Your problem is that the url below returns a Not found.
On Oct 1, 2009, at 5:26 PM, Frank Carmickle wrote:
Can someone point out what is wrong here. Thanks.
Siptrace at http://carmickle.com/fs.txt
include
gateway name=voiptalk.org
param name=username value=81100068/
can you send a link of a text sip trace please.
On Oct 1, 2009, at 3:29 PM, Nicolas Brenner wrote:
Any ideas about this?
The SIP provider is offering H323, but I'm not quite sure about
that, is mod_opal working right?
Thanks!
Nicolas
On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner
You will need to setup 2 sip profiles for this setup, one for each
interface.
Mike
On Oct 2, 2009, at 4:10 AM, Timur Irmatov wrote:
Hi.
We have a local network 192.168.1.0/24, where all the users are. Out
FreeSWITCH server is connected to this network, and has ip address
192.168.1.242.
http://wiki.freeswitch.org/wiki/Mod_limit
On Oct 2, 2009, at 8:32 AM, Tihomir Culjaga wrote:
what if you are running some huge traffic e.g. 2000 calls with media?
a typical application for that is an IVR system handling several
different services. I'd like to dedicate some capacity for
Getting documentation on like this on the wiki would be awesome.
Mike
On Oct 2, 2009, at 12:10 PM, Michael Gende wrote:
Hey Orien,
I'm not using exactly your set up, but am using pfsense/FreeBSD.
Since you're using that, I assume you're going dual homed. I've
got a starter guide that
good registers before the failure and a few after the
initial failure.
Mike
On Oct 4, 2009, at 6:40 PM, Frank Carmickle wrote:
On Sun, Oct 04, Michael Jerris wrote:
Is there any info of what I am looking at here, I just went through
1000's of lines that look like repeated good registers
The best way is to start with normalized sound files, then to use
whatever features are available in your tts engine to send the right
volume matched to the sound files. That being said, a new media bug
was just added in trunk for auto gain control and that might help, but
I would never
Could you test this in svn trunk please.
Mike
On Sep 29, 2009, at 9:33 AM, Jonathan Barou wrote:
Hi everybody,
I have a problem with Alphanumeric to numeric user mapping
I have done like it's written here :
...
But when I want to call my alias-number, FS says No Route, Abording
My
Most likely the client NAT is cutting off the translation due to no
traffic. This could be because the client is not sending any traffic,
regardless of settings you set on FreeSWITCH. Try disabling all vad
and dtx on your soft phone to see if this helps. Also, your email
seems to
http://wiki.freeswitch.org/wiki/Installation_Guide#Obtaining_the_Source_Code
http://wiki.freeswitch.org/wiki/Installation_Guide#Compiling_the_Source_Code
On Sep 29, 2009, at 10:19 AM, Jonathan Barou wrote:
I'm sorry but I'm new in the freeswitch communauty, what I have to
do to test this in
we decrement max forwards across a bridge and on transfer, so they are
supposed to sort themselves out automatically, this of course won't
resolve situations like loops via a provider or pstn that do not pass
along max forwards.
Mike
On Sep 29, 2009, at 11:49 AM, Helmut Kuper wrote:
google site:http://lists.freeswitch.org/pipermail/freeswitch-users/ my
search term here, or try nabble.
Mike
On Sep 29, 2009, at 12:35 PM, Jerry Richards wrote:
Sorry for this mundane question, but how do I search mailing
archives for
keywords? The following link has no search option?
configured mod_cdr_csv to dump CDRs. Well it turned out this module
doesn't work as well in the trunk.
Can it be because of AMD opteron + Debian 5.0 enviorment?
There is something in the 1.0.4/trunk version that is wrong for that
kind of event/CDR.
T.
On Fri, Sep 25, 2009 at 6:44 AM, Michael
see chat_send api command and api_hangup_hook. In combination that
might work.
Mike
On Sep 25, 2009, at 6:07 PM, Pete Mueller wrote:
Hello all,
I was wondering if anyone has used mod_dingaling for messaging
rather than voice/video. Specifically, I would like to have FS send
an XMPP
On Sep 25, 2009, at 10:04 PM, Dome Charoenyost wrote:
Dear All,
How to config freeswitch for support this case ?
1. FS register to provider about 50 user account. (Each account
can't support multiple call in same time)
Sofia gateways
2. FS Check account not inuse before
There are a few other things I can think would be nice additions to
mod_managed. Maybe an event handler that does not require a thread to
be sitting and waiting for events trying in a loop would be nice,
instead something that is triggered each time there is a certain event
class
There are a number of examples out there such as:
http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/intralanman/PHP/fs_curl/
Mike
On Sep 24, 2009, at 7:02 AM, Costa Zikalala wrote:
Hi Gabe
Thanks for you response to this question.
Do you perhaps have a link to an example (or just
If you need to be able to do granular permissions like that you would
either need to extend mod_event_socket or write a proxy that handled
that.
Mike
On Sep 24, 2009, at 5:23 AM, Alberto Escudero wrote:
Hi,
Is there any simple way to know:
who is subscribed to certain events via ESL?
I know of at least one person who has had good luck with small
applications on arm, in fact there are good working instructions for
how to cross for arm on the wiki that are known to work.
Mike
On Sep 24, 2009, at 5:34 AM, Cavalera Claudio Luigi wrote:
Hello guys,
lately I've been trying
I can confirm you should not need the swig dependency at all for
anything.
Mike
On Sep 24, 2009, at 1:49 PM, William King wrote:
Hmm... That is interesting... swig is needed I believe only for the
mod_perl or the esl modules. I'll find out more information and put it
on the correct
Can you get these same values in xml-cdr? I don't think csv was ever
intended to work with different cdrs for a and b leg, it was more
intended as a more familiar interface for those coming over from
asterisk.
Mike
On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote:
hello,
i'm on
I think you need to enable presence as well as have the right profile
aliases in place (they are in the default configuration).
Mike
On Sep 24, 2009, at 4:20 PM, RobertT wrote:
Hi guys!
I'm considering to use SIMPLE protocol for IM in my application, but
get a following error trying to
You have access to the full sdp in channel vars, so you can condition
on those with regex.
Mike
On Sep 17, 2009, at 6:25 PM, Tihomir Culjaga wrote:
Hi Michael, thanks for your response.
i think it will be enough to check the call capability... we always
know the call is fax. We just
This should now be resolved in svn trunk.
Mike
On Sep 16, 2009, at 11:39 AM, Christian Löschenkohl wrote:
as a good fs user - of course i am :-) - i made a jira on this
MODAPP-336 to be precise
i hope this helps to solve my problem
br
On 2009-09-16 17:05, Rupa Schomaker wrote:
Either:
What issues are there with libtool 2 under debian? Libtool 2 issues
that I am aware of were all sorted out quite some time ago.
Mike
On Sep 17, 2009, at 10:07 PM, Jason White wrote:
While trying to build FreeSWITCH rev. 14913, compilation failed with
the
following.
the operating
Try taking a list at the info here: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Solaris
You need to be passing any necessary cflags in on configure
Mike
On Sep 18, 2009, at 2:26 PM, email lists wrote:
Forwarding the issue below to see if anyone is familiar with this
if someone can contact we later in the day offlist with credentials
for a box I can try to fix these issues.
Mike
On Sep 23, 2009, at 4:36 AM, Jason White wrote:
Michael Jerris m...@jerris.com wrote:
What issues are there with libtool 2 under debian? Libtool 2 issues
that I am aware
A couple people have taken on major work on packages for ubuntu. Most
of that work will translate directly back to debian, we should just
need people to do testing of debian pacakges once their work is done.
Also we had one more person step up to help with spec file work. I
still need
Are you using freeswitch to detect the inband dtmf or are you getting
both inband and some other method (rfc 2833?) of dtmf as well?
Mike
On Sep 13, 2009, at 10:27 AM, Morten Henckel wrote:
Hi
I need to measure DTM digits duration and interdigit delay for
various phones in a two stage
Please catch up on irc to discuss this real time, this shouldn't be
happening and bkw or I likely will need remote access to your box to
figure out why it is doing this.
Mike
On Sep 21, 2009, at 1:06 PM, Luis Manuel Zuccolo wrote:
I' ve get the same error with a fresh tree
Thanks in
We already have ice support in freeswitch, granted it is the slightly
twisted ice from the old jingle, but this should not be difficult to
fix. Knowing what I know about libnice architechture I can say almost
without doubt that it will never fit well into freeeswitch. Is the
basis of
You can use Answer-State, CS_DESTROY won't happen until the call is
over.
On Sep 23, 2009, at 1:26 PM, Alberto Escudero wrote:
Yes, sounds the best way to go.
I assume that Unique-ID is the unique key to track the call via ESL
Unique-ID: a984afd4-a865-11de-a5b4-fb5a867b002c
and
http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux
Mike
On Sep 23, 2009, at 2:25 PM, Fred-145 wrote:
Hello
I don't have the technical expertise to tell, so here goes: Unless
mistaken,
Freeswitch is written in C and/or C++, so I guess it's
There are tons of details on this at
http://wiki.freeswitch.org/wiki/Mod_xml_curl
Are you having an issue?
Mike
On Sep 23, 2009, at 2:37 PM, Anil Kumar S. R. wrote:
I didn't get much help for my problem with XML CURL. What I meant to
say is, suppose I want to have some 1 users on
This should defiantly be in there, please double check if its in a
different name, and if not, please post a bug to jira.freeswitch.org.
Mike
On Sep 8, 2009, at 5:27 PM, Tina Martinez wrote:
Using the verbose-events definitely improved my ability to see the
custom
variables, but now I
We don't currently have the build environment setup, but I can tell
you it was a pretty basic scratchbox setup and then a normal build
just enabling mod_alsa and other appropriate modules.
Mike
On Sep 18, 2009, at 11:26 AM, Valentin Doroga wrote:
I'm coming again with the same question:
Are those in the Tarball?
On Sep 15, 2009, at 11:47 PM, Nandy Dagondon nandy1...@gmail.com
wrote:
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,
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 wrote:
hi,
i want to enable odbc support which is required in mod_lcr feature.
however, i encounter
We currently don't support forked dialogs.
Mike
On Sep 8, 2009, at 12:16 PM, Humberto Quintana wrote:
Hi Brian,
Thank you very much for your answer but both, Freeswitch and
Kamailio have public IPs, it's my NAT'd IP phone who has private IP
but this is fixed by Kamailio.
The problem is
if you want you could contribute a patch to make that a config option
(of course defaulting to the current value).
Mike
On Sep 4, 2009, at 5:51 AM, Peter P GMX wrote:
Thanks Anthony,
that did the trick.
Best regards
Peter
Anthony Minessale schrieb:
you can edit mod_xml_curl.c line 64
What errors do you get?
Mike
On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote:
Hi,
i am have FS SVN revision 14760, i am trying to use mod_xml_curl
against mod_dingaling. When i call xml_curl url in browser i get
mod_dingaling configuration correctly, also when i do reload
You can do it in perl or lua using a startup script that creates an
event listener.
Mike
On Sep 4, 2009, at 10:32 AM, Mathieu Parent wrote:
Hi
On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Renemrene_li...@avgs.ca
wrote:
See
generally it keeps the overhead of running the script around during
the whole call.
Mike
On Sep 4, 2009, at 10:37 AM, Shameem Shiek wrote:
Hi Michael,
Why is it not recommended to do the brdge app right in the script?
The reason I ask this, I did have lot of trouble using Park/Fifo app
Please open a bug on http://jira.freeswitch.org for this issue.
Please test it on current svn trunk first as well.
Mike
On Sep 4, 2009, at 7:54 PM, DJB wrote:
I have a call transfer problem with Freeswitch
Here is the call flow:
I call from the PSTN (A party) into my Polycom phone
Following up, did a bug get created for this issue?
Mike
On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote:
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi
mayamatake...@gmail.com wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi
mayamatake...@gmail.com wrote:
Hello,
I'm testing FS
This would require changes to the c code in mod_sofia. If you have a
patch to change this behavior (probably should address configuration
and authentication as well as this could be a denial of service path)
you can post it to http://jira.freeswitch.org.
Mike
On Sep 6, 2009, at 6:32 AM,
It should be the same, except using php syntax instead of perl.
Mike
On Sep 2, 2009, at 11:43 AM, Dome Charoenyost wrote:
2009/9/2 Michael Collins m...@freeswitch.org:
Are you trying to get a channel variable or capture DTMF input from
the
caller?
i try to make IVR by php outbound
You can use a phrase macro but I am not sure that we set the position
in a way that you can expand it for the macro.
Mike
On Sep 1, 2009, at 10:37 AM, Dome Charoenyost wrote:
Dear sir,
I want to say posision in queue to caller but
fifo_chime_list can't say more than one sound
somewhere in that mess of my commands solaris is not liking
something. I tested this a lot on solaris and had it working on every
box i was in, so not sure what this could be. If you can get me into
a box in this state via ssh I can take a look.
Mike
On Sep 3, 2009, at 6:42 AM, Bruce
Please post this patch to http://jira.freeswitch.org in the build
system project and I will get this merged in
Mike
On Sep 8, 2009, at 10:28 AM, Christian Löschenkohl wrote:
hi
just a quick patch for the debian init script debian/freeswitch.init
i do use the reload function and the script
We don't currently have a full set of UK English prompts, the prompts
list (soon to be updated with some new prompts) is available at:
http://svn.freeswitch.org/svn/freeswitch/trunk/docs/phrase/phrase_en.xml
If you are going to get a set professionally recorded, we would be
happy to host
In esl you get events for each dtmf.
Mike
On Sep 1, 2009, at 9:54 AM, Greg Thoen wrote:
Thanks for the input.
You'll have to decide on static vs. dynamic based on your needs. In
either case, once the call is connected to your socket you've got
all sorts of control options. PHP has an ESL
nat mapping on the client box not allowing the invite from another ip?
Mike
On Aug 31, 2009, at 3:56 PM, Juan Manuel Vicente wrote:
Hello:
I am using two boxes, with same domain, to register about 1000 users
working together a DNS box doing round robin to resolve two
differentes Ips
compile/make attempt. Let me know if
this
is
something that I can resolve.
Vladimir
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf
Of
Michael Jerris
Sent: Saturday, August 08, 2009 12:37 AM
Try reviewing the debug log to see why it is delayed. You may want to look
at the inline sip trace at all to see if anything is amiss there as well.
Mike
On Sat, Aug 29, 2009 at 2:21 PM, Erik Wickstrom e...@erikwickstrom.comwrote:
Hi,
I'm having some trouble debugging why the transfer
This sounds to me like a phone that doesn't know that a re-invite is
not a new call and session timers are turned on. A sip trace will
show you if this is the case, and if it is, that phone is indeed badly
broken.
Mike
On Aug 29, 2009, at 11:15 AM, Brian West wrote:
Can you get a sip
Running out of stack space? The stack space we run freeswitch in is
fairly small. Programs launched from the freeswitch process inherit
this.
Mike
On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote:
I ran strace from freeswitch and from the command line. lame segfaults
On Aug 26, 2009, at 9:18 AM, Dennis wrote:
hi,
we have set manage-presence = true to see, who is talking on the phone
and who is free. everyone here has his own snom-voip phone with 12
led-lights.
incoming calls are handled in groups. this means, if someone is
ringing, all voip-phones of
On Aug 26, 2009, at 11:34 AM, Dennis wrote:
Do about what? Your description sounds fine, what is the problem
2 people are talking and only ONE led is on? for me it sounds wrong.
the led of p1 AND p2 should be on! the led should indicate, who in the
company/group is busy/talking and who is
I beleive this is following the right rfc rules for dialog matching.
If it is not, please open up a bug on jira.freeswitch.org with
references of what exactly is not right.
Mike
On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga tculj...@gmail.com wrote:
Hello Takeshi,
Thanks for your hint...
You can also do xml config hooks in perl and some of the other
embedded languages.
Mike
On Aug 25, 2009, at 5:22 AM, Juan Backson wrote:
Hi Ken,
xml_curl is a great idea. Is there anyway to not having to setup
another HTTP server? For instance, can I have freeswitch to call an
api or
Culjaga wrote:
Exactly... the scenario i use seems operating on a single thread...
why is that ? can it be changed?
T.
On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris m...@jerris.com
wrote:
Actually in this case, we are bound to one thread in sofia.
Mike
On Aug 25, 2009, at 9:47 AM, Giovanni
if you want to run that at your prompt instead of at the fs_cli you
can do this:
function sofia() { fs_cli -x $(echo sofia $@); }
(thanks ray for the bash foo)
Mike
On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
I get a bash: sofia: command not found. Is there something I need to
add to
On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:
On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga
tculj...@gmail.com wrote:
Of course i removed everytihng from teh dialplan except my
extension :)
when exactly do you react and bring up a new thread ? ... is it on
INVITE or on 1st
It supports FreeSWITCH and soon other engines as well. More
information is at:
http://www.freepbx.org/freepbx-v3
Mike
On Aug 25, 2009, at 2:20 PM, Michael Di Martino wrote:
Is FreePBX V3 based on Freeswitch?
Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd |
That is the remote sdp, not the local sdp. They are sending ptime 20,
not us. Are they actually sending 20 ms packets or are they sending 30?
MIke
On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
Hello Anthony,
I set p...@30i,p...@30i and I can see in the logs that PCMA is used.
However
Do you have an answer in the dialplan for that extension? Also, check
out the ignore_early_media variable.
Mike
On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
Hi,
I managed to get our A500 running with FreeSWITCH 1.0.4 stable using
wanpipe 3.4.4 drivers. But now I have another
Every page on the wiki should be editable. If you don't already have
an account, go to:
http://wiki.freeswitch.org/index.php?title=Special:UserLogintype=signup
Mike
On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:
I think there’s something wrong with the script at
No, you don't get the full sip uri in the dialplan like that. You do
have a whole bunch of variables of the parsed sip header you can use.
Use the info application to see all the vars so you can see what you
have to route the call on.
Mike
On Aug 22, 2009, at 2:40 AM, Henry Huang
a call coming from sofia would never hit that in the dialplan. That
extension is useful for dialing a sip url from mod_portaudio.
Mike
On Aug 22, 2009, at 10:09 AM, Henry Huang wrote:
Jason:
I fully understand how the regex works in the dialplan. If you look
closely in my original email
Check out mod_event_multicast. I think you should be able to share
using that to share events between boxes (although I have never tried
it).
Mike
On Aug 20, 2009, at 4:29 AM, afshin afzali wrote:
Hi Kenneth,
I'm not going to answer your question! Instead I would like to
emphasize on
my bet is if mod_vmd is not getting them that they are not going to
work with tone detect either. Someone needs to look at the tone and
see what frequencies are really involved and if they change throughout
the beep.
Mike
On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:
On Wed,
you can export nolocal the execute_on_answer var to do this. Set it
to playback the file.
Mike
On Aug 19, 2009, at 5:31 AM, Max Ivanov wrote:
Hi all!
Is it possible to execute playback application to legB before
bridge? I mean sequence of actions similar to this:
1. originate legB
2.
try running freeswitch with -vg command line arg.
Mike
On Aug 18, 2009, at 11:04 AM, Juan Backson wrote:
Hi,
I am getting some strange vg malloc error message in
switch_core_hash_insert. Does anyone know what is wrong with these
few lines? Am I missing something?
. If I am missing
anything someone please chime in and correct me.
Mike
Contact Me
SIP
INUM
PSTN
Email
Michael Jerris wrote:
I think sticking with standard WAMP is preferable. What is the
advantage to creating yet another installer over the one that we
have already done
can someone post a patch to that makefile to jira.freeswitch.org please.
Mike
On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a
libesl.so with a cmd like this:
g++ -shared esl_wrap.o -o
Thanks, I'll get that merged in and clear from this we need to start
using proper autoconf checks for java.
mike
On Aug 17, 2009, at 5:35 PM, Fernando Testa wrote:
Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185
Testa
On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris
I think sticking with standard WAMP is preferable. What is the
advantage to creating yet another installer over the one that we have
already done and maintained?
Mike
On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
hello
i'm rewriting this executable file in MSI format
i can use:
increments are in seconds, not microseconds. In IMS for example I
think it defaults to 20 or 30 second nibbles, depending on your
tolerances and billing increments something much larger may even make
sense. Doing billing in sub second increments doesn't make a lot of
sense to me.
I know sangoma is working on their new pri stack, maybe it has proper
qsig support. The openzap isdn stack does not right now and I was
under the impression that the libpri support was pretty limited, but
have no direct knowledge there.
Mike
On Aug 13, 2009, at 3:34 PM, Ryan Wagoner
Issue is we don't handle progress and progress media differently,
maybe we should. The message is however harmless, but annoying and
should probably be dealt with a bit better. Patches welcome.
Mike
On Aug 13, 2009, at 9:04 PM, Moises Silva wrote:
Yes, agreed, but there is no point in
...@freeswitch.org
Date: Mon, Aug 10, 2009 at 8:53 PM
Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
To: freeswitch-users@lists.freeswitch.org
switch_rtp.c has a simple one for the zrtp cache storing.
/b
On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
Re schedule is done in your
for control, my current plan is to use
sched_api to play a file with a short (20ms?) clip of silence,
capture the play_file event and use it to reschule another one for a
couple of seconds later.
I'll let you know what happens.
BB
On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris m
, starting to
flush data buffer...\n);
task-runtime = switch_time_now() + 10;
}
Any suggestion why?
Thanks,
Mark
On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris m...@jerris.com
wrote:
thats in seconds.
Mike
On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
Hi,
Thank you for your help.
I
Please open a bug on jira.freeswitch.org
mike
On Aug 14, 2009, at 6:27 PM, Marc Orenberg wrote:
Hello,
I'm trying to play a prompt to the B-leg of a bridged call in Python.
I place the call to the B-leg, play the prompt, and then bridge it
with the A-Leg, but then FreeSWITCH crashes when
wiki, but I could not
find that function in the mod_dptools.c, shouldn't that be part of
the mod_conference wiki article? =D.
Best regards,
Diego
On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris m...@jerris.com
wrote:
Sip does not support this functionality. The called device would have
My guess is that its the other end killing the call due to rtp
timeouts, not us killing the call. If you can confirm this the best
method would be to get them not to do rtp timeouts.
On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
I'm sure that would work, but I'm worried about it
Please post a bug for this on jira.freeswitch.org.
Mike
On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
This is an integral part of my application. I need to have
FreeSWITCH outside of the media path as well as be able to do
multiple bridges for the same A leg.
/*WORKS*/
action
If your seeing a segfault, please report it to jira.freeswitch.org
with a backtrace and details of how to reproduce.
Mike
On Aug 12, 2009, at 2:37 AM, Charles Boening wrote:
Greetings,
I have the following LUA script (at end of email) in a fresh FS
1.0.4 install. I originally did an
Sip does not support this functionality. The called device would have
to support this via some other mechanism such as ctsa which I have
seen recently someone was looking at for freeswitch. So the first
issue you must resolve is the called device needs to support some way
to do this.
again, this issue should be addressed when you do a sound set for that
dialect, we are attempting to keep the c code common for all dialects
within a language, we will see if this works unless anyone can point
to a place this will not work.
Mike
On Aug 11, 2009, at 3:25 PM, Alan Chandler
We have a plan to address this already, I can't recall if we added the
gender types in code yet.
Mike
On Aug 11, 2009, at 3:56 PM, João Mesquita wrote:
Oops, I thought you were saying different languages. Sorry about that.
jmesquita
On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins
Can you rephrase your question?
On Aug 10, 2009, at 2:16 AM, daqiang wang wangdq@gmail.com wrote:
Hello every one :
I tested FS. and when I use group_call. But I can't call the
extension in group and not registered in FS.
Why ? and What ?
thanks .
sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
- Original Message -
From: Michael Jerris
To: freeswitch-users@lists.freeswitch.org
Sent: Sunday, August 09, 2009 12:19 AM
Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on
holdforbriged channel
What do the debug logs on fs say when you
Re schedule is done in your callback, take a look at places that use
these apis in the code for details.
On Aug 10, 2009, at 6:58 AM, mark morreny markmorr...@gmail.com wrote:
Hi,
Thank you for pointing out sched_api.
What about if I want to do a recurring schedule. is it possible or I
check the sample config files for options to specify these:
http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747
On Aug 6, 2009, at 8:55 PM, Vladimir Rodionov wrote:
Good evening,
This is newbie question.
The FreeSWITCH lua module does not support
is this just the timeout waiting for your max digits?
param name=max-digits value=11/
On Aug 8, 2009, at 4:58 AM, Merul Patel wrote:
Hi,
I came across this thread
(http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html
) from January, and I'm having a
how many does it stop at? is it the same number each time?
Mike
On Aug 8, 2009, at 7:53 AM, Benedikt Fraunhofer wrote:
Hi Phillip,
2009/8/8 Phillip Jones pjinthe...@gmail.com:
Not sure whether this helps but test this without set bypass_media.
In
my setup I have noticed the leg A
loopback ends up using extra threads which we are only able to drop
later in certain situations so it will decrease your total amount of
calls you can do if your not careful with them.
Mike
On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
Thanks very much for that - very help.
Why would
use the SayPhrase method of sessin instead of executing the phrase
application then your input callback should work as expected.
Mike
On Aug 8, 2009, at 11:15 AM, Chad Phillips -- Apartment Lines wrote:
in a lua script, i've tried using session:setInputCallback() to catch
DTMF tones while a
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