Dear All,
I have played the list of voice files in playback like the following by
using ESL perl module,
$conn-execute(set,playback_delimiter=!);
$conn-execute(set,playback_sleep_val=100);
Hello,
in this page
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
we can read:
=
Variables:
Any variables defined in the domain or user will be defined as channel
variables when there is a call to user or when there is an inbound calls
from that user.
Type conference at the cli and the help will be displayed... same
can be called over XML RPC.
/b
On Oct 27, 2009, at 11:51 PM, Lon Baker wrote:
Just like you can call conference list for conferences, is there a
way to retrieve the profile state of a conference member using the
cli or
http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
Are you using sync or async socket?
/b
On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:
Dear All,
I have played the list of voice files in playback like the
following by using ESL perl module,
Yes it is the expected behavior... if you wish to set them on an
inbound call TO the user you'll use the set_user api to do so. Its
kinda like sudo in FreeSWITCH it'll load up all the variables for that
user.
/b
On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:
I can see the channel
Hi,
can i use sqlserver instead of sqllite. and can two freeswitch servers can
share same database(sqllite or sqlserver). any help would be great.
Thanks
Srinivasula Reddy K
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Got it. Thanks.
On Wed, Oct 28, 2009 at 5:57 PM, Brian West br...@freeswitch.org wrote:
Yes it is the expected behavior... if you wish to set them on an
inbound call TO the user you'll use the set_user api to do so. Its
kinda like sudo in FreeSWITCH it'll load up all the variables for that
Handling of fastStart in CallProceeding is commented out in h323plus
library,
this is exploration from h323plus developers about this:
Yes that should be mera.
The problem is that Callproceeding does not always come from the remote it
may be generated by the gatekeeper.
this is a
I am using async socket. Shall I set the regular expression for
palyback_terminators?
On Wed, Oct 28, 2009 at 2:26 PM, Brian West br...@freeswitch.org wrote:
http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
Are you using sync or async socket?
/b
On Oct 28, 2009, at
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
Please don't cross post if possible.
/b
On Oct 28, 2009, at 4:07 AM, srinivasula reddy wrote:
Hi,
can i use sqlserver instead of sqllite. and can two freeswitch
servers can share same database(sqllite or sqlserver). any help
would
Hello Velusamy!
You may also be interested in using the method:
http://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits
It play a file, gets DTMFs and can be terminated if the maximum number of
digits is reached (i.e. 1 digit).
Delian Tashev
-Original Message-
From: Brian West
What kind of router are you behind?
/b
On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:
Thanks for the reply, Brian.
Did something in FS change between v15183 and v15225 to make this
occur? I ask because this same configuration worked OK in the
earlier version.
Lars
Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578
Ivan
Den 27. okt. 2009 kl. 18:34 skrev Eliot Gable:
No, the IP address the media originates from does not need to be tied
to the SIP IP address. Can
Correct - compiled but did not run. Works fine now.
I'll see if the error shows up again and let you know if it does.
Thanks,
MA
Anthony Minessale wrote:
won't compile or won't run?
maybe you should try rebuilding it.
On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz
I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets from
eyebeam's nat address (confirmed with tcpdump and mod_event_socket DTMF) but
it doesn't change the RTP destination to the source of those packets and
sends the packets to the
are you = r15256 because an issue with this was fixed in that revision.
On Wed, Oct 28, 2009 at 11:35 AM, mayamatakeshi mayamatake...@gmail.comwrote:
I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets from
eyebeam's nat
have you updated to the latest SVN?
/b
On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote:
I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets
from eyebeam's nat address (confirmed with tcpdump and
mod_event_socket DTMF)
Thank you very much, Anthony. I really appreciate your help and excellent work
of you and your team.
Regards,
Dorn B.
From: Anthony Minessale anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Mon, October 26, 2009 12:59:01 PM
Hello FreeSWITCHers,
The latest FreeSWITCH version is now available for download on the
fileshttp://files.freeswitch.org/freeswitch-1.0.5pre3.tar.gzsite.
The announcement story is in the main
FreeSWITCH http://www.freeswitch.org/node/212 site. Please download and
test, and then test some more. We
I'm really sorry, guys.
I was just some revisions behind and forgot to make current and test before
posting.
It's fine now.
Thank you.
On Thu, Oct 29, 2009 at 1:51 AM, Brian West br...@freeswitch.org wrote:
have you updated to the latest SVN?
/b
On Oct 28, 2009, at 11:35 AM, mayamatakeshi
Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on
fedora 8 VM. Any clue? thanks,
[CRIT] switch_loadable_module.c:871 Error Loading module
/usr/local/freeswitch/mod/mod_spidermonkey.so
**/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**
Per http://jira.freeswitch.org/browse/MDXMLINT-23
Try to configure it with --without-libcurl.
-- W
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On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org wrote:
Here is a debug log from a call from an internal phone out to an
external (my iPhone with nbr 91316356):
http://pastebin.freeswitch.org/108578
Ivan
Uh... you wanna try that PB number again?
-MC
Okay, obligatory questions:
Which version of FreeSWITCH are you running?
Which PRI library are you using?
Which BRI card are you using?
-MC
On Wed, Oct 28, 2009 at 9:46 AM, Jakub Pawliński ja...@pawlinski.pl wrote:
Hi,
I have some problems with bri status. I have 3 chanel isdn modem, and
On Mon, Oct 26, 2009 at 12:20 AM, velusamy velu velu.techni...@gmail.comwrote:
Dear All,
What is the value of created_time channel variable? Is this epoch
seconds?
It's epoch microseconds. Divide by 1,000,000 to get epoch seconds...
-MC
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BroadXent ADSL 8120 è Netscreen 5XP
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, October 28, 2009 7:13 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Problem
The libcurl is broken on your distro. You can fix this by configuring
with --without-libcurl which will use our working in tree copy instead
of the broken one from your distro.
Mike
On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote:
Hi, I got an error in loading mod_spidermonkey. my fs in
Hello!
I'm researching a phone system for an organization with 8-12 members.
The place has an Internet connection via Verizon FIOS. My first choice
of the operating system is FreeBSD (preferably -- on amd64 platform).
FreeBSD's port (net/freeswitch) seems nice and currently installs
version
Hello, it's my first post, and I'm quite new to Freeswitch...
I'm looking for a bit of direction / advice on Multi-tenant set-ups. I
have configured Freeswitch as per the fine Multi-tenant guide on the
Wiki, but this leaves some aspects such as transfers and IVR still in
the default context.
For
Hi,
I have configured FreeSwitch and played it by making calls. Every thing
went fine. Then I tried to see what happens if network cable is
unplugged. Freeswitch shown an error saying ping failed to the gateway
and IP address is changed to local system IP. After sometime network
cable plugged
I notice that when I call IVR from the PSTN, the Welcome to Freeswitch...
introduction is clipped at the beginning, so it sounds like come to
Freeswitch. If I call 5000 internally, then I always hear the full
introduction. What can I do to resolve this?
My XML config looks like:
extension
Sleep 1000 ms... we usually bring up media too fast before the other
end is ready.
/b
On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
I notice that when I call IVR from the PSTN, the Welcome to
Freeswitch...
introduction is clipped at the beginning, so it sounds like come to
Oh, what happened to it?
Anyway, here is a new pb:
http://pastebin.freeswitch.org/10867
Ivan
Den 28. okt. 2009 kl. 19:12 skrev Michael Collins:
On Wed, Oct 28, 2009 at 7:37 AM, Ivan C Myrvold i...@myrvold.org
wrote:
Here is a debug log from a call from an internal phone out to an
Hi
I'm also working on this project, so i can answer your questions
Which version of FreeSWITCH are you running?
FreeSWITCH Version 1.0.trunk (15246)
Which PRI library are you using?
openzap Native stack
openzap.conf
[span zt BRI1]
trunk_type = bri
b-channel = 1-2
d-channel= 3
openzap.conf
I modified my dialplan as shown, but the clipping persists. Should the
sleep be placed somewhere else?
extension name=ivr_demo
condition field=destination_number expression=5000
action application=sleep data=1000\
action application=answer/
action application=start_dtmf/
Hello
Yes. Answer first.
--FC
On Wed, Oct 28, Jerry Richards wrote:
I modified my dialplan as shown, but the clipping persists. Should the
sleep be placed somewhere else?
extension name=ivr_demo
condition field=destination_number expression=5000
action application=sleep
Jerry,
Put the sleep after the answer. That should fix it.
Shelby
Jerry Richards wrote:
I modified my dialplan as shown, but the clipping persists. Should the
sleep be placed somewhere else?
extension name=ivr_demo
condition field=destination_number expression=5000
action
That's really nice looking software. Thanks for the pointer.
Cliff
On Tue, 2009-10-27 at 16:06 -0400, Gregory Boehnlein wrote:
I'm fond of Vqmanager from ManagEngine. It is a passive SIP monitor. I.E.
you mirror the ports that your FS or Asterisk boxes, and VQmanager sniffs
the mirrors,
Hi Shelby,
Thanks! That's pretty useful. I also note that this same info is
available from the CLI, although I was curious as to what some of the
numbers indicated:
UP 0 years, 13 days, 19 hours, 25 minutes, 48 seconds, 887 milliseconds, 975
microseconds
529509 session(s) since startup
26
sessions per second now/max
/b
On Oct 28, 2009, at 4:58 PM, Cliff Wells wrote:
current sessions, but I'm unsure of what the 0/30 means.
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Thanks. Can you collect debug logs of this happening? See
http://wiki.freeswitch.org/wiki/Reporting_Bugs for helpful tips on
collecting debug information. Use pastebin to dump all the log info and
reply here with the link. We don't have too many BRI users but I believe
there are a few so hopefully
On Wed, Oct 28, 2009 at 11:17 AM, Martin Hickman ma...@makolink.com wrote:
Hello, it's my first post, and I'm quite new to Freeswitch...
I'm looking for a bit of direction / advice on Multi-tenant set-ups. I
have configured Freeswitch as per the fine Multi-tenant guide on the
Wiki, but this
On 2009-10-06 22:47 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
Hi, now i have another questions about fs and not sip protocols,
call scheme like this - user1-sip-fs-sip-user2,
as i see if in sip profile enable inbound-proxy-media SDP is goes
end-to-end if sip is on both call
Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it
be the case of just follow the trunk like it is with any problems you
encounter in 1.0.4?
2009/10/29 Michael Collins m...@freeswitch.org
Hello FreeSWITCHers,
The latest FreeSWITCH version is now available for download
Craig Askings cr...@overthewire.com.au wrote:
Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it
be the case of just follow the trunk like it is with any problems you
encounter in 1.0.4?
I can't speak for the developers, but I would expect the current policy to
Hi to all,
this is my first message here!
I'm trying to setup a freeswitch box using 1.0.4 version compiled from
sources, ubuntu 8.04.03 lts with lastest updates and lastest zaptel
modules/tools compiled from sources. As hardware i'm using an OpenVox
A800P.
My problem is the following: if
fixed. Thanks,
On Wed, Oct 28, 2009 at 11:51 AM, Michael Jerris m...@jerris.com wrote:
The libcurl is broken on your distro. You can fix this by configuring
with --without-libcurl which will use our working in tree copy instead
of the broken one from your distro.
Mike
On Oct 28, 2009, at
Hi, new to javascript. I tried to pass two arguments into javascript,
action application=javascript data=test.js $1 $2/
In test.js, I tried to use argv[1] to retrieve $1 and argv[2] to retrieve
$2, however, the javascript test.js complained about argv[] as undefined
variables. How to retrieve
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send the ACK, you will
probably have your answer to why the other system did not receive it.
If you're still not
Should work fine. I use this:
var calling_num = argv[0];
var called_num = argv[1];
Are you sure you actually had valid data in $1 and $2? Try to call it
from the CLI:
jsrun test.js testvar1 testvar2
On Wed, Oct 28, 2009 at 10:22 PM, Erwin Davis davis.er...@gmail.com wrote:
Hi, new to
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