The password is set in conf/autoload_configs/event_socket.conf.xml
-MC
Sent from my iPhone
On Dec 20, 2009, at 7:58 AM, "Joseph L. Casale" wrote:
> Trying to setup a new config in the pfSense 1.2.3 final package and
> when
> I try to connect to the console I get an auth error?
>
> # ./fs_cli
On Dec 17, 2009, at 11:34 PM, Jason White wrote:
> Edmar Cruz wrote:
>>
>> Is there a link or tutorial for the expressions format.
>
> Anything that describes Perl regular expressions should help, and for
> reference, see the pcre(3) manual page, and use the pcretest program
> to
> exper
On Dec 12, 2009, at 7:41 PM, Robert Clayton wrote:
David,
Thanks for your hard work.
Maybe more organization will make the areas needing substance or
explanation more obvious.
That's my hope as well.
-MC
Bob
On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler > wrote:
I am new to FreeS
On Dec 12, 2009, at 7:28 PM, "David V. Fansler" wrote:
> I am new to FreeSWITCH (ok a month old) and am still learning as
> hard as I
> can. In the recent talk about documentation, I had noticed that
> finding
> documentation on the FreeSWITCH wiki was a bit of a chore. So I
> decided to
On Dec 11, 2009, at 4:02 PM, bcxml wrote:
>
> The version is
>
> FreeSWITCH Version 1.0.4 (14460)
>
Ouch. You are nearly 6 months and 1500 revs behind. You badly need to
update to latest trunk.
-MC
>
>>
>>
>>
>>>
>>>
>>
>>
___
FreeSWITCH-users mail
Version 1.0.5 pre 8 is due out any minute. Definitely upgrade to trunk
or at least pre8 when it's available.
-MC
Sent from my iPhone
On Dec 7, 2009, at 6:29 PM, DJB wrote:
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic. I normally check the concurrent calls b
On Nov 6, 2009, at 3:21 PM, Orien Love wrote:
> First of all, Thanks to the help I received on my pfSense
> installation,
> especially to Michael. I have a basic test system up and running. I
> am
> still waiting on some hardware but the base system is working
>
> I am looking on advice
On Nov 6, 2009, at 3:21 PM, Orien Love wrote:
> First of all, Thanks to the help I received on my pfSense
> installation,
> especially to Michael. I have a basic test system up and running. I
> am
> still waiting on some hardware but the base system is working
>
> I am looking on advic
On Nov 6, 2009, at 3:21 PM, Orien Love wrote:
> First of all, Thanks to the help I received on my pfSense
> installation,
> especially to Michael. I have a basic test system up and running. I
> am
> still waiting on some hardware but the base system is working
>
> I am looking on advice
On Nov 6, 2009, at 3:59 PM, "Dave Stevenson"
wrote:
> Hi,
>
> can someone pointme to where the valid dialing strings are specified ?
>
For SIP dialstrings check here:
http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
Also, if you send us examples of what you've tried we ca
I can guarantee that the FS devs are well aware of pj-sip. If it was/
is a viable alternative then it would be considered. The fact that it
isn't being used is a pretty good indication that it isn't suitable
for FS at this time.
-MV
Sent from my iPhone
On Oct 31, 2009, at 1:21 PM, Meftah Ta
Judging by this error I would assume that you're still calling
sched_api as a Dialplan application and not as an FS API command.
You need to figure out how to create an API obj in java and call
sched_api from that object.
-MC
Sent from my iPhone
On Oct 21, 2009, at 2:44 AM, Henry Huang
Everyone repeat after me:
make current is my friend
make current is my friend
make current is my friend
ALWAYS use make current unless you know more about FreeSWITCH than the
project's authors do.
-MC
On Oct 14, 2009, at 8:56 PM, Brian West wrote:
Make current will NOT touch your configs a
On Oct 13, 2009, at 8:30 PM, Henry Huang
wrote:
> Diego:
>
> You probably miss understood me. I said I was able to make
> "sched_hangup" work, but not the "sched_api" in the same way I
> script for "sched_hangup"
>
> The problem was on the second paragraph.
Henry, can you capture the debu
On Oct 13, 2009, at 6:45 AM, lakshmanan ganapathy
wrote:
We are using Reliance as the Carrier.
I think, with this same Reliance carrier, in my office, they are
able to make outgoing calls through asterisk+libpri.
If that's the case I would be very interested in seeing a pri debug
from
cards have all been pretty successful. But thanks
for info! Anyone else using Rhino, Digium, or compatible analog cards?
Thanks!
-MC
> On Sat, Sep 12, 2009 at 21:06, Michael S Collins
> wrote:
>> Gang,
>>
>> Does anyone have a working analog setup? I've reproduc
Gang,
Does anyone have a working analog setup? I've reproduced symptoms on
two different systems, one with the Rhino 8 port card and one with the
Nxtvox TDM400 clone. Symptoms are identical when dialing out on the
FXO port. The outbound dialing established a connection with the
called part
I get the feeling that you are trying to use the wrong tool for the
job. If you need to launch a script after the call ends AND you need
access to the CSV file then you either should switch to XML CDR or
just write a Perl script that runs as a daemon that sits and waits for
CSV files to app
On Sep 5, 2009, at 8:25 PM, Adam Wilt wrote:
> Hi, the documentation says that mod_commands is available from
> within mod_lua. But when I try to access it like this:
>
> session:execute("uuid_broadcast",session_id .. " " .. filename .. "
> both")
>
> I get: Invalid Application uuid_broadcas
Sent from my iPhone
On Sep 4, 2009, at 10:03 PM, Ujjval Karihaloo
wrote:
Would that be firewall on the CentOS machine that FS is installed on?
Correct
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Brian West
Sent from my iPhone
On Sep 3, 2009, at 6:08 AM, 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:
>
> /../../FS/batchfil
Tom,
You should not have to do anything other than make samples. If you are
having these kinds of problems then something went very wrong during
the install. Try starting from scratch just to be sure. Also, what
distro are you running?
-MC
Sent from my iPhone
On Aug 29, 2009, at 8:40 AM,
Sent from my iPhone
On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy
wrote:
No. In the dial plan I said, application="perl" data="The perl
script".
I also checked $session->execute("bridge","user/1010"). This is
working fine.
But originate is not working as I expected.
I think you
Sent from my iPhone
On Aug 26, 2009, at 6:00 PM, "Lars Zeb" wrote:
> Is there a way to dial an external 10-digit phone number, wait a
> second or two after connecting, and then dial a 4-digit extension?
>
>
From the CLI:
originate sofia/gateway/mygw/1234567890 1234
As for the short pause y
I have looked at that but I am confused on which files need to be
edited. Since I have already installed in Wanpipe mode with the
Sangoma card I skipped straight to the Wanpipe section. It mentions
setting the [span wanpie PRI_1] etc in the openzap.conf then further
down it mentions editi
Be sure to visit wiki.freeswitch.org and search for "rosetta stone"
which will take you to a page that helps translate Asterisk concepts
into FreeSWITCH concepts.
It will seem strange at first but once you get over the hump you will
really appreciate the power of FreeSWITCH. Be sure to visi
Jim,
Just curious - could you document this use case on the wiki? Maybe you
could create a page describing the setup and then link to it from the
TLS page.
Thanks,
MC
Sent from my iPhone
On Aug 5, 2009, at 9:07 AM, Jim Burke wrote:
> Hi NOx,
>
> Can you clarify the direction of the calls.
Pastebin your configs. Also, are you using libpri?
-MC
Sent from my iPhone
On Jul 30, 2009, at 7:26 AM, Niall Crosby
wrote:
> Hi,
>
> This might be Sangoma config issue, so apologies in advance for
> posting it here if it is. I am waiting for Sangoma helpdesk to get
> back to me!
>
> But I h
How about this: strftime
Try it at the CLI
-MC
Sent from my iPhone
On Jul 25, 2009, at 7:20 AM, Milena wrote:
> Hello everyone,
>
> I'm using the inbound event socket to receive some information about
> the status of my FreeSWITCH system and i wanted to know if there is
> an api command tha
What svn rev are you on? There have been some important changes
recently.
-MC
Sent from my iPhone
On Jul 19, 2009, at 5:11 AM, Mark Campbell-Smith wrote:
> Hi All,
>
> I know this question has come up before but I couldn't find the answer
> that I could understand! Sorry in advance.
>
> My
I wonder if it would make sense to create a separate sub-command like
"show channels stats" or something. That way we could put all sorts of
nifty info there without breaking the existing command.
Thoughts?
-MC
Sent from my iPhone
On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos
wrote
On Jul 15, 2009, at 5:40 PM, freeswitch-users@lists.freeswitch.org
wrote:
Could you please just tell me where to set it??
The menu actions are defined in conf/autoload_configs/ivr.conf.xml
The audio played for the menus is defined in conf/lang/en/demo/demo-
ivr.xml
-MC
>Please try lo
On Jul 14, 2009, at 5:02 AM, Eli Hayun wrote:
> Hi
> I am not using fixed xml files for the extension registration. I have
> LUA script to return an XML string to FS.
> Everything goes fine until I am trying to get the voice messages.
> When am entering my id, FS (or voicemail module) try to ge
IIRC you need to supply the uuid because the socket doesn't make any
assumptions about the APIs you send.
-MC
Sent from my iPhone
On Jul 11, 2009, at 2:21 PM, Brian West wrote:
I think you do ...
/b
On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote:
Excellent. Do I need to supply uuid on
A few questions for you if I may:
FreeSWITCH doesn't yet have a GUI -are you okay with XML config files?
Do you have TDM circuits for your outbound traffic or are you using a
SIP provider?
BTW, mod_vmd is used to detect an answering machine beep, but it does
not detect human vs. machine. For
Look more closely at the output. It looks like mod_libpri.so didn't
get installed properly. I think this is a bug in the ozmod_libpri
build. For now just locate that missing .so file in your oz build
environment and copy it to the freeswitch/mod directory and try again.
-MC
Sent from my iP
On Jun 30, 2009, at 8:43 PM, Edmar Cruz wrote:
>
> I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i
> shall load
> to G729 for freeswitch that needs a license?
>
You need a license if you are transcoding to or from G729. If you are
just passing the media stream or you stay
If you're already on trunk then just do "make current"
-MC
Sent from my iPhone
On Jun 30, 2009, at 2:04 AM, Saeed Ahmad
wrote:
Hi,
What is the best way to update to latest version if we are already
running an older stable version?
I am using SVN trunk, sources are in /usr/src and its
Couldn't you just throw all the calls into a conference at this point?
-MC
Sent from my iPhone
On Jun 26, 2009, at 9:30 PM, Matthew Fong wrote:
can you 3 way with uuid_bridge?
--matt
On Fri, Jun 26, 2009 at 9:08 PM, Brian West
wrote:
Not sure what you want to do is doable via XML RPC.
On Jun 23, 2009, at 7:04 AM, Max Bridgewater
wrote:
>
> Hi Michael,
>
> Using loopback solves my problem. Thanks a lot.
> There is a strange thing i observed though. I need to paste my
> extension in the default.xml file. Having them in the default
> directory isn't enough. Is that normal?
FYI, the devs report that they are at the restaurant! Last chance to
pitch in and feed the troops. :) hit the paypal button on the main
FreeSWITCH page:
http://www.freeswitch.org
Keep those devs happy and fed and version 1.0.4 will be here before
you know it!
-MC
_
Hehe, where can I buy stock in this company? :)
-MC
Sent from my iPhone
On Jun 15, 2009, at 4:33 AM, David Knell wrote:
> On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote:
>> What is the current status of Freeswitch? Can I safely use it in a
>> large scale commercial environment? How active
Just curious, why are you dialing out the external gw?
-MC
Sent from my iPhone
On Jun 13, 2009, at 11:11 PM, Edmar Cruz wrote:
>
> Ip Windows: 192.168.0.104
> Ip Linux:192.168.0.105
>
>
> My windows:
>
> My sample on sip_profiles/external/dialus.xml
>
>
>
>
>
>data="sofia/
I will clarify the point on the wiki since it is a bit inaccurate
-MC
Sent from my iPhone
On Jun 10, 2009, at 10:40 AM, Brian West wrote:
Yes.
/b
On Jun 10, 2009, at 12:34 PM, Ben Jones wrote:
Does this mean if a user is set for rfc2833 OR info that FS will
generate inband tones to send
Also, could you describe your scenario a bit? It may be that others
have blazed a trail from ast to FS and you can take advantage of their
experience.
-MC
Sent from my iPhone
On Jun 3, 2009, at 9:42 PM, "Joseph L. Casale" wrote:
> I have been lurking for a while here looking for an alternat
You have a few choices. If you can hold out until g729 licensing is
available in FreeSWITCH then that's a viable option. If not then
you'll need to go the hardware route. Go to wiki.freeswitch.org and
search for "mid_dahdi_codec" and you can learn more about the details.
Essentially you can
Did you try clean solution as brian suggested?
-MC
Sent from my iPhone
On May 23, 2009, at 8:33 AM, mszla...@aol.com wrote:
I get a lot of these errors on vc++ express 2008:
59>Linking...
59>LINK : fatal error LNK1181: cannot open input file '..\..\..\..
\w32\library\debug\freeswitchcore.li
We are pretty much booked solid as we've got some unconfirmed speakers
we haven't posted yet. I'm redoing the schedule and will have an
updated one out this next week. One thing that we really need is
backup speakers. Our experience is that there are always people who
have emergencies and c
If you installed the default configuration then you've already got a
PBX. I recommend looking at the default.xml file in conf/dialplan and
also the user setup files like 1000.xml in conf/directory/default.
The default config has 20 users already setup including voicemail. It
also has example
Do you mean from the CDR? I recommend XML CDRs because they give tons
of information. If you are talking about gathering this stuff midcall
then you'll need to supply more information about your setup.
-MC
Sent from my iPhone
On May 10, 2009, at 6:17 PM, Diego Toro wrote:
Hi,
How can I g
On Apr 1, 2009, at 6:59 AM, Raymond Chandler
wrote:
> seven wrote:
>> I know that. And I'd like to read code. Developers written great code
>> and also plenty of comments(which is documentation) in code. However,
>> there are sth. don't need to comment in code but should be available
>> on wiki
On Mar 31, 2009, at 5:59 AM, Anthony Minessale wrote:
> if you set the channel variable 'session_in_hangup_hook=true' early
> in the call, the session will be present in your script.
>
Very cool. I will get this chan var documented on the wiki right away.
-MC
On Mar 30, 2009, at 1:34 AM, François Delawarde wrote:
> Thanks for your quick&clear answers.
>
>
> On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote:
>
>> If you are using on-hook agents, it will place as many outbound calls
>> as there are people waiting.
>> If you are using off-hook a
On Mar 11, 2009, at 8:34 PM, Brian West wrote:
> ;) I expect to see you at cluecon this year?
>
> /b
>
Notice how he threw in a compliment and an invite to CC but didn't
actually address the question? ;) pretty sneaky bkw!
BTW, the best way to come to CC is to get your boss to sponsor the
e
On Feb 19, 2009, at 7:53 PM, Jason White wrote:
> I have it working now. The relevant changes were as follows.
>
>
>
>
>
>
>
Jason,
I like this approach. Good use of the many Dialplan tools.
-MC
___
Freeswitch-users mail
Sent from my iPhone
On Feb 18, 2009, at 6:00 PM, Philip Patterson > wrote:
Hi All.
Have a fresh server and going to install FS on it. Went to the
download page (http://wiki.freeswitch.org/wiki/Installation_Guide)
and tried to download the "Phoenix" build, which is supposed to be
fou
Sent from my iPhone
On Feb 18, 2009, at 3:55 AM, "UV" wrote:
> Awesome work, Kristian!
> And very much needed for the Freeswitch platform (to me, at least).
>
> A suggestion: if the FS team doesn't mind (after getting over the
> naming
> issue), it would be a good idea to put Kristian's late
On Feb 9, 2009, at 11:04 PM, David Knell wrote:
> Oops - I did it again ;-)
>
You Britney Spears wannabe!! :p
-MC
> --Dave
>
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswi
Nik,
I see your point about the wiki entry regarding luasql. If someone
could clarify then I will be happy to help get the wiki documentation
updated appropriately.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:41 PM, "Nik Middleton" > wrote:
Hi Guys
I want to access Mysql 5 from lua
Print out the variable to make sure it is what you expect:
io.write("argv is " .. argv[1] .. "\n";
Also, if you don't give the sound file an absolute path name then it
will automatically use the sound dir path.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:31 PM, "Nik Middleton" > wrote:
Sent from my iPhone
On Feb 7, 2009, at 9:03 PM, "Paul D." wrote:
> Followed Wiki to install and configure mod_cepstral. The problem is FS
> always defaults to one voice, which I installed first, and ignores
> others.
> I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf.
> Af
On Feb 6, 2009, at 6:27 PM, Michael Jerris wrote:
> Not sure what you mean by playback speed. All the prompts for
> voicemail are defined in the phrase macros in the configuration.
Check out conf/lang/en/vm/sounds.xml
-MC
___
Freeswitch-users maili
Nik,
Welcome to FreeSWITCH! The short answer is "yes, FS can do that." The
first thing that you should do is unlearn "the Asterisk way" of
thinking. Usually there is an elegant way of doing things in FS that
wasn't possible in Ast.
I would recommend that you start by looking at the event
Hold tight for an official announcement shortly. :)
-MC
Sent from my iPhone
On Jan 31, 2009, at 9:46 AM, mszla...@aol.com wrote:
What's will be new or fixed in the upcomimg 1.03 FS release?
A Good Credit Score is 700 or Above. See yours in just 2 easy steps!
_
Lookup mod_limit on the wiki.
-MC
Sent from my iPhone
On Jan 31, 2009, at 5:18 AM, shehzad p wrote:
>
> Hi all
>
> Is there any way so that I can limit the concurrent active calls.
> When the call comes to Freeswitch server I route the calls to another
> Gateway using Javascript.
>
> While rout
Turn on XML CDR and try that. Compare the variables in an XML record
to the same CSV record. You'll be able to see them names of the
variables you want to add. Or just use XML :)
-MC
Sent from my iPhone
On Jan 30, 2009, at 4:55 PM, Brian Deacon
wrote:
> Hiya,
>
> So my xml cdr has just o
We don't have any yet but we'd love to get some. If someone is willing
to donate the money then we could have GM Voices do them. Barring
that, if someone has a voice talent who can record the Spanish prompts
and donate them to the project then I'm sure they would be graciously
accepted and
Sent from my iPhone
On Jan 28, 2009, at 6:41 PM, Brian Deacon
wrote:
> Greetings,
>
> So I see from nabble and on the wiki that others have had problems
> with
> sqlalchemy and the hangup hook, but I appear to be having a more basic
> problem.
>
> I'm running FS 1.0.2, python 2.4.3, and Sq
Out of curiosity which SIP messages have you been watching for on the
event socket? Also, how are you connected to the event socket? Are you
subscribing to all events and sifting through them to confirm that no
events are being fired when SIP messages are being sent?
-MC
Sent from my iPhone
Did you get this resolved? Just curious.
-MC
Sent from my iPhone
On Jan 26, 2009, at 5:05 PM, Ron McCarthy wrote:
> Hi,
>
> I am having a weird issue with setting the callerID number for
> outbound calls, I have this:
>
> data="effective_caller_id_number=17025551234"/>
>
>
> Ive set the cal
On Jan 22, 2009, at 3:15 PM, Brian West wrote:
> Not really what I would call a break... but at some point in the $1.6
> million range you stop paying.
>
> /b
Like I said, OSS FTW baby!
-MC
>
>
> On Jan 22, 2009, at 5:13 PM, Kristian Kielhofner wrote:
>
>>
>> Also remembers what happens to vol
On Jan 22, 2009, at 1:55 PM, Kristian Kielhofner wrote:
> On Thu, Jan 22, 2009 at 4:40 PM, Michael Collins
> wrote:
>> On Thu, Jan 22, 2009 at 1:35 PM, Brian West
>> wrote:
>>> Speex, while nice I think it would use more resources in some cases.
>>
>> True. Occasionally more resources, alw
On Jan 22, 2009, at 1:11 PM, "Gregory Boehnlein" wrote:
You use it on your own risk
>>
>> Also, G.729 is patent encumbered big-time. Instead of lining the
>> pockets of lawyers and mega-corporations by perpetuating the use of a
>> crusty old codec we should all twist arms and get our provid
Anthm,
I will add the substance of this to the wiki.
-MC
Sent from my iPhone
On Jan 21, 2009, at 2:09 PM, Anthony Minessale > wrote:
lua:
local event = freeswitch.Event("custom");
js:
e = new Event("custom", "message");
in js you specify a subclass which means you would need to subscrib
Can you join irc later today? I will be on as mercutioviz. I would
like to discuss this more.
-MC
Sent from my iPhone
On Jan 20, 2009, at 2:10 AM, "Krzysztof Zimnicki"
wrote:
>
> >Can you post your openzap.conf file?
> >-MC
>
> Sure.
>
> [span zt]
> name => OpenZAP
> number => 1
> trunk_ty
Guys this is awesome! Helmut, if you need any help with jira just let
me know.
-MC
Sent from my iPhone
On Jan 17, 2009, at 9:20 AM, Brian West wrote:
> Maybe open a jira with this info? Maybe it can all be done as a one
> step process in the ozmod_isdn ;)
>
> /b
>
> On Jan 17, 2009, at 10:5
Tamas,
The channel variable won't work for you if you can't ignore early
media. Your best bet is to use the variable execute_on_answer to
transfer an answered call to a new extension. Then you could just
sleep for 15sec and then check the value of endpoint_disposition.
What is the applicati
On Jan 15, 2009, at 1:02 PM, Brian West wrote:
> On that note the OpenVZ instances could live migrate from box to box
> without dropping calls and usually had a small acceptable blip in
> audio.
>
I'd say a small blip is quite acceptable compared to the alternative!
-MC
> /b
>
> On Jan 15, 2
This looks like a zaptel issue. Do we have any zaptel users familiar
with this issue? If not you should probably check on the asterisk list
for help. Once zaptel is working we can then see about getting openzap
up.
-MC
Sent from my iPhone
On Jan 12, 2009, at 6:36 AM, "Franziska Röhler"
Could you please do a backtrace and post it to a pastebin? If in Linux
do this:
gdb /path/to/freeswitch /path/to/corefile
-MC
Sent from my iPhone
On Jan 12, 2009, at 5:23 AM, shehzad p wrote:
>
> Hi all,
> I am also testing FS release 1.0.2, but I faced strange problem.
> When I stop freesw
Dude, you ROCK!
-MC
Sent from my iPhone
On Jan 9, 2009, at 3:06 PM, Andrew Thompson wrote:
> Hi,
>
> Today I'd like to announce the open-sourcing of a distributed
> callcenter platform I've been designing/building using
> FreeSWITCH/Erlang. The goal is to allow multiple callcenter branch
> offi
:)
Sent from my iPhone
On Jan 9, 2009, at 7:38 AM, Peter P GMX wrote:
> Hello Michael,
>
> sorry for the inconvenience. It turned out that our Telco had to reset
> the second PRI line. Now it works.
>
> Best regards
> Peter
>
> Peter P GMX schrieb:
>> Hello Michael,
>>
>> here is a log of 2 cal
Please try latest first. If you still have errors then put them on
pastebin.
-MC
Sent from my iPhone
On Jan 9, 2009, at 11:36 AM, Tim B wrote:
MC was using 1.0.2. I don't have access to the machine right now.
I can post the errors later tonight or try to build with latest
build. Whic
can I make the debug mode on in freeswitch?
> Can it be done from cli on fly?
> If I put debug mode on, will it affect performance of the
> freeswitch, as the
> freeswitch is currently used as production system.
>
>
> Michael S Collins wrote:
>>
>> So the issue i
Jeng,
Your condition expressions are not right. Could you describe what you
hope to accomplish with those two expressions? Once you get the
regular expressions figured out then it should all work.
-MC
Sent from my iPhone
On Jan 8, 2009, at 6:44 PM, Jian Yuan Peng wrote:
Hi,
Can you he
What FS revision and is this a default Dialplan? Please pastebin the
output of the CLI while making test calls. Be sure to press F8 to
enable debug messages. Also, if you can do so turn on SIP messages by
launching FreeSWITCH with TPORT_LOG=1.
-MC
Sent from my iPhone
On Jan 6, 2009, at 10:
I wonder if putting a sleep statement in your shell script might help.
If it's a timing issue then possibly the shell script is trying to
access the file before FS and/or the OS are done with it. You would
need to tinker with how long to sleep in order to find a value that
works in all case
For the sake of testing can you record a call that gets a busy signal?
At least then we could analyze the audio and see what's going on.
If you need a dialplan example for this let me know.
-MC
Sent from my iPhone
On Dec 29, 2008, at 5:04 AM, Baskar wrote:
Hi Michael,
Steps I follow for
A jira for the core dumps would be good, especially if you can
reproduce the behavior.
Question: where does mod_vmd come into play?
-MC
Sent from my iPhone
On Dec 29, 2008, at 6:28 AM, "Adam Wilt" wrote:
Should I add this to Jira?
On Sat, Dec 27, 2008 at 9:15 PM, Adam Wilt
wrote:
I'
Phil,
Can you do the same test with debug turned on? F8 or "console loglevel
debug" will do the trick.
-MC
Sent from my iPhone
On Dec 25, 2008, at 12:38 PM, can_...@gmx.de wrote:
> Hello,
>
> I am trying to replace some static settings with dynamic ones which
> are provided by a webserver.
I'm pretty sure that this is doable. Could you give us a hint as to
what arguments you want to send? For example, do you have one or more
channel variables you'd like to pass to the shell script?
-MC
Sent from my iPhone
On Dec 23, 2008, at 6:25 PM, Jason White wrote:
> Frank @ Impact wrot
Have you checked out 'sched_api'?
-MC
Sent from my iPhone
On Dec 23, 2008, at 4:25 AM, Alexandru Nedelcu
wrote:
> Hi,
>
> When I make a unsuccesfull call using session.originate, I'd like to
> have a 10 minutes pause and then try again.
>
> For our dialer we are using JS scripts, and setTime
Carole,
Are you calling the hangup app from the Dialplan?
-MC
Sent from my iPhone
On Dec 23, 2008, at 7:04 AM, Brian West wrote:
> Well in this context the phones need to hangup... they aren't going to
> do so automatically. So you'll need to hang up on them or they will
> need to hangup...
Hehe, you just stepped on a land mine! There was A LOT of discussion
about this. The simple fact of the matter is that there was no way to
make everyone happy so the devs chose a layout that might be "ugly" to
some. The key is that XML isn't really "pretty" anyway. The point of
XML is that
I am not a lawyer so I can't tell you for sure. However, I'm not aware
of any US laws against beep detection.
-MC
Sent from my iPhone
On Dec 21, 2008, at 7:09 AM, "Gopalakrishnan A.N"
wrote:
Hi Micheal,
Is it anything like i am violating the laws? please let me know.
On Fri, Dec
529b0
> pool = (switch_memory_pool_t *) 0x80528f0
>
> So I would guess it is trying to access an invalid memory location,
> but
> why, I have no idea
>
> Any ideas?
>
>
> Peder
>
>
> Michael S Collins wrote:
>> Check out this page:
>> wiki.free
Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch
-MC
Sent from my iPhone
On Dec 18, 2008, at 6:38 AM, "pe...@networkoblivion.com"
wrote:
> What is the process for capturing and submitting a core dump?
>
> I am messing around with the Cisco 79x1 phones and tcp and multiple
You've got ignore_early_media set to true but busy signals might be
sent during early media. Why are you ignoring early media?
Also, you might need to check your tone_detect syntax. You're set to
detect 400Hz but you haven't told the system what to do if it does
detect that tone. Please loo
By ignore do you mean filter out? Or do you mean don't do anything but
do audiblize the tones? Do you have some sort of application that does
something with dtmfs?
-MC
Sent from my iPhone
On Dec 17, 2008, at 10:00 AM, "stephen at stephenjc" wrote:
> I have a click to click system written i
Helmut,
Can you turn on full debug and capture the output? It's a lot so put
it in a pastebin.
-MC
Sent from my iPhone
On Dec 17, 2008, at 7:30 AM, Helmut Kuper
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Anthony,
>
> thx, but that doesn't work very good. Outgoing call
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