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 :
>> Are you trying to get a channel variable or capture DTMF input from
>> the
>> caller?
>
> i try to make IVR by php outbound socket. in XML d
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 soun
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 McAl
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.
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
mod_dingali
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 Rene
> wrote:
>> See
>> http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_ev
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 (B-par
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
wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi
wrote:
Hello,
I'm testing FS support for the header Path (FS is behind opensips
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, J
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
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 ./config
Are those in the Tarball?
On Sep 15, 2009, at 11:47 PM, Nandy Dagondon
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, 2009 at 11:26 AM, N
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: doe
It is down at the bottom of the email http://www.cudatel.com
Mike
On Sep 21, 2009, at 7:13 PM, Gavin Henry wrote:
> URL???
>
> On 21/09/2009, Michael Collins wrote:
>> At ClueCon 2009 we had an exciting announcement: Barracuda Networks
>> and the
>> FreeSWITCH team have been working together
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
Did you ever resolve this issue? If not, please make sure you open a
bug on jira.freeswitch.org with as much detail to reproduce this as
possible.
Mike
On Sep 10, 2009, at 6:14 PM, Jan Kubr wrote:
> Hi,
> we have a Freeswitch server on a public IP and a few phones behind
> NAT. The phones a
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 need
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:
>>
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 s
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
is
This is a well known and documented problem already fixed in upstream
kernels. There was quite a bit of discussions about tests that came
out that showed this issue. Opening another bug on it is not likely
to help.
Mike
On Sep 19, 2009, at 2:49 AM, Jason White wrote:
> Karl Vesterling w
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 wrote:
>> What issues are there with libtool 2 under debian? Libtool 2 issues
>> that I am aw
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 he
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 di
If it is no where in the code I would assume it is not implemented,
try catching up with pyite on irc to confirm.
Mike
On Sep 20, 2009, at 12:03 AM, João Mesquita wrote:
> Guys, I have been testing mod_nibblebill lately and there are 2
> params that I could not make work.
>
>
>
>
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 advan
Its probably trivial to add MRCP for them now with the unimrcp lib. I
would suggest making them aware of it. At this point that is the best
way to go so we don't have to support a bunch of custom interfaces.
Mike
On Sep 16, 2009, at 5:21 PM, Gerry Hull wrote:
Has anyone integrated Vestec
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 this
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 Ans
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
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 freesw
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 triggered
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 furthe
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 ES
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 packa
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 lates
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
ething wrong, i
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
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 me
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
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 us
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 v
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 indicat
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?
there is a profile param to enable 3pcc. It should be documented in
the default configs.
Mike
On Sep 29, 2009, at 5:22 PM, Jerry Richards wrote:
> Hello All,
>
> I have an internal extension that needs to send an INVITE without
> SDP body
> (Content Length 0). Freeswitch is replying with 4
This sounds like a bug in the snom to me, we keep changing the expire
on to the future so it should never expire in the first place. You
will have to look at a longer running sip trace to see what exactly is
going on.
Mike
On Oct 1, 2009, at 4:52 AM, Helmut Kuper wrote:
> -BEGIN PGP S
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
>
>
>
>
>
>
>
>
>
>
>
>
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 > wro
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
> 192.168.1.24
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 and a working
call.. What exactly is not working?
Mike
On Oct 4, 2009, at 6:00 PM, Frank Carmickle wrote:
> On Sun, Oct 04, Michael Jerris wrote:
>> Y
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 in
I updated the tiff lib to build better inline, try make tiff-reconf
Mike
On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:
hello,
i just got the last trunk and tried to compile it on one of my
development machines... Well configure fails on tiff-3.8.2 where it
is unable to find Makefile.
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 mi
I sent on the first email of this thread,
converted to text with 'tshark -V -r')
On Sun, Oct 4, 2009 at 5:40 PM, Michael Jerris
wrote:
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 i
check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
Note, you can't just have tone_detect as your last iten in the
dialplan as the call will just get hung up.
Mike
On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
> Hi
>
> I was hoping someone could help me to setu
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 loo
esn't work with media bypass (which I don't
> use).
>
> Thanks!
>
>
> On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris
> wrote:
>> check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>
>> Note, you can't just have tone_detect
http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
>
> Is it possible to treat a recorded voice as voice mail?
>
> Assume that, I've recorded a conversation and I want this recorded
> file to
> be treated like voicemail. So, I could check
On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
> Is is possible to override any of the setting specified in the
> conference profile?
Just the flags you can pass per user such as pin and mute
>
> What I want to do is to have a default profile, and be able to
> modify certain fields if ne
voip codecs is fixed, ptlib I can't recall if we ever did full build
integration or if you needed to manually download the libraries, can
someone who has done mod_opal build on windows comment?
Mike
On Oct 5, 2009, at 5:14 PM, David Clark wrote:
> Ok I found spandsp.h. It is a case of the
As I said in the duplicate thread, the voip codecs issue has been
resolved in trunk, I had a change 1/2 done waiting for testing and it
is now complete.
Mike
On Oct 6, 2009, at 12:30 AM, David Clark wrote:
> No I found the one header. I added it to the include list for the
> project. It in
I am not sure what you mean, do you think that fixes from today should
somehow go somewhere else before we do a release?
On Oct 6, 2009, at 3:21 AM, Vladimir Elizarov wrote:
> Brian West пишет:
>> Because TRUNK is stable... its only fixes going in usually and if
>> things do break they don't s
service? Is it possible to generate curl xml using scripts?
woody
On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris
wrote:
On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
> Is is possible to override any of the setting specified in the
> conference profile?
Just the flags you can pa
Could you open a bug on jira.freeswitch.org as a feature request to
make this a configurable param. (patches that do it even better)
Mike
On Oct 6, 2009, at 12:55 PM, Christian Damianidis wrote:
I’ve tested this and making the change from ANY to BASIC worked.
Thanks for the help.
It no lo
What codecs are all the call legs using, also, please try current svn
trunk.
Mike
On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
>
> Sorry about posting several questions at once, I wasn't aware it's
> "rude".
> Let's concentrate on this issue then.
>
> I use FS rev 14994. Phones on e
switch_ivr_async.c:480
On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:
Hi,
When I record a call in FS, it only creates a 388-byte-long wav
file. The conversation is no written there, and FS deletes the file
when the session finishes.
What can cause this strange behavior?
__
Incorrect NAT configuration so one of the boxes is not actually
getting a BYE.
On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all
users register to the 'external' instance which acts as a gateway by
'internal' in
On Oct 6, 2009, at 4:14 AM, Yehavi Bourvine wrote:
Hello,
We have Polycom and SNOM phones running with FreeSwitch. The
Polycoms have shared lines defined and the SNOMs have both shared
lines and BLFs (defined as extensions in the phone config). I've
tried supporting both, but have some
On Oct 9, 2009, at 7:58 AM, srinivasula reddy wrote:
> Hi all,
>
> does any know about How apr_queue is maintaing and retriving all
> registered and all stuff
>
parse error
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
Hi
can any please tell me where registered calls are stored, so when
incoming call came to mod_sofia.c how it will check it is registered
or not?\\
Calls are not registered and calls have nothing to do with
registration. Users
On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:
> On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins
> wrote:
>> Thanks for reporting back. Please let all the Asterisk users know
>> that they
>> are welcome to join us in #freeswitch on irc.freenode.net and that
>> they will
>> not be ab
On Oct 9, 2009, at 10:40 AM, Maciej Aniserowicz wrote:
Hello,
The issue is resolved. I feel stupid, because Michael Jerris was
right the first time. Setting external_rtp_ip and external_sip_ip to
$${local_ip_v4} made it work.
But the strange thing is: it SOMETIMES worked before without any
There is this endless push and pull on this topic, those who want them
assume it should be default, those who don't assume that should be
default. This probably needs a configuration option defaulting to
pass them (those who don't want to pass them are usually a bit more
educated and would
We don't have session messages directly exposed, except for things
like display, respond, and deflect. What specifically are you trying
to send ?
Mike
On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
> I'm used to using the onInput callbacks inside lua and javascript to
> listen for dtmf a
=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
>
>
>
> Anthony Minessale wrote:
>>
>> you probably have some device lying about ptime everywhere
>> look at a sip trace an pay especially close attention to ptime:x
>> param in
>> sdp
>> if y
On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
> Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
>
> You can pass your parameters in second to these two.
>
> Example:
>
>
>
>
> Where 1 in this case is the number of heartbeats per seconds.
>
Number of seconds between hearb
I am still working on the new build system for esl, stay tuned for
more info soon, it should be in 1.0.5.
Mike
On Oct 11, 2009, at 5:36 PM, Herman Griffin wrote:
Although probably not the best solution, I figured out a way to make
it compile and install:
I removed all of the -Werror insta
Group information is not stored in sqlite, it is pulled from the xml
registry (switch_xml_locate_group function can find them) . Also,
please do not cross post between lists.
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
http://wiki.freeswitch.org/wiki/Mod_commands#in_group
On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris
wrote:
We don't have session messages directly exposed, except for things
like display, respond, and deflect. What specifically are you trying
to send ?
Mike
On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
> I'm used to using the
sched_api is a fsapi command not a dialplan application, I believe
sched_hangup is both.
Mike
On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute("sched_hangup", "+300 alloted_timeout");
However, when I try to run
I would love to see this work in tree, but i am pretty sure it has
never worked. I would gladly accept patches that implement this.
Mike
On Oct 14, 2009, at 2:33 AM, Simon J Mudd wrote:
> br...@freeswitch.org (Brian West) writes:
>
>> You shouldn't have to make clean usually ... doing so mig
There was just a bunch of work on UPDATE, can you confirm this is the
same behavior with trunk?
On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote:
> Hello,
>
> we have the following problem.
> 2 Fax machines are communicating via Freeswitch. One is externally
> attached via a Telco who is able to
Try turning up all the sofia debug to 9.
Mike
On Oct 14, 2009, at 2:16 AM, Szasz Szabolcs wrote:
> Hi,
>
> Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got
> to work TLS between freeswitch and a softphone (phonerlite), but I
> have problem with Audiocodes during the TLS
I think we strongly lean towards using a sub-domain (if necessary) and
maintaining other language content in the same wiki in alternate
pages. If the wiki software we are using is not effective to create
multiple languages we should find a way to do it all in one. We can
set up an additio
If you don't have working stun, jingle is not going to work very
well. It is a required part of the protocol. You need to be able to
determine your external ports for media on each call, using a host
name will not do this for you.
Mike
On Oct 16, 2009, at 10:48 AM, Brian West wrote:
> I
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0
Thanks.
--matt
On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris
wrote:
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
we added some params for new better automatic nat handling, grep the
new defailt configs for localnet and you will find what you are missing.
Mike
On Oct 18, 2009, at 11:14 PM, Chris Fowler wrote:
> I've tried all sorts of debug and parameter changes over the weekend,
> but still can't figure
There is an event you can send as well to switch them, it your trying
to switch it via event socket, that should be better, its not on the
wiki, but
a session message with
eavesdrop-command header with data as the same as dtmf
should do the trick
Mike
On Oct 16, 2009, at 11:54 AM, Nikita Be
Try starting out reading this.
http://wiki.freeswitch.org/wiki/Mod_managed
Mike
On Oct 19, 2009, at 9:14 AM, srinivasula reddy wrote:
> Hi,
>
> How can i use freeswitch.managed project. what are the parameters
> for calling Execute method? and how can i call?
> any help
___
You need a sofia profile for each identity, if your using multiple
external ip addresses, you will need a profile for each. If you are
using bgp or something of the sort and only using one external ip, you
can use a single profile and route using standard routing.
Mike
On Oct 13, 2009, at
I can't recall if we ever exposed an option for this, take a look at
sofia-sip and see if they have a tag to enable this, if so it would
probably be a fairly simple patch to add.
Mike
On Oct 15, 2009, at 3:20 PM, Alexandre Savard wrote:
> Hi,
>
> Does Freeswitch support TLS Client-Authentica
inline is new, it won't work unless your using recent trunk. That
being said, read is not being run inline, so the set is actually being
run before digits_dialed is set. You will most likely need to use
transfer in this situation.
Mike
On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wro
This is what force-rport is supposed to do. That being said, I can't
tell from your trace where it is actually going to, just what it says
in the packet, which can be different.
Mike
On Oct 19, 2009, at 3:23 AM, Tzury Bar Yochay wrote:
> Hi,
>
> I am struggling with a cellular operator whic
Try out trunk and see if this issue is resolved please.
mike
On Oct 19, 2009, at 3:11 AM, Durk de Beer wrote:
Hello,
This is something I came across on Freeswitch 1.0.4
First let me explain what I'm trying to do.
I want Free-Switch to behave as a proxy so in the settings section
of Sofia.con
syntax for this session
message?
I tried this:
sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05
eavesdrop-command: 1
but it doesn't work.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-
users-boun...@lists.freeswitch.org] On Behalf Of Mi
we do have a license for this, people didn't seem to like it last time
we looked at it, I can't recall why.
On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote:
> what about http://www.atlassian.com/software/confluence/ they give
> free licenses to open source project, and FS is using JIRA.
>
>
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