this action can be accomplished using Group Dialing (Sequential). this may
not answer your problem but have you considered it?
-nandy
On Tue, Dec 8, 2009 at 2:31 AM, Peter P GMX wrote:
> I have a Problem with continue_on_fail.
>
>
> I have setup a hunt group
>
> data="sofia/external/2...@10.1
have you created Extension 1002?
-nandy
On Wed, Dec 9, 2009 at 3:20 AM, mailinglist 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:
> default (dir)
> default
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 wrote:
> Add the following:
>
> .
>
> after
>
>data="insert/${doma
hi simon, implementation is almost the same. here's my dialplan:
... add more files to play here ...
via " where might
> be a country, or area and may be the name of one of several
> SIP gateways I have configured. Each phrase has been a prerecorded wav
> file played i
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 wrote:
> also, you can store files in PCMA/PCMU format and avoid transcoding at
> all... and as said disk space is cheap.. go get
please check Redfone's foneBridge. i come across an article about this
before.
On Tue, Sep 29, 2009 at 5:26 AM, SP wrote:
> Yes, with OpenSER/Kamailio/OpenSIPs/SER (you pick a name).
>
> On Mon, Sep 28, 2009 at 15:12, Mike van Lammeren
> wrote:
> > Hello!
> >
> > I followed a tutorial that desc
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 wrote:
> I think the file was there but deleted by FreeSWITCH if it thinks
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 wrote:
> Are those in the Tarball?
>
>
> On Sep 15, 2009, at 11:47 PM, Nand
mike, got it from tarball. - nandy
On Wed, Sep 16, 2009 at 11:51 AM, Michael Jerris 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
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 wrote:
> is the Erlang source needed in the FS sou
is the Erlang source needed in the FS source directory?
/nandy
On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon wrote:
> the ./configure script aborts after the last error message. any hint where
> to look for the problem? tks.
>
> /nandy
>
>
>
> On Wed, Sep 16,
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 wrote:
> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote:
> > hi folks, anyone encountered this problem? t
hi folks, anyone encountered this problem? tks.
/nandy
On Mon, Sep 14, 2009 at 2:20 PM, Nandy Dagondon wrote:
> meftah,
>
> i disabled mod_erlang_event in modules.conf. unixodbc is installed already.
> still ... the same error message. tks for your input.
>
> /nandy
>
>
dbc 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:
>
> ..
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... /us
o see sample configs
in the distribution.
tks for the clarification,
/nandy
On Thu, Sep 10, 2009 at 5:52 PM, Jason White wrote:
> Nandy Dagondon wrote:
> > for outside clients to register w/ the internal profile, the router has
> to
> > forward port 5060 to FS. am i correct?
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 wrote:
> Those configs will still work.
> /b
>
> On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote:
>
> Hi there,
>
> the
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 wrote:
> Hi,
> I`m trying to implement Call Forwarding in my FS setup. I set a user
> variable managing the type of forwarding (busy,no ans
; force FS to negotiate
> G729 on leg A. But Leg B is willing to use G711 and FS is unable to
> transcode G729 <---> G711.
>
> I was wondering 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
&
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 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 phone is using
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.
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, 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 posts.
>
> I have another q
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 wrote:
> Where can I find a sample tones.conf file for the UK? Am trying to
> configure a USBFXO device for outbound calls.
>
> Thank
ed,
i mean you use separate extension names:
btw, you should also use separate gateway names "sip1" and "sip2". so
differentiate them in the bridge application.
On Mon, Jul 27, 2009 at 4:16 PM, Jason White wrote:
> Edmar Cruz wrote:
> >
> > Not working just the same both of th
in my implementation, i would use 2 separate conditions that looks like
this:
On Mon, Jul 27, 2009 at 2:42 PM, Edmar Cruz wrote:
>
> Hi FS Users,
>
> I just want to try multiple gateways. It works actually like this...
>
>
>
>But I test call like 513 at 222.333.444.5
ok. w/ my apologies. - nandy
On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice 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
> *Reply-To: *
> *D
just bumping this topic.
-nandy
On Fri, May 8, 2009 at 12:44 AM, Fred-145 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 pcs
>
> Unfo
we have a forum on compact,fanless last may.
On Sun, Jul 5, 2009 at 6:21 AM, William Suffill
wrote:
> Ya I have a SheevaPlug but yet to have anything interesting to report
> about making it do anything. The potential is there tho.
>
> -- W
>
> ___
> Fr
is PCMA enabled in X-Lite, too?
On Wed, Jul 1, 2009 at 9:25 PM, qian ma wrote:
>
> inbound_codec_negotiation is generous
> and the xlite PCMU is enabled.
>
> my var.xml.conf:
>
>
>
>
>
> 2009/7/1 Nandy Dagondon
>
>> sorry. i mean check the x-lite
sorry. i mean check the x-lite client if PCMA is enabled?
On Wed, Jul 1, 2009 at 9:14 PM, Nandy Dagondon wrote:
> check the value of "inbound_codec_negotiation" in the sip_profiles/*.xml
> files. is it "generous" or "greedy"? you should also chec
; i add "PCMA" in global_codec_prefs and outbound_codec_prefs, it doesn't
> work. FS only accept PCMU.
> why??
>
>
>
>
> 2009/7/1 Nandy Dagondon
>
> you FS doesn't accept PCMU. try to add "PCMU" on both variables.
>>
>&
you FS doesn't accept PCMU. try to add "PCMU" on both variables.
On Wed, Jul 1, 2009 at 3:44 PM, qian ma wrote:
> thanks for your replies.
>
my var.xml:
>
>
>
>
> below is the sip trace:
> recv 781 bytes from udp/[58.212.219.104]:40508 at 07:42:37.086711:
>
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
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, 2
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 wrote:
>
>
>
>
>
> data="effective_caller_id_name=${effective_caller_id_name}"/>
> data="effect
ows internal IP address on the external
profile. it should display the external IP address instead.
On Sun, Jun 21, 2009 at 2:20 PM, Nandy Dagondon wrote:
> the default setting is "auto-nat".
>
> i changed ext-sip-ip=$${external_sip_ip} and
> ext-rtp-ip=$${extern
DNS lookup?
>
> jmesquita
>
>
> On Sun, Jun 21, 2009 at 3:20 AM, Nandy Dagondon 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:
ason White wrote:
> Nandy Dagondon 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
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
_
calls.
FS community is really great!
tks once again,
nandy
On Sun, Jun 21, 2009 at 11:50 AM, David Knell 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
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 setup.
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
Phone
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 wrote:
> You shouldn't be having problems... what version are you using?
> /b
>
> On Jun 5, 2009, at 9:58 PM, Nandy Dagondon wrote:
>
dujinfang,
hv u tried OSLEC? it's really reduced echo even on the cheapy X100P card on
*. oslec works w/ FS, too.
-nandy
On Sat, Jun 6, 2009 at 10:11 AM, dujinfang wrote:
>
> On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:
>
>
>
> On Thu, Jun 4, 2009 at 6:59 PM, seven wrote:
>
>> I'm using
there 10 client seats so at max. 10 simultaneous calls. however, the number
of clients may be increased.
-nandy
On Sat, Jun 6, 2009 at 10:32 AM, Brian West wrote:
>
> On Jun 5, 2009, at 9:24 PM, Nandy Dagondon wrote:
>
> we experience some latency in the recording files even wi
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 recor
what are the error messages in the FS CLI output?
On Thu, Jun 4, 2009 at 3:10 AM, Matthew Lockwood 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 it with Zoiper, YakaPhone
10:05 AM, Brian West 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
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 ap
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 wrote:
> I say bkw
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 wrote:
> I have two intel atom boxes sitting on a shelf above my desk ... works like
> a charm!
> /b
>
> On Apr 30, 2009, at 10:31 PM
hi guys,
i've installed FreeNas using CF-to-IDE adaptor and SanDisk 128MB CF. it's
working fine. but i want to try FS on a 16GB Kingston CF. anyone tried this?
if none, i can also settle down for 8GB. pls mention which brand/size works.
tks,
nandy
On Thu, Apr 30, 2009 at 10:46 PM, Fred-145 wr
perhaps you're referring to VPN (Virtual Phone Number). you can visit
http://www.ipkall.com that offers free Washington state numbers.
On Sun, Mar 15, 2009 at 12:47 AM, Meftah Tayeb wrote:
> hello,
> please ho to get a free did number ?
> also, is it pocible to link it to my FS ?
> thanks
>
> __
take a look at intel atom mobo d945gclf2 (dual-core). it has one PCI slot
and an S-Video and VGA video ports. rhino is using this platform.
On Sat, Mar 7, 2009 at 12:30 AM, Fred wrote:
> Hello
>
>I'm looking for a small, lunchbox-like PC to build a small-form
> factor CRM server to sell
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
_
detecting a dial tone on the failure 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" wrote:
>
> hi everybod
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 attempts.
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