What's wrong of the contact string? 639(snom) works but 637(zoiper)
doesn't.
user sip:6...@192.168.1.27:5070;rinstance=e1e47e9a22f3e450;transport=UDP;fs_nat=yes;fs_path=sip%3A637%40192.168.1.27%3A5070%3Brinstance%3De1e47e9a22f3e450%3Btransport%3DUDP
seven
comment lines in the user directory do the trick:
variable name=effective_caller_id_name value=Extension
1000/
variable name=effective_caller_id_number value=1000/
variable name=outbound_caller_id_name value=$$
{outbound_caller_name}/
variable
depending how do you make out going call.
On Jun 23, 2009, at 2:39 PM, Edmar Cruz wrote:
Actually the extension_caller_id=Extension 1001 and
extension_caller_number=1001 is set as Harmeet says but the same issue
FreeSwitch the caller name and the number is 000 i just want
1001 the
3) param name=disable-rtp-auto-adjust value=true/ is not
really required at least for my working setup behind the NAT router.
ok, this param is originally added for another problem
http://jira.freeswitch.org/browse/MODENDP-198
. But I think it might be useful for this.
Thanks. I installed libtiff from source to /usr/local/ and now it works.
On Jun 30, 2009, at 11:49 PM, Michael Jerris wrote:
I think that detection is not working on mac right due to it looking
in default search paths. I am in process of fixing this to use in
tree libtiff soon so this should
for outbound:
extension name=test_gtalk
condition field=destination_number
expression=^(@gmail.com)
action application=bridge data=dingaling/gmail.com/$1/
/condition
/extension
for inbount:
you should be able to set context and extension in your dingaling
chances are the tcp/udp port (5060?) already used by other software,
are you running softphone on the same computer?
On Jul 11, 2009, at 4:29 PM, velusamy velu wrote:
Dear Friends,
When I reload the mod_sofia I have got the following error.
2009-07-11 13:19:32 [ERR] sofia.c:739
ALL,
I know you guys more prefer a CLI version of softphone to a GUI
version. But I still would like to share this:
http://wiki.freeswitch.org/wiki/FsAir
And feel free to give me feedbacks.
I'v only played a few days of ActionScript, it's highly appreciated if
someone can give me help on
Thanks for the great work.
Just want you know that 20 channels with the same username works well on my
server. And echo() works without any problem.
An updated version of Round Robin hunt and a minor bug posted on jira.
Thanks again.
2009/7/27 Giovanni Maruzzelli gmar...@celliax.org
Ciao
Hi Brian,
Sorry responding late. I still cannot get this work, can you take a look?
http://pastebin.freeswitch.org/9877
Everything works fine on Linux but not on my MAC. I have the default ruby
framework and port install on /opt/local/bin/ruby, however, even I changed
the Makefile to use the
On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
Hi,
I'm taking a close look at event socket on FS 1.0.3. Configuration is
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
fsconsole.pl work but I was not able to telnet to port 8021. As I've
done that I received
On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
I've gotten 1.0.4pre9 , but i can not see it :(
-- afshin
In trunk.
On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris m...@jerris.com
wrote:
It is still there.
On Aug 3, 2009, at 8:38 AM, afshin afzali a.afzali2...@gmail.com
wrote:
conference!
Seven Du (seven)
Jonathan Palley (jpalley_idapted)
Idapted Ltd.
*How FreeSWITCH has created hundreds of job opportunities and changed
lives. *
We want to share our experience working with FreeSWITCH. FreeSWITCH has
been a key enabler of our business. We hope this story can
And I added this on the wiki page:
mod_conference and mod_fifo: We also use FreeSWITCH in our office
environment as a PBX for call center and customer service connected with
VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
naturally and just made sales process, business logic
I think you can check the loopback endpoint or inline dialplan.
On Aug 4, 2009, at 3:23 AM, Michael Frager wrote:
Hello,
I'm in the process of moving my VOIP application from Asterisk to
FreeSWITCH.
I was wondering if it is possible to emulate the call announcement
feature that is
variable_originate_disposition
On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
Hi,
Say i originate a call to a mobile phone and the call fails. There
are many possible reasons: congestion, user busy, call rejected by
user, etc. Is there a way i can get the failure code from
mod_easyroute?
2009/8/6 Vladimir Rodionov vladrodio...@gmail.com
Hi, everybody
This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
VOIP provider ABC MUST be routed to the same lua/js/whatever
ALL-
I have a few questions when scripting lua. According to wiki, it is possible
to run looping forever lua scripts through start-up config or luarun.
1) Will the lua script stop when unload mod_lua? I experienced core dump
when unload mod_lua while there was a running lua script. Reported on
) or
regurds
Eli
On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
ALL-
I have a few questions when scripting lua. According to wiki, it is
possible to run looping forever lua scripts through start-up config
or luarun.
1) Will the lua script stop when unload mod_lua? I experienced core
(CHANNEL_HANUP CUSTOM lua::stop) doesn't seem to work. Any
idea to this?
isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
code, too?
On Thu, Aug 6, 2009 at 17:52, Seven Du dujinf...@gmail.com wrote:
for e in (function() return con:pop(1) end) do
btw, the script works
You can run FreeSWITCH as a softphone and control it.
http://wiki.freeswitch.org/wiki/Freeswitch_softphone
2009/8/7 Artem Vasiliev ryde...@googlemail.com
Hi
I have FreeSwitch and external application, which communicates to it via
event socket - listens for events for certain number and gives
On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
Ciao Ivan,
it seems that you do not have the libX11 **development** package
installed.
Unfortunately I don't know about OSX, so I cannot help you, but many
on the list know.
BTW: it will probably be of no use to you to compile
will be
in my branch soon.
7.
On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
Yes, I am interested in this, and if you have any source I could have
a look at it.
Ivan
Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
Ciao Ivan,
it seems
On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvoldi...@myrvold.org
wrote:
Yes, I am interested in this, and if you have any source I could have
a look at it.
Ivan
Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
Ciao Ivan,
it seems
yesterday, and will investigate
more today, to try to understand how you have done the Skype
integration to the Freeswitch in the Carbon code.
And I am glad that someone have contributed to get skypiax working in
OS X. Great work so far!
Ivan
Den 9. aug.. 2009 kl. 20:02 skrev Seven Du:
Ivan
answer only works on outbound event socket. why you don't answer in a
dialplan? what's scenario you use this?
On Aug 11, 2009, at 9:31 PM, Maxim Tsvetov wrote:
I've tried all this command from FS console
and all of them return Unknown command
daqiang wang wrote:
why not use:
On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
Hi,
I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:
$ cd /usr/local/freeswitch/conf/directory/default
created 1050.xml having all '1000' strings replaced by '1050' by
typing
It's not Eyebeam but FS hung up the call because it have nothing to do
after answer.
You should either playback a sound, do the echo command, record, hold
the call, bridge to another channel or transfer somewhere else.
On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
I've tried to use
Got this:
Forbidden
You don't have permission to access /cluecon_2009/presentations/
Dale_Building_FreeSWITCH_App_Lua.pptx on this server.
Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80
$ ping files.freeswitch.org
PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56
Great works. I tested and reported results in jira.
And as I noticed you removed the sequential line hunting methods.
Though I don't use that I think someone else may need that. Think
about the guy want skypeout accounts in a round robin manner, others
might use that in a priority manner,
On Aug 16, 2009, at 12:02 AM, Giovanni Maruzzelli wrote:
On Sat, Aug 15, 2009 at 5:41 PM, Seven Dudujinf...@gmail.com wrote:
And as I noticed you removed the sequential line hunting methods.
Because was broken. So, I aliased it to the RR.
If you think it can be useful, add a Jira for it
Hi,
According to wiki it still in development status, but should compile
right? Any idea about this? thanks.
make
In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for ‘virtual
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)’
Thanks, will try later.
On Aug 16, 2009, at 2:20 PM, Peter Olsson wrote:
Make sure to do a complete rebuild. And also read the comments in
jira MODOPAL-10.
/Peter
On 09-08-16 03.51, Seven Du dujinf...@gmail.com wrote:
Hi,
According to wiki it still in development status, but should
I run into this problem before. Don't remember the exact error but
might be segfault of lame runing in freeswitch-lua.
If you use Linux you would like to try iwatch. It's a perl program
watching your file system and can execute the lame command as soon as
it got the CLOSE_WRITE(or other)
hangup hook api? system or system_bg ? not sure
On Sep 3, 2009, at 9:08 PM, NOx-WHV wrote:
Hi,
does anybody have a tip how to start a batchfile after hanging up.
After ext. 1000 calls 1001 and hang up, i need a request to call:
/../../FS/batchfile 1000
if 1001 calls 1000 i need:
freeswi...@foosball fsctl
-USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap|
restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration
[num]|loglevel [level]]
On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote:
After some testing (fs_cli -x 'fsctl shutdown
gm,
Thanks a lot you can merge into the mainline. I check into my branch
because it's currently not as useful as on Linux and Windows and the
solution is not good. But it works and it is a good start that
mod_skypiax runs on OSX. Sure it would be easier for people want to
test and improve
I'm not sure this and don't have time to debug. But last time I tried
incoming calls worked by setting skype to Auto-answer calls. Can you
try that?
On Sep 6, 2009, at 3:19 PM, Ivan C Myrvold wrote:
I have got outgoing call to Skype to work, and the audio quality is
excellent. But I also
On Sep 6, 2009, at 7:25 AM, Tapan Parikh wrote:
Yes, thanks a lot!
Im still having a bit of trouble getting it working though. When
skypiax_proxy starts, I get the following errors:
2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init
theDOProxy
2009-09-05 15:39:42.139
As a work around, record to stereo, and use sox to split channes ?
On Sep 9, 2009, at 12:44 AM, Anthony Minessale wrote:
that would have to be filed as a feature request as we do not
currently have a way to do that.
On Mon, Sep 7, 2009 at 11:50 PM, Matthew Fong mattdf...@gmail.com
Hi MC,
Months ago we had tried the multi-language plugin on MediaWiki, I know
you are still planning to do this but I just want how far it goes.
Count me in when you are short of hand.
On Sep 15, 2009, at 11:25 PM, Michael Collins wrote:
Demuel,
Thanks for the input. Yes, we want to avoid
I think the file was there but deleted by FreeSWITCH if it thinks it
was too short (like 3 seconds?). If I'm not wrong, someone requested
this feature becuase FreeSWITCH left too many small recordings.
On Sep 17, 2009, at 1:27 AM, João Mesquita wrote:
I think you need to upgrade your
sorry when I said on profile I want to say one profile
2009/9/24 Seven Du dujinf...@gmail.com
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
2009/9/24
It not possible to use 0.0.0.0 for on profile. however, you can create more
sip profiles for each of your interfaces. Search freeswitch-users archievs
then you will find similar topics.
2009/9/24 Yehavi Bourvine yehavi.bourv...@gmail.com
Hello,
I am trying to run FreeSwitch on a machine
I would write a simple lua script to do a round robin hunt.
2009/9/27 Dome Charoenyost d...@tel.co.th
2009/9/26 Brian West br...@freeswitch.org:
Why would they require you to have 50 accounts? Doesn't seem sane to
me.
They provide for pc to phone user and i want to use for my corp.
:)
what's your rev? I think rev14494 might related to you.
2009/9/27 Vinuth Madinur vinuth.madi...@gmail.com
Hello All,
I'm trying to do a simple dialer, where I am:
1. Initiating mod_vmd on channel answer.
2. Staying quiet until there is a beep.
3. Leave a message on beep.
4. Hangup.
(in
Hi, is this a bug?
freeswi...@internal regex 10|09|10
false
freeswi...@internal regex 10|10
true
freeswi...@internal regex 10|(09|10)
false
freeswi...@internal 2009-09-27 11:47:00.815355 [ERR] switch_regex.c:101
COMPILE ERROR: 4 [missing )][(09]
the first one should be true?
Thanks. But I think it would be nicer if the regex looks the same as in
dialplan. can we add a optional separator arg on this case?
regex data|pattern [seperator]
regex 10:09|10 :
2009/9/29 Brian West br...@freeswitch.org
Yep escape it.
/b
On Sep 28, 2009, at 10:47 AM, Michael Collins
FS support recording to mp3 directly through mod_shout but you might not
want to use that for performance reason.
You can use lame to convert .wav to .mp3 regularly( by crontab if you on
linux) or immediately after record(by using iwatch, or listening to event
socket to see when the record is
maybe you can check this: http://www.gsmopen.org/
2009/10/6 Moiz Chinoy moizchi...@gmail.com
Hi,
Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
use this as a GSM gateway?
--
Regards,
Moiz Chinoy.
___
FreeSWITCH-users
change the xml and execute reloadxml in FS console or fs_cli
or you can check mod_xml_curl
2009/10/6 srinivasula reddy srinivas.ksvre...@gmail.com
Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.
Thanks
Srinivasula Reddy K
Yes, it's discussed before.
http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
set that var to false before you record.
2009/10/10 Jason White ja...@jasonjgw.net
Nagalenoj nagale...@gmail.com wrote:
No, When I do voicemail_inject and check through voicemail, it is not
search this list, just has been discussed.
2009/10/10 velusamy velu velu.techni...@gmail.com
Dear All,
Could you please any one explain the difference between parking and
valet parking?
___
FreeSWITCH-users mailing list
stereo files it'll just mux them down to mono
before playing... can you elaborate on the error you're getting?
/b
On Oct 10, 2009, at 12:40 AM, Seven Du wrote:
Yes, it's discussed before.
http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
set that var to false before you
I set to true because brian said it can play stereo files but no lucky for
me.
2009/10/12 Jason White ja...@jasonjgw.net
Seven Du dujinf...@gmail.com wrote:
originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
bridge(sofia/gateway/yy/yy)
Shouldn't
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
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
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
that will make life easier.
2009/10/13 Brian West br...@freeswitch.org
Does anyone see a problem with hosting mod_h323 in our SVN? I would
like to centralize everything we can to reuse our issue tracking
resources and not fragment the community if possible.
/b
On Oct 12, 2009, at 2:43
There are some plugins for mediawiki to support multilanguage, either inline
or on a separate page. However, both have Pros and Cons. e.g.
multi-language support this with a separate page
some_page #default to en
some_page/fr Franch
some_page/zh Chinese
Pros: clear, available languages can be
not sure about this, but did you try send dtmf to uuid
723f3dbb-b87b-4cd4-98fc-698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
3d18cc6e8454 ?
2009/10/20 Nikita Belov nbe...@abisoft.spb.ru
Yeah. I'm using it for starting eavesdrop:
SendMsg cced4b9a-b6de-4be1-8c12-3d18cc6e8454
I also got some zombie channels, if someone can help me take a look that's
really nice.
http://pastebin.freeswitch.org/10912
I only loaded mod_cdr_csv
Is it ok to use mod_xml_cdr?
Thanks.
2009/11/1 Dome Charoenyost d...@tel.co.th
I found bug in fscore_pb $pwd should be $mypwd
Now i post
Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.
-- Forwarded message --
From: Seven Du dujinf...@gmail.com
Date: Sun, 1 Nov 2009 10:24:32 +0800
Subject: Fwd: [Freeswitch-users] Many
?
On Nov 1, 2009 7:45 PM, Seven Du dujinf...@gmail.com wrote:
Just suspicious would be possible that happened on sqlite stage? I
manually deleted the channels from sqlite and nothing bad happend.
just FYI.
-- Forwarded message --
From: Seven Du dujinf...@gmail.com
Date: Sun, 1 Nov
2009/11/6 Giovanni Maruzzelli gmar...@celliax.org
On Thu, Nov 5, 2009 at 6:57 PM, Seven Du dujinf...@gmail.com wrote:
Ciao Giovanni,
Do you still plan to merge this?
Sorry Seven,
I've lost track of this, and now I'm so sick I'm completely un-useful ;).
That's OK, we all have a lot
Would it be better to change the list subject prefix from [Freeswitch-users]
to FreeSWITCH-Users?
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
I once wrote a patch for fifo delete, but didn't submit to jira. If
someone think it's useful to merge into trunk, I think I can still find the
code, but sure need to test with the current trunk.
2009/11/19 Michael Collins m...@freeswitch.org
On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi
lol:
2009/11/19 Anthony Minessale anthony.miness...@gmail.com
maybe you could send them 183 then 4 180's or send them an invite and
pretend to deadlock and not send any more sip traffic as a way of
identifying yourself
On Wed, Nov 18, 2009 at 4:05 PM, Brian West br...@freeswitch.org wrote:
did you try without any .wav or .PCMU?
2009/11/23 Matthew Fong mattdf...@gmail.com
I'm trying to conserve processor power by recording in native file format,
PCMU in my case. It works great with the following line
session:execute(record,
/tmp/my_recording...session:getVariable(read_codec));
And because it's static string for on-hook members, it's hard to set
dynamically. For now, I'm using a callback way - whenever the sip client
answered the call, it fetch the real connected number from a http server.
That's not ideal because not only it add the complexity but also the callee
have
Yes, that's what we are doing.
2009/11/24 Brian West br...@freeswitch.org
You do realize that the whole concept is OLD skewl. You should be
popping this info via external resources when the agent is bridged to
the caller and the info is there before they are done saying thanks
for calling
XML has basic conditioning, but lua rocks.
-- Time condition for sales 1
--session:setAutoHangup(false)
function do_transfer(extn)
--print(extn)
session:transfer(extn, XML, sales)
end
now = os.date(%H:%M)
w = tonumber(os.date(%w))
if w = 1 and w =5 then
if ( now = 09:00
http://code.google.com/p/mod-recpld/
It's out-dated. I originally wrote it to record raw G.729 codec on
passthrough mode. It worked before and then we abandoned that since We felt
G729 cannot deliver good sound particularly on a cross-continent network.
The code is written when I don't know much
And you may also would like to update the wiki as well if the var is not
there.
2009/11/26 kokoska rokoska kokoska.roko...@post.cz
Thank you very much, Anthony, for your help!
I'm nearly at current trunk (15653) and
action application='set' data='RECORD_MIN_SEC=1'/
works great :-)
Many
Yeah, that's why I had to record to two files(readwrite) and need to mix
together by using sox. Do you only try to using PCMU to save CPU power
matt? As Anthony said, the difference can be ignored. And you also need to
take extra effort to make sure transcoding will not happen on a
conversation.
Not sure about js, but in lua, you can use luarun to run a
long-running script like
loop
do sth.
sleep 5min
end
and also it can be set to start with freeswitch in lua.conf.xml
I guess you can also use jsrun to run js.
And, if you run every 5 min, why not use crontab?
fs_cli -x jsrun xx.js
why not try mod_xml_cdr?
2009/12/4 Mouncif Benniane mounci...@gmail.com:
is it possible to run a javascript at the end of dialplan to generate cdrs?
because (mod_cdr_csv) is giving me hard time as it rotates Master file on
machine reboots or shutdown signals.
javascript or LUA for
You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ?
2009/12/4 Samuel Abekah-Mensah ab...@greatiam.com:
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and
the answer is yes but where would you store the collected info?
Thanks
On Dec 3, 2009, at 7:02 PM, Seven Du dujinf...@gmail.com wrote:
why not try mod_xml_cdr?
2009/12/4 Mouncif Benniane mounci...@gmail.com:
is it possible to run a javascript at the end of dialplan to
generate cdrs?
because
Hi,
I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
to trunk code, no sound after att_xfer.
Then I rebuild FS 15807 with a fresh checkout, but still using the old
conf/ settings, sound is ok, but there are other problems:
A call B, and B att_xfer C
1)
Thanks, done.
2009/12/7 Michael Jerris m...@jerris.com:
Please report bugs to jira.freeswitch.org.
Mike
On Dec 6, 2009, at 11:45 PM, Seven Du wrote:
Hi,
I know there's some chang on att_xfer, and after upgrade(re-bootstrap)
to trunk code, no sound after att_xfer.
Then I rebuild FS
I also have this problem on a trunk version more than 1000 revisions
behind, so I think the best way is to upgrade to trunk and report this
again if still have problem.
2009/12/8 DJB djbin...@yahoo.com:
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic. I normally
I think if you listen to CUSTOM FIFO::INFO you can get
Caller-Caller-ID-Number on event socket.
2009/12/13 Diego Toro dft...@yahoo.com
Hello,
I want to know how can I get caller id after call is out queue fifo, I read
about fifo_caller_consumer_import and fifo_consumer_caller_import
you can use the same ip with different port
2009/12/13, Yehavi Bourvine yehavi.bourv...@gmail.com:
Hello,
In the WIKI page that talks about Freeswitch performance there is a
sentence:
*libsofia only handles 1 thread per profile, so if that is your bottle neck
use more profiles*
How can
I'm using contrib/seven/sip/sip2db.rb
2009/12/18 David Villasmil david.villasmil.w...@gmail.com:
i agree with christian, though i would use tshark. you can actually
get the fields you want (method and callid) and store them in a dB.
then you need to match them with a query. it is simple but
I couldn't guess what you want, pastbin your full config and logs and
give more detail of your story perhaps someone can help you.
2009/12/18 yvonne ding yhding2...@yahoo.ca:
param name=username value=1101
param name=password value=1234
param name=proxy value=192.168.129.194:5060
param
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