have you created Extension 1002?
-nandy
On Wed, Dec 9, 2009 at 3:20 AM, mailinglist mailingl...@fribert.dk wrote:
Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I
don't know how to fix it properly.
Looking in the Dialplan directory I see the following:
just change the dialplan/default.xml as mentioned by brian but i think you
can't use # as the first key 'cuz it normally used as a Send key. you may
change # to * (star key).
On Wed, Nov 11, 2009 at 12:06 AM, Ognjen Seslija osesl...@gmail.com wrote:
Add the following:
action application=set
hi simon, implementation is almost the same. here's my dialplan:
extension name=say_destination_info
condition field=destination_number expression=^0044(\d)$
action application=answer/
action application=playback data=misc/dialing.wav/
action application=say data=en name_spelled
i come across the valet_park application when i just finished an
improved-version of the call parking (using mod_fifo) such that it is
parked to different ext'n numbers when the caller is att_xfer'd to ext 777.
i used strftime(%s) to generate 700~759 parking numbers. i also added
feature that if
agree that WAV/PCMA/PCMU formats are best for performance. you can use
mp3/ogg ONLY to archive recorded files.
/nandy
On Sun, Oct 4, 2009 at 7:38 AM, Tihomir Culjaga tculj...@gmail.com wrote:
also, you can store files in PCMA/PCMU format and avoid transcoding at
all... and as said disk space
hi mike, i download the tarball file to check the configure script. it's
clean. so, there must be an error during my first download or build. - nandy
On Wed, Sep 16, 2009 at 3:54 PM, Michael Jerris m...@jerris.com wrote:
Are those in the Tarball?
On Sep 15, 2009, at 11:47 PM, Nandy Dagondon
this makes sense. a workaround would be to provide an optional variable to
delete recording file if it's less than N seconds. otherwise, it defaults to
a preset duration.
/nandy
On Thu, Sep 17, 2009 at 7:46 AM, Seven Du dujinf...@gmail.com wrote:
I think the file was there but deleted by
hi folks, anyone encountered this problem? tks.
/nandy
On Mon, Sep 14, 2009 at 2:20 PM, Nandy Dagondon nandy1...@gmail.com wrote:
meftah,
i disabled mod_erlang_event in modules.conf. unixodbc is installed already.
still ... the same error message. tks for your input.
/nandy
On Sun, Sep
the ./configure script aborts after the last error message. any hint where
to look for the problem? tks.
/nandy
On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson and...@hijacked.uswrote:
On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
hi folks, anyone encountered
is the Erlang source needed in the FS source directory?
/nandy
On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon nandy1...@gmail.comwrote:
the ./configure script aborts after the last error message. any hint where
to look for the problem? tks.
/nandy
On Wed, Sep 16, 2009 at 10:53 AM
it's working now. the problem? it's the configure script itself. some ^M
characters somehow crept into the line containing ac_config_files. tks for
the tip Andrew!
/nandy
On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon nandy1...@gmail.comwrote:
is the Erlang source needed in the FS source
mike, got it from tarball. - nandy
On Wed, Sep 16, 2009 at 11:51 AM, Michael Jerris m...@jerris.com wrote:
something is messed up in your build environment, it has nothing to do with
erlang. Is this with a fresh svn checkout or tarball?
Mike
On Sep 13, 2009, at 10:27 AM, Nandy Dagondon
install unixodbc if is not installed
thanks
Nandy Dagondon a écrit :
hi,
i want to enable odbc support which is required in mod_lcr feature.
however, i encounter ./configure problem after installing Erlang R13B01.
this is the portion of the error messages:
...
checking for erl... /usr
hi,
i want to enable odbc support which is required in mod_lcr feature. however,
i encounter ./configure problem after installing Erlang R13B01. this is the
portion of the error messages:
...
checking for erl... /usr/local/bin/erl
checking erlang version... 5.7.2
checking erlang libdir...
hi brian,
for outside clients to register w/ the internal profile, the router has to
forward port 5060 to FS. am i correct?
/nandy
On Wed, Sep 9, 2009 at 10:28 PM, Brian West br...@freeswitch.org wrote:
Those configs will still work.
/b
On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote:
Hi
sample configs
in the distribution.
tks for the clarification,
/nandy
On Thu, Sep 10, 2009 at 5:52 PM, Jason White ja...@jasonjgw.net wrote:
Nandy Dagondon nandy1...@gmail.com wrote:
for outside clients to register w/ the internal profile, the router has
to
forward port 5060 to FS. am i
nik,
please try the legs variable
http://www.nabble.com/CDR-accounting-question-td19212516.html
/nandy
On Sun, Sep 6, 2009 at 6:40 PM, Nikolai Geordzhev n.geordz...@gmail.comwrote:
Hi,
I`m trying to implement Call Forwarding in my FS setup. I set a user
variable managing the type of
if there is a way for FS to force the codec order on Leg
A with some knowledge of the preferred codec on Leg B, ie I know that
Leg B will always use G711 so that I want to biase the SDP answer on Leg
A based on this fact.
regards,
rod
Nandy Dagondon a écrit :
rod,
have you tried this?
http
rod,
have you tried this?
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html
/nandy
On Thu, Sep 3, 2009 at 2:50 PM, rod kawa...@laposte.net wrote:
Hi Michael,
I did some tests but I haven't been successful, so there is what I'm
trying to achieve:
On A leg, my
no need to modify anything on the directory entries - unless you add more
extension numbers. FS has a uPNP feature already. if your router has uPNP,
the external phone only needs the ff:
1. public IP of the FS (or hostname)
2. username
3. password
Items 2 3 are found in the directory entries.
you can create your tones.conf using call progress tones found at
http://www.3amsystems.com/wireline/tone-search.htm
On Fri, Aug 7, 2009 at 7:17 PM, Merul Patel me...@mac.com wrote:
Where can I find a sample tones.conf file for the UK? Am trying to
configure a USBFXO device for outbound
r 13612 is after 1.0.3. you better get 1.0.4 recently released.
-nandy
On Fri, Aug 7, 2009 at 8:19 PM, a...@chandlerfamily.org.uk wrote:
I sent my first e-mail to the list this morning (about 4 hours ago) but it
does not seem to have arrived back, even though I have received other,
later
in my implementation, i would use 2 separate conditions that looks like
this:
condition field=destination_number data=(51\d*)\
action application=bridge data=sofia/sip/$...@222.333.444.555/
condition field=destination_number data=(63\d*)\
action application=bridge
ed,
i mean you use separate extension names:
extension name=prefix-51
condition field=destination_number data=(51\d*)\
action application=bridge data=sofia/sip1/$...@222.333.444.555/
/extension
extension name=prefix-63
condition field=destination_number data=(63\d*)\
action
ok. w/ my apologies. - nandy
On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice kr...@suspicious.org wrote:
No need to bump these things as this is a mailing list and it annoys
quite a few people when you do that
--
*From: *Nandy Dagondon g...@i.ph
*Reply
we have a forum on compact,fanless last may.
On Sun, Jul 5, 2009 at 6:21 AM, William Suffill
william.suff...@gmail.comwrote:
Ya I have a SheevaPlug but yet to have anything interesting to report
about making it do anything. The potential is there tho.
-- W
just bumping this topic.
-nandy
On Fri, May 8, 2009 at 12:44 AM, Fred-145 codecompl...@free.fr wrote:
Antonio Gallo wrote:
Alix cases are like 6/9 € from their shop site. I think its easy to find
someone who work with aluminium that can make for you custom boxes for
like like 6/20 € at
fs can support lots of codecs. you can find the ff variables defined in
vars.xml:
global_codec_prefs
outbound_codec_prefs
then look for inbound_codec_negotiation in
sip_profiles/internal.xml,sip_profiles/external.xml if you want your
codec_prefs to set priority or not.
-nandy
On Wed, Jul 1,
seven,
of course, codec negotiation depends on the order of codecs in the
*_codec_prefs variables. but, the opposite end has also it's own codecs
prefs, too. fs can accept the other end's prefs
(inbound_codec_negotiation=generous) or imposes it's own prefs (=greedy).
you must include the codec in
you FS doesn't accept PCMU. try to add PCMU on both variables.
On Wed, Jul 1, 2009 at 3:44 PM, qian ma god.nirv...@gmail.com wrote:
thanks for your replies.
my var.xml:
X-PRE-PROCESS cmd=set data=global_codec_prefs=PCMA/
X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMA,GSM,iLBC/
below
in global_codec_prefs and outbound_codec_prefs, it doesn't
work. FS only accept PCMU.
why??
2009/7/1 Nandy Dagondon g...@i.ph
you FS doesn't accept PCMU. try to add PCMU on both variables.
On Wed, Jul 1, 2009 at 3:44 PM, qian ma god.nirv...@gmail.com wrote:
thanks for your replies.
my var.xml
sorry. i mean check the x-lite client if PCMA is enabled?
On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon g...@i.ph wrote:
check the value of inbound_codec_negotiation in the sip_profiles/*.xml
files. is it generous or greedy? you should also check if the endpoint
is offering PCMU
=PCMA/
2009/7/1 Nandy Dagondon g...@i.ph
sorry. i mean check the x-lite client if PCMA is enabled?
On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon g...@i.ph wrote:
check the value of inbound_codec_negotiation in the sip_profiles/*.xml
files. is it generous or greedy? you should also check
you can combine the 2 gateways into one bridge app. pls see
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
/nandy
On Fri, Jun 26, 2009 at 1:48 PM, Edmar Cruz darklio...@yahoo.com wrote:
include
extension name=multiple
condition field=destination_number
...@jasonjgw.net wrote:
Nandy Dagondon g...@i.ph wrote:
hi,
i tested the latest SVN build (13884) using the sample configuration
files
... no modifications whatsoever. but in sofia external profile, the IP
address is my internal address instead of my external IP address.
did i miss
DNS lookup?
jmesquita
On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon g...@i.ph wrote:
the default setting is auto-nat.
i changed ext-sip-ip=$${external_sip_ip} and
ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun:
stun.freeswitch.org. result: same problem
i tried
profile. it should display the external IP address instead.
On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon g...@i.ph wrote:
the default setting is auto-nat.
i changed ext-sip-ip=$${external_sip_ip} and
ext-rtp-ip=$${external_rtp_ip}. both of them are set in vars.xml as stun
hi everybody,
i'm interested to know if anyone employed FS as a local exchange switch. i'm
confident FS can handle several calls using RTP by-pass mode. however, i'm
more concerned on handling the large dialplan with hundreds (or even a few
thousand) exchange prefixes nationwide during call
community is really great!
tks once again,
nandy
On Sun, Jun 21, 2009 at 11:50 AM, David Knell d...@3c.co.uk wrote:
Hi Nandy.
On Sun, 2009-06-21 at 08:58 +0800, Nandy Dagondon wrote:
i'm interested to know if anyone employed FS as a local exchange
switch. i'm confident FS can handle several
hi,
i tested the latest SVN build (13884) using the sample configuration files
... no modifications whatsoever. but in sofia external profile, the IP
address is my internal address instead of my external IP address.
did i miss something here? tks.
-nandy
what is PCCW? could you please fill in more details what you like to do. to
connect mobile phones w/ FS, the mobile phone has to have SIP feature. pls
search the Wiki for some models.
-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
i'm using version build 13245M on an Intel D945GCLF2 Atom Dual-core mobo w/
2GB ram.
-nandy
On Sat, Jun 6, 2009 at 11:02 AM, Brian West br...@freeswitch.org wrote:
You shouldn't be having problems... what version are you using?
/b
On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote:
there 10
we experience some latency in the recording files even with PCMU-PCMU
session to a stereo WAV file. i want to reduce the CPU load hoping to reduce
this problem. would it help if do the ff?
1. save it in PCMU file. i can use sox at the end of the shift.
2. record in mono. does it help?
3. will
what are the error messages in the FS CLI output?
On Thu, Jun 4, 2009 at 3:10 AM, Matthew Lockwood matthew.lockw...@gmail.com
wrote:
Okay, I did that. I first ran it using X-Lite (which does connect on the
second try; the first one always times out) - there was lots of output. When
I tried
hi to all,
i'm looking for a default action in the IVR in case the caller doesn't press
any key. is this option available? with this option, we can add this prompt
... please stay on the line to be connected.. i know this can be done
using scripts but it's better to have this feature on the app
at 10:05 AM, Brian West br...@freeswitch.org wrote:
Set the app after ivr to transfer and set the exit sound to please stay on
the line to be connected/b
On May 28, 2009, at 8:58 PM, Nandy Dagondon wrote:
hi to all,
i'm looking for a default action in the IVR in case the caller doesn't
press
IMHO, you have tons of features w/ FS. i've setup FS on a low-power
consumption Intel D945GCLF2 motherboard (Atom dual-core CPU) ideal for 24/7
operation on a 10-seat contact center w/ default conversation recording. no
problem.
another cool feature. you can route the call based on the Caller ID.
how about InterTalk or InterMedia?
-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA: +1-360-8122281
http://sites.google.com/site/lanvoxphils
On Fri, May 22, 2009 at 11:52 PM, Brian West
rhino used the dual-core atom mobo d945gclf2 but it requires
downloading/building the linux r8168 LAN driver.
-nandy
On Fri, May 1, 2009 at 11:40 AM, Brian West br...@freeswitch.org wrote:
I have two intel atom boxes sitting on a shelf above my desk ... works like
a charm!
/b
On Apr 30,
hi everybody,
i'm looking for a default action in an IVR if the caller doesn't press any
key. for example, the caller will be transferred to the operator (or fifo)
if no key is received after, let's say 5 seconds. is this available in the
IVR? pls show a sample.
tks,
-nandy
hi everybody,
i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's working using
IP phones, softphones and digium FXS port. but there's a problem in dialing
out to PSTN using digium tdm400 fxo - it works fine on the first attempt
(after starting FS) but it fails on the subsequent
case.
Before dialing it waits until it picks up dialtone.
Try the svn trunk version to see if it works any better or verify there is
a dialtone on the line.
On Jan 25, 2009 6:19 PM, Nandy Dagondon g...@i.ph wrote:
hi everybody,
i installed FS 1.0.2 to an Atom mobo (Intel D945GCLF2). it's
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