.
--FC
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Hello,
I am trying to use luasql with freeswitch but having a hard time compile it
with x86_64 Server running CentOS 5.3
Linux 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64
GNU/Linux
[r...@fsx1 luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET=10.3; gcc -O2 -Wall
DJB wrote:
Hello,
I am trying to use luasql with freeswitch but having a hard time compile it
with x86_64 Server running CentOS 5.3
Linux 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64
x86_64 GNU/Linux
[r...@fsx1 luasql-2.1.1]# make
export
Hi,
I'm facing an issue and I don't know why this happens or what I can do to solve
this.
Here's the scenario:
- FS call timeout set to 30 sec
- Setting continue_on_fail=true and hangup_after_bridge=false
- 1 Snom phone and an external SIP/ISDN gateway (Lancom) connected to FS
established a
On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins m...@freeswitch.org wrote:
On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson g...@gundy.org wrote:
On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins m...@freeswitch.org
wrote:
It seemed appropriate to do so, therefore I added a small snippet
If anyone ran into the same problem, here is the fix that I got from [stkn] on
IRC:
make CFLAGS=-O2 -fPIC
I hope it might help someone in the future.
Regards.
- Original Message
From: DJB djbin...@yahoo.com
To: FREESWITCH-USERS MAILING LIST freeswitch-users@lists.freeswitch.org
This is what I have in my dialplan and the fax is detected
beautifully. Note that in my case, extension 1000 will ring for a
second or two before the fax is detected. So in your example, the fax
does not have time to be detected, the dialplan exists and the call is
hungup.
When the fax is
Sure, I'm happy to put my little two cents to help the project :).
Diego
On Mon, Oct 12, 2009 at 10:38 PM, Michael Collins m...@freeswitch.orgwrote:
On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola diego.vi...@gmail.comwrote:
Hello,
I have been doing some work recently on the FreeSWITCH wiki,
I won't try until I need that, but I believe it works. Thanks Brian.
2009/10/13 Brian West br...@freeswitch.org
Fixed... svn up.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
It was a problem and has been fixed in the last trunk. Just update to the
latest code should be ok.
btw, the developers using jira to track bugs, so feel free to report one (as
you see http://jira.freeswitch.org/browse/FSCORE-463) if you think it's a
bug next time.
2009/10/12 Nagalenoj
exactly that is what i am trying to do now, if any help that would be great.
On Mon, Oct 12, 2009 at 5:07 PM, Eli Hayun eli...@savion.huji.ac.il wrote:
Is it possible to keep a list of registered phone, and when FS will start
it will register them all automatically?
On Mon, 2009-10-12 at
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